From covici at ccs.covici.com Fri Sep 1 05:00:35 2017 From: covici at ccs.covici.com (John Covici) Date: Fri, 01 Sep 2017 01:00:35 -0400 Subject: [Freeswitch-users] mod_v8" I cannot build using master branch Message-ID: Hi. I am trying to update to masterand mod_v8 will not build and I have no idea even what this error means. Here is the complete build log. It is saying ninja subcommand failed. https://covici.com/owncloud/index.php/s/6MBIR4hnWdu8lWG Am I doing something wrong or should I file a bug? Thanks in advance for any suggestions. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici at ccs.covici.com From sebastian_ml at gmx.net Fri Sep 1 06:21:06 2017 From: sebastian_ml at gmx.net (Sebastian Kemper) Date: Fri, 01 Sep 2017 06:21:06 +0000 Subject: [Freeswitch-users] mod_v8" I cannot build using master branch In-Reply-To: References: Message-ID: Hi, Looks like something is missing: ../../third_party/llvm-build/Release+Asserts/bin/clang++: error while loading shared libraries: libtinfo.so.5: cannot open shared object file: No such file or directory If libtinfo.so.5 is also somewhere in the third_party directory maybe an rpath should be added to clang++. Alternatively having a system libtinfo.so.5 should work as well. Regards, Seb Am 1. September 2017 07:00:35 MESZ schrieb John Covici : >Hi. I am trying to update to masterand mod_v8 will not build and I >have >no idea even what this error means. Here is the complete build log. >It is saying ninja subcommand failed. > >https://covici.com/owncloud/index.php/s/6MBIR4hnWdu8lWG > >Am I doing something wrong or should I file a bug? > >Thanks in advance for any suggestions. > >-- >Your life is like a penny. You're going to lose it. The question is: >How do >you spend it? > > John Covici > covici at ccs.covici.com > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://confluence.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org From covici at ccs.covici.com Fri Sep 1 07:26:54 2017 From: covici at ccs.covici.com (John Covici) Date: Fri, 01 Sep 2017 03:26:54 -0400 Subject: [Freeswitch-users] mod_v8" I cannot build using master branch In-Reply-To: References: Message-ID: Thanks a lot! I had to recompile the ncurses libs (I am not using Debian) with a special flag for tinfo and it worked. On Fri, 01 Sep 2017 02:21:06 -0400, Sebastian Kemper wrote: > > Hi, > > Looks like something is missing: > > ../../third_party/llvm-build/Release+Asserts/bin/clang++: error while loading shared libraries: libtinfo.so.5: cannot open shared object file: No such file or directory > > If libtinfo.so.5 is also somewhere in the third_party directory maybe an rpath should be added to clang++. Alternatively having a system libtinfo.so.5 should work as well. > > Regards, > Seb > > Am 1. September 2017 07:00:35 MESZ schrieb John Covici : > >Hi. I am trying to update to masterand mod_v8 will not build and I > >have > >no idea even what this error means. Here is the complete build log. > >It is saying ninja subcommand failed. > > > >https://covici.com/owncloud/index.php/s/6MBIR4hnWdu8lWG > > > >Am I doing something wrong or should I file a bug? > > > >Thanks in advance for any suggestions. > > > >-- > >Your life is like a penny. You're going to lose it. The question is: > >How do > >you spend it? > > > > John Covici > > covici at ccs.covici.com > > > >_________________________________________________________________________ > >Professional FreeSWITCH Consulting Services: > >consulting at freeswitch.org > >http://www.freeswitchsolutions.com > > > >Official FreeSWITCH Sites > >http://www.freeswitch.org > >http://confluence.freeswitch.org > >http://www.cluecon.com > > > >FreeSWITCH-users mailing list > >FreeSWITCH-users at lists.freeswitch.org > >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >http://www.freeswitch.org > -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici at ccs.covici.com From gregor at infomedia.si Fri Sep 1 07:45:19 2017 From: gregor at infomedia.si (Gregor Nanger) Date: Fri, 1 Sep 2017 09:45:19 +0200 Subject: [Freeswitch-users] Inbound call - Dialed number on transfer In-Reply-To: References: Message-ID: sip_invite_to_uri did the trick 2017-08-31 12:51 GMT+02:00 Gregor Nanger : > Ok, thank you. Will try this. > > What about this: https://wiki.freeswitch.org/wiki/Variable_sip_route_uri > in real life example on page? > > Best regards, Gregor > > 2017-08-31 12:44 GMT+02:00 Alexandr Popov : > >> what about to use remote-party-id? >> >> 2017-08-31 12:17 GMT+03:00 Gregor Nanger : >> >>> Can you please help me with best practice. >>> >>> I am transfering incoming call to extenstion via bridge to >>> user/user at domain. But I want that sip client sees on which number >>> (external DID) call was placed. Which variable should I set? >>> >>> Thank you, Gregor >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Gregor Nanger > > *CTO* > t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 > • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia > • www.infomedia.si > -- Gregor Nanger *CTO* t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia • www.infomedia.si -------------- next part -------------- An HTML attachment was scrubbed... URL: From nikitanikitenko at gmail.com Fri Sep 1 14:40:37 2017 From: nikitanikitenko at gmail.com (=?UTF-8?B?0JzQuNC60LjRgtCwINCd0ZbQutGW0YLQtdC90LrQvg==?=) Date: Fri, 1 Sep 2017 17:40:37 +0300 Subject: [Freeswitch-users] Fwd: Call drops after 32 seconds with ACK Timeout In-Reply-To: References: Message-ID: Hello! I have server at digital ocean with a floating ip address and I have routed sip traffic to this floating ip. At this server installed kamailio which balancing between freeswitch servers. I have a problem that call drops after 32 seconds with error ACK Timeout. Traffic comming throug the kamailio proxy. kamailio.cfg - https://pastebin.com/x0CL3TEZ kamailio sip messages log - https://pastebin.com/wyT3cBbD freeswitch logs with sip debug - https://pastebin.com/zPysfQkV I'm struggling for a week and have no idea how to fix it. Thank you very much. Mykyta Nikitenko -------------- next part -------------- An HTML attachment was scrubbed... URL: From sean at missionlabs.co.uk Fri Sep 1 15:49:27 2017 From: sean at missionlabs.co.uk (Sean Ingham) Date: Fri, 1 Sep 2017 16:49:27 +0100 Subject: [Freeswitch-users] Fwd: Call drops after 32 seconds with ACK Timeout In-Reply-To: References: Message-ID: Once a call is answered FreeSWITCH sends a 200 OK SIP message to your Kamailio loadbalancer. As per the SIP spec your kamailio box should respond to a 200 OK with an ACK. The FS logs show that it never receives an ACK, and it repeatedly tries sending the 200 OK with no response. After 30 seconds without an ACK FS will assume the call has failed and hangup. Looking at your Kamailio SIP logs you can see the 200 OKs getting through okay, and no attempt to send an ACK back to FS. I think this indicates an issue with your kamailio.cfg rather than FreeSWITCH. Sean www.wirex-precision.co.uk On Fri, Sep 1, 2017 at 3:40 PM, Микита Нікітенко wrote: > > Hello! I have server at digital ocean with a floating ip address and I > have routed sip traffic to this floating ip. At this server installed > kamailio which balancing between freeswitch servers. I have a problem that > call drops after 32 seconds with error ACK Timeout. Traffic comming throug > the kamailio proxy. > > kamailio.cfg - https://pastebin.com/x0CL3TEZ > kamailio sip messages log - https://pastebin.com/wyT3cBbD > freeswitch logs with sip debug - https://pastebin.com/zPysfQkV > > I'm struggling for a week and have no idea how to fix it. > Thank you very much. > Mykyta Nikitenko > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From khorsmann at gmail.com Fri Sep 1 17:04:38 2017 From: khorsmann at gmail.com (Karsten Horsmann) Date: Fri, 1 Sep 2017 19:04:38 +0200 Subject: [Freeswitch-users] Fwd: Call drops after 32 seconds with ACK Timeout In-Reply-To: References: Message-ID: Hi Mykyta, since you are plain sip without encryption you can see your sip-traffic on the wire, use sngrep on the kamailio box to figure out whats with the sip-dialogs - it helps a lot. https://github.com/irontec/sngrep and if its more an kamailio config problem, it would be better to stay at sr-users at . Or you can setup an webhomer for debugging. Kind Regards 2017-09-01 17:49 GMT+02:00 Sean Ingham : > Once a call is answered FreeSWITCH sends a 200 OK SIP message to your > Kamailio loadbalancer. As per the SIP spec your kamailio box should respond > to a 200 OK with an ACK. The FS logs show that it never receives an ACK, > and it repeatedly tries sending the 200 OK with no response. After 30 > seconds without an ACK FS will assume the call has failed and hangup. > > Looking at your Kamailio SIP logs you can see the 200 OKs getting through > okay, and no attempt to send an ACK back to FS. I think this indicates an > issue with your kamailio.cfg rather than FreeSWITCH. > > > Sean > www.wirex-precision.co.uk > > On Fri, Sep 1, 2017 at 3:40 PM, Микита Нікітенко < > nikitanikitenko at gmail.com> wrote: > >> >> Hello! I have server at digital ocean with a floating ip address and I >> have routed sip traffic to this floating ip. At this server installed >> kamailio which balancing between freeswitch servers. I have a problem that >> call drops after 32 seconds with error ACK Timeout. Traffic comming throug >> the kamailio proxy. >> >> kamailio.cfg - https://pastebin.com/x0CL3TEZ >> kamailio sip messages log - https://pastebin.com/wyT3cBbD >> freeswitch logs with sip debug - https://pastebin.com/zPysfQkV >> >> I'm struggling for a week and have no idea how to fix it. >> Thank you very much. >> Mykyta Nikitenko >> >> >> >> -- Mit freundlichen Grüßen *Karsten Horsmann* -------------- next part -------------- An HTML attachment was scrubbed... URL: From henning at inter.net Fri Sep 1 17:05:28 2017 From: henning at inter.net (Henning Heedfeld) Date: Fri, 1 Sep 2017 19:05:28 +0200 Subject: [Freeswitch-users] detect_speech via Event Socket not working Message-ID: Hi, I cannot get the detect_speech app working. I fire the command as following: SendMsg 31ada607-aede-4073-86ff-4dcc318ee839 call-command: execute execute-app-name: detect_speech execute-app-arg: unimrcp:uni2 builtin:speech/transcribe This is the only result I see in fs_cli: 2017-09-01 17:15:26.518812 [DEBUG] switch_ivr.c:623 sofia/internal/1001 at tst Command Execute detect_speech(unimrcp:uni2 builtin:speech/transcribe) EXECUTE sofia/internal/1001 at tst detect_speech(unimrcp:uni2 builtin:speech/transcribe) When I use the play_and_detect_speech app instead, the MRCP connection is established immediately and everything is working fine. SendMsg 31ada607-aede-4073-86ff-4dcc318ee839 call-command: execute execute-app-name: play_and_detect_speech execute-app-arg: /tmp/beep.wav detect:unimrcp:uni2 builtin:speech/transcribe What am I doing wrong? Thanks for any hints, Henning -------------- next part -------------- An HTML attachment was scrubbed... URL: From steveayre at gmail.com Fri Sep 1 19:08:13 2017 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 1 Sep 2017 20:08:13 +0100 Subject: [Freeswitch-users] session timer - enable/disable per call In-Reply-To: References: Message-ID: AFAIK it's negotiated session timers before hitting the dialplan so you couldn't do that without significant changes to FS. Can you run two sofia profiles on different ports with and without timers? You could get the vendor to use the profile with timers disabled. If they follow it you might be able to redirect them in dialplan from one profile to the other on a call-by-call basis. On 31 August 2017 at 16:27, Gabriel Kuri wrote: > Is it possible to either: > > a) enable/disable the session timer per call, via the dialplan? > b) change the session timeout value per call, via the dialplan? > > Currently, it's set in the sofia profile, either: > > > > or > > > > > However, it would be great if it could be changed/set per call. > > If you ask why, a particular device when it receives a re-invite > mid-call decides to re-invite an already established T.38 call to PCMU > which breaks the FAX transmission. > > Yes, this is a long fax :) > > Unfortunately, the vendor won't fix, so it'd be nice for these devices > to be able to simply disable or set the timer to a really long time in > the dialplan. > > Thanks ... > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From npaleari at phonecallsrl.com.ar Sat Sep 2 00:36:24 2017 From: npaleari at phonecallsrl.com.ar (Phonecall SRL - Nicolas Paleari) Date: Fri, 1 Sep 2017 21:36:24 -0300 Subject: [Freeswitch-users] Stop packet radius Message-ID: <2dc43389-c8e9-5522-5ec9-f74aa4e3d6e8@phonecallsrl.com.ar> An HTML attachment was scrubbed... URL: From colton.conor at gmail.com Sat Sep 2 19:12:51 2017 From: colton.conor at gmail.com (Colton Conor) Date: Sat, 2 Sep 2017 14:12:51 -0500 Subject: [Freeswitch-users] RFC 4662 Support in Freeswitch? Message-ID: Does Freeswitch support RFC 4662 for BLF Resource list URI? Asterisk does: https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=30278158 They have a good overview page Kamillio does: https://kamailio.org/docs/modules/3.2.x/modules_k/rls.html Good info from Grandsteam about why to use this: http://www.grandstream.com/ sites/default/files/Resources/GXP21x0_Eventlist_BLF_Guide.pdf Same with Polycom: http://community.polycom.com/polycom/attachments/ polycom/VoIP/19112/1/Technical%20Brief%20-%20Busy%20Lamp%20Field.pdf Others have asked this question on the freeswitch users list, but responses were limited. -------------- next part -------------- An HTML attachment was scrubbed... URL: From fs at teamviewer.com Mon Sep 4 09:22:39 2017 From: fs at teamviewer.com (Freeswitch) Date: Mon, 4 Sep 2017 09:22:39 +0000 Subject: [Freeswitch-users] Channel Event Issue Message-ID: Hi, since updating from FreeSWITCH 1.6.8 to FreeSWITCH 1.6.17 (Windows Server 2012 R2), the Event Socket Handler(inbound mode) sporadically stops sending Channel related Event Info. Once in this situation, the Event Socket Handler does not recover until either reset of the ESL connection by the Host that established the ESL connection or reload of mod_event_socket. Shortly before Event Socket connection drops it seems that there is a problem trying to close a channel. Example Log for failure case: Channel with uuid d9be476b-6a9d-4c96-95fb-5d8728bcc08e is never closed after leavig the conference. d9be476b-6a9d-4c96-95fb-5d8728bcc08e 2017-08-29 10:59:35.399989 [NOTICE] switch_channel.c:1104 New Channel sofia/external/149 at dialin [d9be476b-6a9d-4c96-95fb-5d8728bcc08e] d9be476b-6a9d-4c96-95fb-5d8728bcc08e 2017-08-29 10:59:35.399989 [INFO] mod_dialplan_xml.c:637 Processing 149 <149>->4600 in context public d9be476b-6a9d-4c96-95fb-5d8728bcc08e 2017-08-29 10:59:35.399989 [NOTICE] switch_ivr.c:2172 Transfer sofia/external/149 at dialin to XML[4600 at default] d9be476b-6a9d-4c96-95fb-5d8728bcc08e 2017-08-29 10:59:35.419958 [INFO] mod_dialplan_xml.c:637 Processing 149 <149>->4600 in context default d9be476b-6a9d-4c96-95fb-5d8728bcc08e 2017-08-29 10:59:35.439966 [NOTICE] mod_dptools.c:1312 Channel [sofia/external/149 at dialin] has been answered d9be476b-6a9d-4c96-95fb-5d8728bcc08e 2017-08-29 10:59:55.940144 [NOTICE] sofia.c:1012 Hangup sofia/external/149 at dialin [CS_EXECUTE] [NORMAL_CLEARING] d9be476b-6a9d-4c96-95fb-5d8728bcc08e 2017-08-29 10:59:55.940144 [INFO] conference_loop.c:1469 Channel leaving conference, cause: NORMAL_CLEARING Example Log for regular case: Channel with uuid 982453be-a321-4825-a944-ecfffa715914 is closed correctly after leaving the conference): 982453be-a321-4825-a944-ecfffa715914 2017-08-29 10:55:51.340132 [NOTICE] switch_channel.c:1104 New Channel sofia/external/149 at dialin [982453be-a321-4825-a944-ecfffa715914] 982453be-a321-4825-a944-ecfffa715914 2017-08-29 10:55:51.340132 [INFO] mod_dialplan_xml.c:637 Processing 149 <149>->4600 in context public 982453be-a321-4825-a944-ecfffa715914 2017-08-29 10:55:51.360126 [NOTICE] switch_ivr.c:2172 Transfer sofia/external/149 at dialin to XML[4600 at default] 982453be-a321-4825-a944-ecfffa715914 2017-08-29 10:55:51.360126 [INFO] mod_dialplan_xml.c:637 Processing 149 <149>->4600 in context default 982453be-a321-4825-a944-ecfffa715914 2017-08-29 10:55:51.380148 [NOTICE] mod_dptools.c:1312 Channel [sofia/external/149 at dialin] has been answered 982453be-a321-4825-a944-ecfffa715914 2017-08-29 10:56:11.900321 [NOTICE] sofia.c:1012 Hangup sofia/external/149 at dialin [CS_EXECUTE] [NORMAL_CLEARING] 982453be-a321-4825-a944-ecfffa715914 2017-08-29 10:56:11.900321 [INFO] conference_loop.c:1469 Channel leaving conference, cause: NORMAL_CLEARING 982453be-a321-4825-a944-ecfffa715914 2017-08-29 10:56:11.900321 [NOTICE] switch_core_session.c:1683 Session 9744 (sofia/external/149 at dialin) Ended 982453be-a321-4825-a944-ecfffa715914 2017-08-29 10:56:11.900321 [NOTICE] switch_core_session.c:1687 Close Channel sofia/external/149 at dialin [CS_DESTROY] The channel with uuid d9be476b-6a9d-4c96-95fb-5d8728bcc08e will still be displayed at fs_cli on request of "show channels" freeswitch at 1250-0110> show channels uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,accountcode,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num,initial_cid_name,initial_cid_num,initial_ip_addr,initial_dest,initial_dialplan,initial_context d9be476b-6a9d-4c96-95fb-5d8728bcc08e,inbound,2017-08-29 10:59:35,1503997175,sofia/external/149 at dialin,CS_EXECUTE,149,149,dialin,4600,playback,{playBackId=2573}us/announcement.wav,XML,default,L16,8000,128000,PCMA,8000,64000,,1250-0110,,,,ACTIVE,,,,,,,149,149,dialin,4600,XML,public 1 total. However, trying to retrieve the channel data for this channel e.g. "uuid_dump" fails. freeswitch at 1250-0110> uuid_dump d9be476b-6a9d-4c96-95fb-5d8728bcc08e -ERR No such channel! Same issue exists with FreeSWITCH 1.6.19. We are continuously sending api commands to the channel. Could this be a race condition, receiving an api, while the Channel is trying to close ? Would bgapi help ? Why is it not an issue with FreeSWITCH 1.6.8 ? Regards, Tobias -------------- next part -------------- An HTML attachment was scrubbed... URL: From dragos.oancea at vonage.com Mon Sep 4 14:46:25 2017 From: dragos.oancea at vonage.com (Oancea, Dragos) Date: Mon, 4 Sep 2017 15:46:25 +0100 Subject: [Freeswitch-users] Playback of AMR-WB files. In-Reply-To: <41D1E239-0B68-499F-8653-D5D1CB3580A7@jerris.com> References: <2D2AD379-782F-4F3E-934A-2A119BAA85DA@jerris.com> <41D1E239-0B68-499F-8653-D5D1CB3580A7@jerris.com> Message-ID: Hi Mike, I did the PR, please see FS-10642 . Ivan, I think that trick would not work when trying to record when in BE mode: https://tools.ietf.org/html/rfc4867#section-5 . And yes, it will not work when the mode is changed during the call, but I guess the mode change is uncommon, even if AMR means actually "Adaptive Multi-Rate" :) . Cheers, Dragos On Wed, Aug 23, 2017 at 11:25 PM, Michael Jerris wrote: > Thats a bug. AMR-WB is not fixed rate so it should be setting that value > to 0. That change will make mod_native_file not load it. Toss me a pull > request to fix and I’ll merge it. > > > On Aug 23, 2017, at 6:06 PM, ha.ppy.neko wrote: > > Indeed. If mod_amrwb is compiled in paththrough mode it sets > encoded_bytes_per_packet to 0 and mod_native_file skips this codec. > But if mod_amrwb is compiled with transcoding support it sets > encoded_bytes_per_packet to SWITCH_AMRWB_OUT_MAX_SIZE constant (61) and > mod_native_file registers AMR-WB extension. That was the case for my > installation. > > So my theory was that native playback fails because of incorrect packet > size. > According to module source AMR-WB has 9 bitrate modes with corresponding > voice frame sizes (in bytes): > 0 - 17 > 1 - 23 > 2 - 32 > 3 - 36 > 4 - 40 > 5 - 46 > 6 - 50 > 7 - 58 > 8 - 60 > > I recorded test RAW file with FS and record application at max bitrate and > resulting file was indeed stream of 62 bytes RTP payloads. 2 bytes AMR-WB > header + 60 bytes of voice data. > I recompiled mod_amrwb with encoded_bytes_per_packet set to 62 and was > able to successfully play RAW file! > There was warning "switch_core_file.c:358 File /tmp/recording.AMR-WB > sample rate 8000 doesn't match requested rate 16000" but it sounded fine > nevertheless. > > I think this will work with passthrough mode too (change > encoded_bytes_per_packet param from 0 to 62), but I don't have time to test > it yet. > > Of course I don't know if this change will not break other FS functions, > any ideas? > > And second problem is that caller may request lower bitrate with frame > size 40 for example and RAW file will be sliced incorrectly. > Thats the real deal breaker. Any tips how to make native playback honor > negotiated codec and set proper encoded_bytes_per_packet value? > > > 2017-08-23 15:57 GMT+03:00 Michael Jerris : > >> That was actually fixed recently. In current master code it won’t >> register them. >> >> On Aug 22, 2017, at 6:50 PM, ha.ppy.neko wrote: >> >> I am not sure, but it looks like for given mode AMR and AMR-WB files have >> fixed frame size. >> "The length of the speech frame is implicitly defined by the mode >> indicated in the FT field": >> >> https://tools.ietf.org/html/rfc4867#page-35 >> >> Anyway, it is strange that mod_native_file registers AMR and AMR-WB >> formats if it could not support them. >> >> 2017-08-23 0:21 GMT+03:00 Anthony Minessale >> : >> >>> The question was answered by the log line you supplied. >>> >>> -- "cannot play or record native files with variable length data". >>> >>> VBR codecs are not a predictable size so you cannot make a raw file of >>> it because you don't have any idea how big each packet is. >>> >>> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From joe at expert.net Mon Sep 4 23:31:47 2017 From: joe at expert.net (Joseph Barrero) Date: Mon, 4 Sep 2017 18:31:47 -0500 Subject: [Freeswitch-users] Cisco IP Phones & Counterpath Softphones Crashing w/ mod_av Enabled Message-ID: Hi, I’m running Freeswitch v1.6.19. When mod_av is enabled, inbound calls to Cisco SPA504g and SPA514g ip phones crash as they start to ring. The crash seems to occur immediately after Freeswitch sends SIP Invite with the SDP information that includes the video details. I don’t have any problems when I disable mod_av and include PCMU, PCMA, and OPUS. The latest versions of X-Lite and Bria 5 also crash with mod_av enabled. Bria 4 and other models of ip phones (Polycom, Yealink, Grandstream) don’t seem to have a problem. Any ideas what could be wrong? - Joe Barrero -------------- next part -------------- An HTML attachment was scrubbed... URL: From denys.pozniak at crazycall.com Tue Sep 5 11:23:25 2017 From: denys.pozniak at crazycall.com (Denys Pozniak) Date: Tue, 5 Sep 2017 13:23:25 +0200 Subject: [Freeswitch-users] FS with Homer. error: tport_logging: capture socket error: Protocol error Message-ID: Hello! I have FreeSWITCH installation with Homer. But after some period of time FS stops sending traffic to Homer with error in console: freeswitch at freeswitch01> error: tport_logging: capture socket error: Protocol error error: tport_logging: capture socket error: Protocol error error: tport_logging: capture socket error: Protocol error Please help me to find the source of this issue. System has next parameters: freeswitch at freeswitch01> version FreeSWITCH Version 1.6.17~64bit ( 64bit) root at freeswitch:/# cat /etc/lsb-release DISTRIB_ID=Ubuntu DISTRIB_RELEASE=16.04 DISTRIB_CODENAME=xenial DISTRIB_DESCRIPTION="Ubuntu 16.04.2 LTS" root at freeswitch:/# cat sofia.conf.xml And each profile has : BR, Denys -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Tue Sep 5 12:36:23 2017 From: david.villasmil.work at gmail.com (David Villasmil) Date: Tue, 5 Sep 2017 14:36:23 +0200 Subject: [Freeswitch-users] B-leg transfer Message-ID: I have this dialplan i'd like to do, let's see if i can explain it :) I have a call coming in on 5080, which i forward to another box. When B-leg is answered, I want to disconnect the A-leg and bridge B-leg to a conference. I'm trying with the following: The idea is to answer the call with the second extension and execute_on_answer to transfer the b-leg to extension 9999 on public, which sends it over to the conference... But this doesn't seem to be working, the call seems to be sent to 9999 at public: 2017-09-04 23:09:21.243247 [NOTICE] sofia.c:8159 Channel [sofia/external/4345412121] has been answered EXECUTE sofia/external/4345412121 transfer(-bleg 9999 at public) 2017-09-04 23:09:21.243247 [DEBUG] switch_ivr.c:2165 (sofia/external/ 12345 at 1.2.3.4) State Change CS_EXECUTE -> CS_ROUTING 2017-09-04 23:09:21.243247 [NOTICE] switch_ivr.c:2172 Transfer sofia/external/12345 at 1.2.3.4 to XML[9999 at public@public] I've also tried doing uuid_transfer on the bleg like so: and also instead of uuid_transfer, i tried putting the bleg on hold like: But is seems as soon as the aleg dies, the whole thing dies Help is much appreciated! Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 ᐧ -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Tue Sep 5 12:45:01 2017 From: david.villasmil.work at gmail.com (David Villasmil) Date: Tue, 5 Sep 2017 14:45:01 +0200 Subject: [Freeswitch-users] FS with Homer. error: tport_logging: capture socket error: Protocol error In-Reply-To: References: Message-ID: +1 I get the same, even though the box _IS_ sending to Homer, really weird... it could be sometime sending fails, because i have traced with tshark and i see fs sending packets... ᐧ Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 On Tue, Sep 5, 2017 at 1:23 PM, Denys Pozniak wrote: > Hello! > > I have FreeSWITCH installation with Homer. > But after some period of time FS stops sending traffic to Homer with error > in console: > > freeswitch at freeswitch01> > error: tport_logging: capture socket error: Protocol error > error: tport_logging: capture socket error: Protocol error > error: tport_logging: capture socket error: Protocol error > > Please help me to find the source of this issue. > > > System has next parameters: > > freeswitch at freeswitch01> version > FreeSWITCH Version 1.6.17~64bit ( 64bit) > > root at freeswitch:/# cat /etc/lsb-release > DISTRIB_ID=Ubuntu > DISTRIB_RELEASE=16.04 > DISTRIB_CODENAME=xenial > DISTRIB_DESCRIPTION="Ubuntu 16.04.2 LTS" > > root at freeswitch:/# cat sofia.conf.xml > > > > > > > > > > > > And each profile has : > > > > BR, > Denys > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From sean at missionlabs.co.uk Tue Sep 5 13:45:53 2017 From: sean at missionlabs.co.uk (Sean Ingham) Date: Tue, 5 Sep 2017 14:45:53 +0100 Subject: [Freeswitch-users] B-leg transfer In-Reply-To: References: Message-ID: I had a similar use case to this although I use Event Socket to control the call rather than an XML dialplan. You could do something like this: Incoming call from A-Leg hits a dialplan extension that calls the API 'originate' command to create a new call between the proxy endpoint and your conference application. The originate command would look something like this: originate sofia/gateway/proxy/$1 &bridge(9999 at public) This new call is completely independent of the A-leg, so should persist when the A-leg hangs up. Some info on calling api commands from the dialplan can be found here: https://wiki.freeswitch.org/wiki/Mod_commands#From_the_Dialplan Hope that helps! Sean. http://www.wirex-precision.co.uk On Tue, Sep 5, 2017 at 1:36 PM, David Villasmil < david.villasmil.work at gmail.com> wrote: > I have this dialplan i'd like to do, let's see if i can explain it :) > > I have a call coming in on 5080, which i forward to another box. > When B-leg is answered, I want to disconnect the A-leg and bridge B-leg to > a conference. > > I'm trying with the following: > > > > > > > > > > > > > > > > > The idea is to answer the call with the second extension and > execute_on_answer to transfer the b-leg to extension 9999 on public, which > sends it over to the conference... > > But this doesn't seem to be working, the call seems to be sent to > 9999 at public: > > 2017-09-04 23:09:21.243247 [NOTICE] sofia.c:8159 Channel [sofia/external/ > 4345412121 <(434)%20541-2121>] has been answered > EXECUTE sofia/external/4345412121 <(434)%20541-2121> transfer(-bleg > 9999 at public) > 2017-09-04 23:09:21.243247 [DEBUG] switch_ivr.c:2165 (sofia/external/ > 12345 at 1.2.3.4) State Change CS_EXECUTE -> CS_ROUTING > 2017-09-04 23:09:21.243247 [NOTICE] switch_ivr.c:2172 Transfer > sofia/external/12345 at 1.2.3.4 to XML[9999 at public@public] > > I've also tried doing uuid_transfer on the bleg like so: > > > > inline="true"/> > > > > > > > and also instead of uuid_transfer, i tried putting the bleg on hold like: > > > > But is seems as soon as the aleg dies, the whole thing dies > > > Help is much appreciated! > > > > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 <+34%20669%2044%2083%2037> > ᐧ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Tue Sep 5 14:16:28 2017 From: david.villasmil.work at gmail.com (David Villasmil) Date: Tue, 5 Sep 2017 16:16:28 +0200 Subject: [Freeswitch-users] B-leg transfer In-Reply-To: References: Message-ID: Thanks for your reply! I use originate extensively, but it never occurred to me to use it that way... this way would be like using the inbound just as a trigger, I like that! Thanks again On Sep 5, 2017 15:46, "Sean Ingham" wrote: > I had a similar use case to this although I use Event Socket to control > the call rather than an XML dialplan. You could do something like this: > > Incoming call from A-Leg hits a dialplan extension that calls the API > 'originate' command to create a new call between the proxy endpoint and > your conference application. The originate command would look something > like this: > originate sofia/gateway/proxy/$1 &bridge(9999 at public) > > This new call is completely independent of the A-leg, so should persist > when the A-leg hangs up. > > Some info on calling api commands from the dialplan can be found here: > https://wiki.freeswitch.org/wiki/Mod_commands#From_the_Dialplan > > Hope that helps! > > Sean. > http://www.wirex-precision.co.uk > > > > On Tue, Sep 5, 2017 at 1:36 PM, David Villasmil < > david.villasmil.work at gmail.com> wrote: > >> I have this dialplan i'd like to do, let's see if i can explain it :) >> >> I have a call coming in on 5080, which i forward to another box. >> When B-leg is answered, I want to disconnect the A-leg and bridge B-leg >> to a conference. >> >> I'm trying with the following: >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> The idea is to answer the call with the second extension and >> execute_on_answer to transfer the b-leg to extension 9999 on public, which >> sends it over to the conference... >> >> But this doesn't seem to be working, the call seems to be sent to >> 9999 at public: >> >> 2017-09-04 23:09:21.243247 [NOTICE] sofia.c:8159 Channel [sofia/external/ >> 4345412121 <(434)%20541-2121>] has been answered >> EXECUTE sofia/external/4345412121 <(434)%20541-2121> transfer(-bleg >> 9999 at public) >> 2017-09-04 23:09:21.243247 [DEBUG] switch_ivr.c:2165 (sofia/external/ >> 12345 at 1.2.3.4) State Change CS_EXECUTE -> CS_ROUTING >> 2017-09-04 23:09:21.243247 [NOTICE] switch_ivr.c:2172 Transfer >> sofia/external/12345 at 1.2.3.4 to XML[9999 at public@public] >> >> I've also tried doing uuid_transfer on the bleg like so: >> >> >> >> > inline="true"/> >> >> >> >> >> >> >> and also instead of uuid_transfer, i tried putting the bleg on hold like: >> >> >> >> But is seems as soon as the aleg dies, the whole thing dies >> >> >> Help is much appreciated! >> >> >> >> Regards, >> >> David Villasmil >> email: david.villasmil.work at gmail.com >> phone: +34669448337 <+34%20669%2044%2083%2037> >> ᐧ >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From infos at madovsky.org Tue Sep 5 21:56:46 2017 From: infos at madovsky.org (Madovsky) Date: Tue, 5 Sep 2017 14:56:46 -0700 Subject: [Freeswitch-users] leg B video Message-ID: Hi all, is there an easy way to know if the leg B has video or not? Thanks Franck From royj at yandex.ru Wed Sep 6 08:25:28 2017 From: royj at yandex.ru (roy j) Date: Wed, 06 Sep 2017 11:25:28 +0300 Subject: [Freeswitch-users] Cisco IP Phones & Counterpath Softphones Crashing w/ mod_av Enabled In-Reply-To: References: Message-ID: <20501504686328@web53g.yandex.ru> An HTML attachment was scrubbed... URL: From npaleari at phonecallsrl.com.ar Tue Sep 5 13:22:35 2017 From: npaleari at phonecallsrl.com.ar (Nicolas Paleari) Date: Tue, 5 Sep 2017 10:22:35 -0300 Subject: [Freeswitch-users] Stop packet radius Message-ID: Friends, I need help, I get incorrect information in the stop packet of raduis that sends freeswitch, in the field dst-number-out the number has %23 instead of #, I do not understand why it replaces it,dst-number-out should be equal to field dst-number-in = 492482 # 541151995330 Send stop package where you see the problem: [2017-09-01 19:03:46,119] DEBUG WORKER 38 - [pkt#175515/ACCT-STOP] --- START: [pkt#175515/ACCT-STOP] -------------------------------------- pdd-time => 0 called-station-id => 492482%23541151995330 nas-port => 0 dst-number-out => 492482%23541151995330 h323-conf-id => cd0f2a0d-4b7f-414c-b5be-d1b937a414db h323-setup-time => 2017-09-01T15:03:36.328927-0400 acct-status-type => Stop h323-disconnect-cause => 10 dst-gw-ip => 190.210.240.37 h323-disconnect-time => 2017-09-01T15:03:43.929039-0400 h323-call-id => 7a34a7be-d643-4f82-a66a-cd1eb0658b6c src-gw-ip => 190.210.240.37 dst-number-in => 492482#541151995330 h323-call-origin => originate nas-ip-address => 107.170.35.75 src-number-in => 7680858053 dst-gw-name => 492482#541151995330 h323-connect-time => 2017-09-01T15:03:38.268917-0400 src-gw-name => 7680858053 acct-session-time => 5 acct-delay-time => 0 src-number-out => 7680858053 calling-station-id => 7680858053 --- END: [pkt#175515/ACCT-STOP] -------------------------------------- I hope you can help me Thank you Libre de virus. www.avast.com <#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2> -------------- next part -------------- An HTML attachment was scrubbed... URL: From gkuri at ieee.org Tue Sep 5 23:24:55 2017 From: gkuri at ieee.org (Gabriel Kuri) Date: Tue, 5 Sep 2017 16:24:55 -0700 Subject: [Freeswitch-users] session timer - enable/disable per call In-Reply-To: References: Message-ID: Two profiles would be an option. It'd be nice if it was configurable per call, but I get it, if it's done before even hitting the dialplan, it doesn't matter much. I'll give the two profiles a try. On Fri, Sep 1, 2017 at 12:08 PM, Steven Ayre wrote: > AFAIK it's negotiated session timers before hitting the dialplan so you > couldn't do that without significant changes to FS. > > Can you run two sofia profiles on different ports with and without timers? > You could get the vendor to use the profile with timers disabled. If they > follow it you might be able to redirect them in dialplan from one profile to > the other on a call-by-call basis. > > On 31 August 2017 at 16:27, Gabriel Kuri wrote: >> >> Is it possible to either: >> >> a) enable/disable the session timer per call, via the dialplan? >> b) change the session timeout value per call, via the dialplan? >> >> Currently, it's set in the sofia profile, either: >> >> >> >> or >> >> >> >> >> However, it would be great if it could be changed/set per call. >> >> If you ask why, a particular device when it receives a re-invite >> mid-call decides to re-invite an already established T.38 call to PCMU >> which breaks the FAX transmission. >> >> Yes, this is a long fax :) >> >> Unfortunately, the vendor won't fix, so it'd be nice for these devices >> to be able to simply disable or set the timer to a really long time in >> the dialplan. >> >> Thanks ... >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From joelists at tm.net.uk Wed Sep 6 09:16:15 2017 From: joelists at tm.net.uk (Joseph Waite) Date: Wed, 6 Sep 2017 10:16:15 +0100 Subject: [Freeswitch-users] Stop packet radius In-Reply-To: References: Message-ID: <814EFC40-F9E1-438B-A2FD-CA8E92984490@tm.net.uk> I'm your radius config are you using the same variable to set both? If not and they will always be the same, may be worth trying changing to use the same. That way can figure out if it's the radius module or freeswitch making the change. Joe Waite > On 5 Sep 2017, at 14:22, Nicolas Paleari wrote: > > Friends, I need help, I get incorrect information in the stop packet of raduis that sends freeswitch, in the field dst-number-out the number has %23 instead of #, I do not understand why it replaces it,dst-number-out should be equal to field dst-number-in = 492482 # 541151995330 > Send stop package where you see the problem: > > > > [2017-09-01 19:03:46,119] DEBUG WORKER 38 - [pkt#175515/ACCT-STOP] > --- START: [pkt#175515/ACCT-STOP] -------------------------------------- > pdd-time => 0 > called-station-id => 492482%23541151995330 > nas-port => 0 > dst-number-out => 492482%23541151995330 > h323-conf-id => cd0f2a0d-4b7f-414c-b5be-d1b937a414db > h323-setup-time => 2017-09-01T15:03:36.328927-0400 > acct-status-type => Stop > h323-disconnect-cause => 10 > dst-gw-ip => 190.210.240.37 > h323-disconnect-time => 2017-09-01T15:03:43.929039-0400 > h323-call-id => 7a34a7be-d643-4f82-a66a-cd1eb0658b6c > src-gw-ip => 190.210.240.37 > dst-number-in => 492482#541151995330 > h323-call-origin => originate > nas-ip-address => 107.170.35.75 > src-number-in => 7680858053 > dst-gw-name => 492482#541151995330 > h323-connect-time => 2017-09-01T15:03:38.268917-0400 > src-gw-name => 7680858053 > acct-session-time => 5 > acct-delay-time => 0 > src-number-out => 7680858053 > calling-station-id => 7680858053 > --- END: [pkt#175515/ACCT-STOP] -------------------------------------- > > I hope you can help me > > Thank you > > Libre de virus. www.avast.com > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From steveayre at gmail.com Wed Sep 6 12:07:32 2017 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 6 Sep 2017 13:07:32 +0100 Subject: [Freeswitch-users] session timer - enable/disable per call In-Reply-To: References: Message-ID: Actually that might not be correct. Some validation will have happened (422 session interval too small) but while the INVITE will have offered session timers but there's nothing in 100 Trying to accept them, that isn't until 18x/200. So it might be possible to enable/disable them, just not play with the accepted range much. On 6 September 2017 at 00:24, Gabriel Kuri wrote: > Two profiles would be an option. It'd be nice if it was configurable > per call, but I get it, if it's done before even hitting the dialplan, > it doesn't matter much. > > I'll give the two profiles a try. > > On Fri, Sep 1, 2017 at 12:08 PM, Steven Ayre wrote: > > AFAIK it's negotiated session timers before hitting the dialplan so you > > couldn't do that without significant changes to FS. > > > > Can you run two sofia profiles on different ports with and without > timers? > > You could get the vendor to use the profile with timers disabled. If they > > follow it you might be able to redirect them in dialplan from one > profile to > > the other on a call-by-call basis. > > > > On 31 August 2017 at 16:27, Gabriel Kuri wrote: > >> > >> Is it possible to either: > >> > >> a) enable/disable the session timer per call, via the dialplan? > >> b) change the session timeout value per call, via the dialplan? > >> > >> Currently, it's set in the sofia profile, either: > >> > >> > >> > >> or > >> > >> > >> > >> > >> However, it would be great if it could be changed/set per call. > >> > >> If you ask why, a particular device when it receives a re-invite > >> mid-call decides to re-invite an already established T.38 call to PCMU > >> which breaks the FAX transmission. > >> > >> Yes, this is a long fax :) > >> > >> Unfortunately, the vendor won't fix, so it'd be nice for these devices > >> to be able to simply disable or set the timer to a really long time in > >> the dialplan. > >> > >> Thanks ... > >> > >> ____________________________________________________________ > _____________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://confluence.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/ > options/freeswitch-users > >> http://www.freeswitch.org > > > > > > > > ____________________________________________________________ > _____________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From colton.conor at gmail.com Wed Sep 6 14:04:19 2017 From: colton.conor at gmail.com (Colton Conor) Date: Wed, 6 Sep 2017 09:04:19 -0500 Subject: [Freeswitch-users] RFC 4662 Support in Freeswitch? In-Reply-To: References: Message-ID: Is this the correct list to ask this type of question? I am trying to determine if Freeswitch supports RFC 4662 for BLF Resource list URI, and if it does not then how to we add it to the wanted features list to get it on developments radar? On Sat, Sep 2, 2017 at 2:12 PM, Colton Conor wrote: > Does Freeswitch support RFC 4662 for BLF Resource list URI? > > Asterisk does: https://wiki.asterisk.org/wiki/pages/viewpage. > action?pageId=30278158 They have a good overview page > > Kamillio does: https://kamailio.org/docs/modules/3.2.x/modules_k/rls.html > > Good info from Grandsteam about why to use this: http://www.grandstream. > com/sites/default/files/Resources/GXP21x0_Eventlist_BLF_Guide.pdf > > Same with Polycom: http://community.polycom.com/polycom/ > attachments/polycom/VoIP/19112/1/Technical%20Brief%20-% > 20Busy%20Lamp%20Field.pdf > > Others have asked this question on the freeswitch users list, but > responses were limited. > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From npaleari at phonecallsrl.com.ar Wed Sep 6 11:03:14 2017 From: npaleari at phonecallsrl.com.ar (Nicolas Paleari) Date: Wed, 6 Sep 2017 08:03:14 -0300 Subject: [Freeswitch-users] Stop packet radius In-Reply-To: <814EFC40-F9E1-438B-A2FD-CA8E92984490@tm.net.uk> References: <814EFC40-F9E1-438B-A2FD-CA8E92984490@tm.net.uk> Message-ID: Hello, change does Freeswitch, I understand why it does a coding, I need to receive #, this is how the carrier calls waiting El 6 sep. 2017 7:08 AM, "Joseph Waite" escribió: > I'm your radius config are you using the same variable to set both? > > If not and they will always be the same, may be worth trying changing to > use the same. That way can figure out if it's the radius module or > freeswitch making the change. > > Joe Waite > > On 5 Sep 2017, at 14:22, Nicolas Paleari > wrote: > > Friends, I need help, I get incorrect information in the stop packet of > raduis that sends freeswitch, in the field dst-number-out the number has > %23 instead of #, I do not understand why it replaces it,dst-number-out should > be equal to field dst-number-in = 492482 # 541151995330 > Send stop package where you see the problem: > > > [2017-09-01 19:03:46,119] DEBUG WORKER 38 - [pkt#175515/ACCT-STOP] > --- START: [pkt#175515/ACCT-STOP] -------------------------------------- > pdd-time => 0 > called-station-id => 492482%23541151995330 > nas-port => 0 > dst-number-out => 492482%23541151995330 > h323-conf-id => cd0f2a0d-4b7f-414c-b5be-d1b937a414db > h323-setup-time => 2017-09-01T15:03:36.328927-0400 > acct-status-type => Stop > h323-disconnect-cause => 10 > dst-gw-ip => 190.210.240.37 > h323-disconnect-time => 2017-09-01T15:03:43.929039-0400 > h323-call-id => 7a34a7be-d643-4f82-a66a-cd1eb0658b6c > src-gw-ip => 190.210.240.37 > dst-number-in => 492482#541151995330 > h323-call-origin => originate > nas-ip-address => 107.170.35.75 > src-number-in => 7680858053 > dst-gw-name => 492482#541151995330 > h323-connect-time => 2017-09-01T15:03:38.268917-0400 > src-gw-name => 7680858053 > acct-session-time => 5 > acct-delay-time => 0 > src-number-out => 7680858053 > calling-station-id => 7680858053 > --- END: [pkt#175515/ACCT-STOP] -------------------------------------- > > I hope you can help me > Thank you > > > Libre > de virus. www.avast.com > > <#m_3420918769942682619_DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From caioebassis at hotmail.com Wed Sep 6 14:26:55 2017 From: caioebassis at hotmail.com (Caio Assis) Date: Wed, 6 Sep 2017 14:26:55 +0000 Subject: [Freeswitch-users] Multiple network interfaces Message-ID: Good morning. On my freeswitch system, I have 2 network interfaces, one with a public IP address and one with a internal IP address. The issue is that when I make a call, freeswitch uses the public address, but I have to use the internal IP address. Only the internal IP address can connect to a specific gateway, but all attempts try to use the public address. With Asterisk I didn't have that issue, for I used the 'bindaddr' param. Can you help me? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: From joelists at tm.net.uk Wed Sep 6 14:58:09 2017 From: joelists at tm.net.uk (Jospeh Waite) Date: Wed, 6 Sep 2017 15:58:09 +0100 Subject: [Freeswitch-users] Multiple network interfaces In-Reply-To: References: Message-ID: <9438E820-1B36-47F1-9963-522961F4852F@tm.net.uk> Depending on your use case, You could create a separate Sofia profile and bind that to your internal address, then in the dial plan specify that profile when placing the call to that specific gateway. Alternatively if you wanted Freeswitch to only use the internal IP for everything you could specify that all your Sofia profiles bind to the internal address. Regards > On 6 Sep 2017, at 15:26, Caio Assis wrote: > > Good morning. > > On my freeswitch system, I have 2 network interfaces, one with a public IP address and one with a internal IP address. The issue is that when I make a call, freeswitch uses the public address, but I have to use the internal IP address. Only the internal IP address can connect to a specific gateway, but all attempts try to use the public address. With Asterisk I didn't have that issue, for I used the 'bindaddr' param. Can you help me? > > Thanks. > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From joelists at tm.net.uk Wed Sep 6 14:59:34 2017 From: joelists at tm.net.uk (Jospeh Waite) Date: Wed, 6 Sep 2017 15:59:34 +0100 Subject: [Freeswitch-users] Stop packet radius In-Reply-To: References: <814EFC40-F9E1-438B-A2FD-CA8E92984490@tm.net.uk> Message-ID: Could you supply a copy of your xml_radius.conf.xml? Might have a clue in there. > On 6 Sep 2017, at 12:03, Nicolas Paleari wrote: > > Hello, change does Freeswitch, I understand why it does a coding, I need to receive #, this is how the carrier calls waiting > > El 6 sep. 2017 7:08 AM, "Joseph Waite" > escribió: > I'm your radius config are you using the same variable to set both? > > If not and they will always be the same, may be worth trying changing to use the same. That way can figure out if it's the radius module or freeswitch making the change. > > Joe Waite > > On 5 Sep 2017, at 14:22, Nicolas Paleari > wrote: > >> Friends, I need help, I get incorrect information in the stop packet of raduis that sends freeswitch, in the field dst-number-out the number has %23 instead of #, I do not understand why it replaces it,dst-number-out should be equal to field dst-number-in = 492482 # 541151995330 >> Send stop package where you see the problem: >> >> >> >> [2017-09-01 19:03:46,119] DEBUG WORKER 38 - [pkt#175515/ACCT-STOP] >> --- START: [pkt#175515/ACCT-STOP] -------------------------------------- >> pdd-time => 0 >> called-station-id => 492482%23541151995330 >> nas-port => 0 >> dst-number-out => 492482%23541151995330 >> h323-conf-id => cd0f2a0d-4b7f-414c-b5be-d1b937a414db >> h323-setup-time => 2017-09-01T15:03:36.328927-0400 >> acct-status-type => Stop >> h323-disconnect-cause => 10 >> dst-gw-ip => 190.210.240.37 >> h323-disconnect-time => 2017-09-01T15:03:43.929039-0400 >> h323-call-id => 7a34a7be-d643-4f82-a66a-cd1eb0658b6c >> src-gw-ip => 190.210.240.37 >> dst-number-in => 492482#541151995330 >> h323-call-origin => originate >> nas-ip-address => 107.170.35.75 >> src-number-in => 7680858053 >> dst-gw-name => 492482#541151995330 >> h323-connect-time => 2017-09-01T15:03:38.268917-0400 >> src-gw-name => 7680858053 >> acct-session-time => 5 >> acct-delay-time => 0 >> src-number-out => 7680858053 >> calling-station-id => 7680858053 >> --- END: [pkt#175515/ACCT-STOP] -------------------------------------- >> >> I hope you can help me >> >> Thank you >> >> Libre de virus. www.avast.com _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From Alexander.Haugg at c4b.de Wed Sep 6 15:17:24 2017 From: Alexander.Haugg at c4b.de (Alexander Haugg) Date: Wed, 6 Sep 2017 15:17:24 +0000 Subject: [Freeswitch-users] Send a Client SIP Register to a PBX or to a provider Message-ID: <14553f61968d4679ac560485ed3b244c@c4b.de> Hi All, is it possible, that the FS receive a client SIP register and send for this a client register to a PBX? Thanks a lot! -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.com Wed Sep 6 15:31:08 2017 From: brian at freeswitch.com (Brian West) Date: Wed, 6 Sep 2017 10:31:08 -0500 Subject: [Freeswitch-users] RFC 4662 Support in Freeswitch? In-Reply-To: References: Message-ID: There is currently no support for this in FreeSWITCH, you can email consulting at freeswitch.com Thanks, On Wed, Sep 6, 2017 at 9:04 AM, Colton Conor wrote: > Is this the correct list to ask this type of question? I am trying to > determine if Freeswitch supports RFC 4662 for BLF Resource list URI, and > if it does not then how to we add it to the wanted features list to get it > on developments radar? > > On Sat, Sep 2, 2017 at 2:12 PM, Colton Conor > wrote: > >> Does Freeswitch support RFC 4662 for BLF Resource list URI? >> >> Asterisk does: https://wiki.asterisk.org/wiki/pages/viewpage.action? >> pageId=30278158 They have a good overview page >> >> Kamillio does: https://kamailio.org/docs/modules/3.2.x/modules_k/rls.html >> >> Good info from Grandsteam about why to use this: http://www.grandstream.c >> om/sites/default/files/Resources/GXP21x0_Eventlist_BLF_Guide.pdf >> >> Same with Polycom: http://community.polycom.com/polycom/attachments/ >> polycom/VoIP/19112/1/Technical%20Brief%20-%20Busy%20Lamp%20Field.pdf >> >> Others have asked this question on the freeswitch users list, but >> responses were limited. >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.com *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) -------------- next part -------------- An HTML attachment was scrubbed... URL: From tayeb.meftah at gmail.com Wed Sep 6 15:35:59 2017 From: tayeb.meftah at gmail.com (Tayeb Meftah) Date: Wed, 6 Sep 2017 16:35:59 +0100 Subject: [Freeswitch-users] Send a Client SIP Register to a PBX or to a provider In-Reply-To: <14553f61968d4679ac560485ed3b244c@c4b.de> References: <14553f61968d4679ac560485ed3b244c@c4b.de> Message-ID: <7EB5FBF2-D7DB-4FDE-A0C7-FA6414FCC4A5@gmail.com> No But you can create a gateway and link it to the user account See the directory examples. Thanks Envoyé de mon iPad > Le 6 sept. 2017 à 16:17, Alexander Haugg a écrit : > > Hi All, > > is it possible, that the FS receive a client SIP register and send for this a client register to a PBX? > > Thanks a lot! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From s.safarov at gmail.com Wed Sep 6 20:11:53 2017 From: s.safarov at gmail.com (Sergey Safarov) Date: Wed, 06 Sep 2017 20:11:53 +0000 Subject: [Freeswitch-users] Timer issue Message-ID: I have FreeSwitch docker container and cannot receive fax. In FreeSwitch log i see error 2017-09-06 20:08:06.963387 [ERR] switch_core_timer.c:117 Timer is not properly configured. But during freeswitch boots up i not see any timer related errors. This error arise during rxfax application. I have tested timer using "timer_test" command and log results like freeswitch at freeswitch1.kazoo> timer_test Avg: 19.999ms Total Time: 999.999ms 2017-09-06 20:09:50.293389 [CONSOLE] mod_commands.c:955 Timer Test: 1 sleep 20 20000 2017-09-06 20:09:50.313385 [CONSOLE] mod_commands.c:955 Timer Test: 2 sleep 20 20003 2017-09-06 20:09:50.333385 [CONSOLE] mod_commands.c:955 Timer Test: 3 sleep 20 20005 2017-09-06 20:09:50.353384 [CONSOLE] mod_commands.c:955 Timer Test: 4 sleep 20 19995 2017-09-06 20:09:50.373387 [CONSOLE] mod_commands.c:955 Timer Test: 5 sleep 20 19998 Could you advice where to look to find root of issue? -------------- next part -------------- An HTML attachment was scrubbed... URL: From borik.internet at gmail.com Thu Sep 7 07:57:34 2017 From: borik.internet at gmail.com (Dmitriy Borisov) Date: Thu, 07 Sep 2017 07:57:34 +0000 Subject: [Freeswitch-users] Stop packet radius In-Reply-To: References: <814EFC40-F9E1-438B-A2FD-CA8E92984490@tm.net.uk> Message-ID: Hi! %23 is a urlencoded variant of # You must fix mod_xml_radius by hand to decode variable in place or request new feature implementation from developers. It's not hard to implement On ср, 6 сент. 2017 г., 18:00 Jospeh Waite wrote: > Could you supply a copy of your xml_radius.conf.xml? > > Might have a clue in there. > > On 6 Sep 2017, at 12:03, Nicolas Paleari > wrote: > > Hello, change does Freeswitch, I understand why it does a coding, I need > to receive #, this is how the carrier calls waiting > > El 6 sep. 2017 7:08 AM, "Joseph Waite" escribió: > >> I'm your radius config are you using the same variable to set both? >> >> If not and they will always be the same, may be worth trying changing to >> use the same. That way can figure out if it's the radius module or >> freeswitch making the change. >> >> Joe Waite >> >> On 5 Sep 2017, at 14:22, Nicolas Paleari >> wrote: >> >> Friends, I need help, I get incorrect information in the stop packet of >> raduis that sends freeswitch, in the field dst-number-out the number has >> %23 instead of #, I do not understand why it replaces it,dst-number-out should >> be equal to field dst-number-in = 492482 # 541151995330 >> Send stop package where you see the problem: >> >> >> [2017-09-01 19:03:46,119] DEBUG WORKER 38 - [pkt#175515/ACCT-STOP] >> --- START: [pkt#175515/ACCT-STOP] -------------------------------------- >> pdd-time => 0 >> called-station-id => 492482%23541151995330 >> nas-port => 0 >> dst-number-out => 492482%23541151995330 >> h323-conf-id => cd0f2a0d-4b7f-414c-b5be-d1b937a414db >> h323-setup-time => 2017-09-01T15:03:36.328927-0400 >> acct-status-type => Stop >> h323-disconnect-cause => 10 >> dst-gw-ip => 190.210.240.37 >> h323-disconnect-time => 2017-09-01T15:03:43.929039-0400 >> h323-call-id => 7a34a7be-d643-4f82-a66a-cd1eb0658b6c >> src-gw-ip => 190.210.240.37 >> dst-number-in => 492482#541151995330 >> h323-call-origin => originate >> nas-ip-address => 107.170.35.75 >> src-number-in => 7680858053 >> dst-gw-name => 492482#541151995330 >> h323-connect-time => 2017-09-01T15:03:38.268917-0400 >> src-gw-name => 7680858053 >> acct-session-time => 5 >> acct-delay-time => 0 >> src-number-out => 7680858053 >> calling-station-id => 7680858053 >> --- END: [pkt#175515/ACCT-STOP] -------------------------------------- >> >> I hope you can help me >> Thank you >> >> >> Libre >> de virus. www.avast.com >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From caioebassis at hotmail.com Wed Sep 6 17:16:30 2017 From: caioebassis at hotmail.com (Caio Assis) Date: Wed, 6 Sep 2017 17:16:30 +0000 Subject: [Freeswitch-users] Multiple network interfaces In-Reply-To: <9438E820-1B36-47F1-9963-522961F4852F@tm.net.uk> References: , <9438E820-1B36-47F1-9963-522961F4852F@tm.net.uk> Message-ID: Could you give me an XML file example? ________________________________ From: FreeSWITCH-users on behalf of Jospeh Waite Sent: Wednesday, September 6, 2017 11:58 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Multiple network interfaces Depending on your use case, You could create a separate Sofia profile and bind that to your internal address, then in the dial plan specify that profile when placing the call to that specific gateway. Alternatively if you wanted Freeswitch to only use the internal IP for everything you could specify that all your Sofia profiles bind to the internal address. Regards On 6 Sep 2017, at 15:26, Caio Assis > wrote: Good morning. On my freeswitch system, I have 2 network interfaces, one with a public IP address and one with a internal IP address. The issue is that when I make a call, freeswitch uses the public address, but I have to use the internal IP address. Only the internal IP address can connect to a specific gateway, but all attempts try to use the public address. With Asterisk I didn't have that issue, for I used the 'bindaddr' param. Can you help me? Thanks. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From npaleari at phonecallsrl.com.ar Wed Sep 6 17:39:54 2017 From: npaleari at phonecallsrl.com.ar (Phonecall SRL - Nicolas Paleari) Date: Wed, 6 Sep 2017 14:39:54 -0300 Subject: [Freeswitch-users] Stop packet radius In-Reply-To: References: <814EFC40-F9E1-438B-A2FD-CA8E92984490@tm.net.uk> Message-ID: <609287ea-c63f-9655-9d4b-a0c238797033@phonecallsrl.com.ar> An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: Firma Nico Phonecall_new.jpg Type: image/jpeg Size: 42608 bytes Desc: not available URL: From gkuri at ieee.org Wed Sep 6 17:58:28 2017 From: gkuri at ieee.org (Gabriel Kuri) Date: Wed, 6 Sep 2017 10:58:28 -0700 Subject: [Freeswitch-users] session timer - enable/disable per call In-Reply-To: References: Message-ID: Hmmm, enable/disable the timers would be helpful. Actually, I would probably end up using the enable/disable more so than setting the timer value. What would it take to implement it? On Wed, Sep 6, 2017 at 5:07 AM, Steven Ayre wrote: > Actually that might not be correct. Some validation will have happened (422 > session interval too small) but while the INVITE will have offered session > timers but there's nothing in 100 Trying to accept them, that isn't until > 18x/200. So it might be possible to enable/disable them, just not play with > the accepted range much. > > On 6 September 2017 at 00:24, Gabriel Kuri wrote: >> >> Two profiles would be an option. It'd be nice if it was configurable >> per call, but I get it, if it's done before even hitting the dialplan, >> it doesn't matter much. >> >> I'll give the two profiles a try. >> >> On Fri, Sep 1, 2017 at 12:08 PM, Steven Ayre wrote: >> > AFAIK it's negotiated session timers before hitting the dialplan so you >> > couldn't do that without significant changes to FS. >> > >> > Can you run two sofia profiles on different ports with and without >> > timers? >> > You could get the vendor to use the profile with timers disabled. If >> > they >> > follow it you might be able to redirect them in dialplan from one >> > profile to >> > the other on a call-by-call basis. >> > >> > On 31 August 2017 at 16:27, Gabriel Kuri wrote: >> >> >> >> Is it possible to either: >> >> >> >> a) enable/disable the session timer per call, via the dialplan? >> >> b) change the session timeout value per call, via the dialplan? >> >> >> >> Currently, it's set in the sofia profile, either: >> >> >> >> >> >> >> >> or >> >> >> >> >> >> >> >> >> >> However, it would be great if it could be changed/set per call. >> >> >> >> If you ask why, a particular device when it receives a re-invite >> >> mid-call decides to re-invite an already established T.38 call to PCMU >> >> which breaks the FAX transmission. >> >> >> >> Yes, this is a long fax :) >> >> >> >> Unfortunately, the vendor won't fix, so it'd be nice for these devices >> >> to be able to simply disable or set the timer to a really long time in >> >> the dialplan. >> >> >> >> Thanks ... >> >> >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://confluence.freeswitch.org >> >> http://www.cluecon.com >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > >> > >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://confluence.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From josedavid at zennio.com Thu Sep 7 08:55:18 2017 From: josedavid at zennio.com (Jose David Jurado Alonso) Date: Thu, 7 Sep 2017 10:55:18 +0200 Subject: [Freeswitch-users] Implement PushKit or CallKit to call iOS app closed Message-ID: Hi, I can't find any information over how to implement PushKit or Callkit to call an iOS mobile phone that has the SIP client application closed or in the background. When the application is closed it is obviously that SIP account isn't registered and the FS server no send the call. Has anyone worked with this or know how to implement it? I tried with "" option and this temporally solved when the application is alive in background. Thanks, -------------- next part -------------- An HTML attachment was scrubbed... URL: From joelists at tm.net.uk Thu Sep 7 10:09:42 2017 From: joelists at tm.net.uk (Joseph Waite) Date: Thu, 7 Sep 2017 11:09:42 +0100 Subject: [Freeswitch-users] Multiple network interfaces In-Reply-To: References: <9438E820-1B36-47F1-9963-522961F4852F@tm.net.uk> Message-ID: <46B8D610-1A05-43B2-874F-49286A4804FD@tm.net.uk> To set the bind ip on the sofia profile you would edit conf/sip_profiles/external.xml or internal.xml depending on which profile, or both if you want both profiles to use the external ip. You then need to find the line and change the value to be the internal IP. You may also need to change the if you need to ensure the media stream comes from the same IP, which is likely in your use case. Regards > On 6 Sep 2017, at 18:16, Caio Assis wrote: > > Could you give me an XML file example? > > > From: FreeSWITCH-users on behalf of Jospeh Waite > Sent: Wednesday, September 6, 2017 11:58 AM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Multiple network interfaces > > Depending on your use case, You could create a separate Sofia profile and bind that to your internal address, then in the dial plan specify that profile when placing the call to that specific gateway. > Alternatively if you wanted Freeswitch to only use the internal IP for everything you could specify that all your Sofia profiles bind to the internal address. > > Regards >> On 6 Sep 2017, at 15:26, Caio Assis > wrote: >> >> Good morning. >> >> On my freeswitch system, I have 2 network interfaces, one with a public IP address and one with a internal IP address. The issue is that when I make a call, freeswitch uses the public address, but I have to use the internal IP address. Only the internal IP address can connect to a specific gateway, but all attempts try to use the public address. With Asterisk I didn't have that issue, for I used the 'bindaddr' param. Can you help me? >> >> Thanks. >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From joelists at tm.net.uk Thu Sep 7 10:27:32 2017 From: joelists at tm.net.uk (Joseph Waite) Date: Thu, 7 Sep 2017 11:27:32 +0100 Subject: [Freeswitch-users] Stop packet radius In-Reply-To: <609287ea-c63f-9655-9d4b-a0c238797033@phonecallsrl.com.ar> References: <814EFC40-F9E1-438B-A2FD-CA8E92984490@tm.net.uk> <609287ea-c63f-9655-9d4b-a0c238797033@phonecallsrl.com.ar> Message-ID: <6888FB36-04AC-43A2-8077-151BB1925244@tm.net.uk> Ok Firstly, you should always redact and sensitive parts of config files before posting to a public user group. And Hacker now has the access credential for your radius server, I would advise changing the password immediately. This line should have had the IP and password blanked out to look like Hope this helps. Regards > On 6 Sep 2017, at 18:39, Phonecall SRL - Nicolas Paleari wrote: > > Hi, copy the contents of the file: > >  > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > Best regards > > El 6/9/2017 a las 11:59 a. m., Jospeh Waite escribió: >> Could you supply a copy of your xml_radius.conf.xml? >> >> Might have a clue in there. >>> On 6 Sep 2017, at 12:03, Nicolas Paleari > wrote: >>> >>> Hello, change does Freeswitch, I understand why it does a coding, I need to receive #, this is how the carrier calls waiting >>> >>> El 6 sep. 2017 7:08 AM, "Joseph Waite" > escribió: >>> I'm your radius config are you using the same variable to set both? >>> >>> If not and they will always be the same, may be worth trying changing to use the same. That way can figure out if it's the radius module or freeswitch making the change. >>> >>> Joe Waite >>> >>> On 5 Sep 2017, at 14:22, Nicolas Paleari > wrote: >>> >>>> Friends, I need help, I get incorrect information in the stop packet of raduis that sends freeswitch, in the field dst-number-out the number has %23 instead of #, I do not understand why it replaces it,dst-number-out should be equal to field dst-number-in = 492482 # 541151995330 >>>> Send stop package where you see the problem: >>>> >>>> [2017-09-01 19:03:46,119] DEBUG WORKER 38 - [pkt#175515/ACCT-STOP] >>>> --- START: [pkt#175515/ACCT-STOP] -------------------------------------- >>>> pdd-time => 0 >>>> called-station-id => 492482%23541151995330 >>>> nas-port => 0 >>>> dst-number-out => 492482%23541151995330 >>>> h323-conf-id => cd0f2a0d-4b7f-414c-b5be-d1b937a414db >>>> h323-setup-time => 2017-09-01T15:03:36.328927-0400 >>>> acct-status-type => Stop >>>> h323-disconnect-cause => 10 >>>> dst-gw-ip => 190.210.240.37 >>>> h323-disconnect-time => 2017-09-01T15:03:43.929039-0400 >>>> h323-call-id => 7a34a7be-d643-4f82-a66a-cd1eb0658b6c >>>> src-gw-ip => 190.210.240.37 >>>> dst-number-in => 492482#541151995330 >>>> h323-call-origin => originate >>>> nas-ip-address => 107.170.35.75 >>>> src-number-in => 7680858053 >>>> dst-gw-name => 492482#541151995330 >>>> h323-connect-time => 2017-09-01T15:03:38.268917-0400 >>>> src-gw-name => 7680858053 >>>> acct-session-time => 5 >>>> acct-delay-time => 0 >>>> src-number-out => 7680858053 >>>> calling-station-id => 7680858053 >>>> --- END: [pkt#175515/ACCT-STOP] -------------------------------------- >>>> >>>> I hope you can help me >>>> >>>> Thank you >>>> >>>> Libre de virus. www.avast.com _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From tg-maillistings at level5.de Thu Sep 7 11:19:48 2017 From: tg-maillistings at level5.de (=?UTF-8?Q?Thorsten_G=c3=b6llner?=) Date: Thu, 7 Sep 2017 13:19:48 +0200 Subject: [Freeswitch-users] SIP-Client / IOS / PJSIP / Error when CANCEL Message-ID: Hi, currently we are developing a SIP-Client for iOS (using PJSIP/PJSUA with the current version). We use Freeswitch 1.6.18 (64 bit) on a bare metal server. Windows-Clients (such as Blink or Zoiper) work fine. And our iOS-App works fine also. BUT when we initialze an outgoing call, wait for ringing on the other hand and hangup up when it's ringing, we face an error. Freeswitch gives us "481 Call/Transaction Does Not Exist". I attached the corresponding SIP-Trace and you can see the error at the last block. The interesting thing: If we activate PJSIP with TCP and(!) UDP we face the error. BUT when we use only UDP it works fine. Is it a bug within PJSIP? Or anyone see an hint wihtin the SIP-Trace? Thanks in advance! recv 1247 bytes from udp/[77.119.248.251]:5060 at 11:36:21.330337:    ------------------------------------------------------------------------    INVITE sip:+436641540180 at telequest-occ.0049.org SIP/2.0    Via: SIP/2.0/UDP 77.119.248.251:5060;branch=z9hG4bK4b69df210aa918eb70770f8f50681c51    Via: SIP/2.0/UDP 192.168.178.110:5060;rport;branch=z9hG4bKPjBP3AvciR4ifGE8V3KBSuxInUNOQN8VLh    From: ;tag=DDna03R3v1t0hJut1phv2uhb4OxBMYnu    To:    Call-ID: VQlDTx2IUOlzd5QZuZHdLoqbyOGtmO1Z    CSeq: 17632 INVITE    Contact:    max-forwards: 69    supported: replaces    supported: 100rel    supported: timer    supported: norefersub    session-expires: 1800    min-se: 90    user-agent: AdCall    Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS    Content-Type: application/sdp    Content-Length:   480    v=0    o=- 3713765781 3713765781 IN IP4 77.119.248.251    s=pjmedia    b=AS:84    t=0 0    a=X-nat:0    m=audio 7070 RTP/AVP 98 97 99 104 3 0 8 9 96    c=IN IP4 77.119.248.251    b=TIAS:64000    a=rtcp:4001 IN IP4 192.168.178.110    a=sendrecv    a=rtpmap:98 speex/16000    a=rtpmap:97 speex/8000    a=rtpmap:99 speex/32000    a=rtpmap:104 iLBC/8000    a=fmtp:104 mode=30    a=rtpmap:3 GSM/8000    a=rtpmap:0 PCMU/8000    a=rtpmap:8 PCMA/8000    a=rtpmap:9 G722/8000    a=rtpmap:96 telephone-event/8000    a=fmtp:96 0-16    ------------------------------------------------------------------------ send 432 bytes to udp/[77.119.248.251]:5060 at 11:36:21.330679:    ------------------------------------------------------------------------    SIP/2.0 100 Trying    Via: SIP/2.0/UDP 77.119.248.251:5060;branch=z9hG4bK4b69df210aa918eb70770f8f50681c51    Via: SIP/2.0/UDP 192.168.178.110:5060;rport;branch=z9hG4bKPjBP3AvciR4ifGE8V3KBSuxInUNOQN8VLh    From: ;tag=DDna03R3v1t0hJut1phv2uhb4OxBMYnu    To:    Call-ID: VQlDTx2IUOlzd5QZuZHdLoqbyOGtmO1Z    CSeq: 17632 INVITE    User-Agent: FreeSWITCH    Content-Length: 0    ------------------------------------------------------------------------ send 945 bytes to udp/[77.119.248.251]:5060 at 11:36:21.331652:    ------------------------------------------------------------------------    SIP/2.0 407 Proxy Authentication Required    Via: SIP/2.0/UDP 77.119.248.251:5060;branch=z9hG4bK4b69df210aa918eb70770f8f50681c51    Via: SIP/2.0/UDP 192.168.178.110:5060;rport;branch=z9hG4bKPjBP3AvciR4ifGE8V3KBSuxInUNOQN8VLh    From: ;tag=DDna03R3v1t0hJut1phv2uhb4OxBMYnu    To: ;tag=0FgQ57apej24p    Call-ID: VQlDTx2IUOlzd5QZuZHdLoqbyOGtmO1Z    CSeq: 17632 INVITE    User-Agent: FreeSWITCH    Accept: application/sdp    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE    Supported: timer, path, replaces    Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer    Proxy-Authenticate: Digest realm="telequest-occ.0049.org", nonce="01e39fee-93b0-11e7-9b0b-45e8f1c3918a", algorithm=MD5, qop="auth"    Content-Length: 0    ------------------------------------------------------------------------ recv 475 bytes from udp/[77.119.248.251]:5060 at 11:36:21.409776:    ------------------------------------------------------------------------    ACK sip:+436641540180 at telequest-occ.0049.org SIP/2.0    Via: SIP/2.0/UDP 77.119.248.251:5060;branch=z9hG4bK4b69df210aa918eb70770f8f50681c51    Via: SIP/2.0/UDP 192.168.178.110:5060;rport;branch=z9hG4bKPjBP3AvciR4ifGE8V3KBSuxInUNOQN8VLh    From: ;tag=DDna03R3v1t0hJut1phv2uhb4OxBMYnu    To: ;tag=0FgQ57apej24p    Call-ID: VQlDTx2IUOlzd5QZuZHdLoqbyOGtmO1Z    CSeq: 17632 ACK    max-forwards: 69    Content-Length: 0    ------------------------------------------------------------------------ recv 1439 bytes from tcp/[77.119.248.251]:60186 at 11:36:21.457742:    ------------------------------------------------------------------------    INVITE sip:+436641540180 at telequest-occ.0049.org SIP/2.0    Via: SIP/2.0/TCP 192.168.178.110:60186;rport;branch=z9hG4bKPj4mpXCeY7uNSlFOUvRXIvfftWSq9PQE-7;alias    Max-Forwards: 70    From: sip:102 at telequest-occ.0049.org;tag=DDna03R3v1t0hJut1phv2uhb4OxBMYnu    To: sip:+436641540180 at telequest-occ.0049.org    Contact:    Call-ID: VQlDTx2IUOlzd5QZuZHdLoqbyOGtmO1Z    CSeq: 17633 INVITE    Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS    Supported: replaces, 100rel, timer, norefersub    Session-Expires: 1800    Min-SE: 90    User-Agent: AdCall    Proxy-Authorization: Digest username="102", realm="telequest-occ.0049.org", nonce="01e39fee-93b0-11e7-9b0b-45e8f1c3918a", uri="sip:+436641540180 at telequest-occ.0049.org", response="c7fef167bd3ecc40bdaeaaef9c9b8085", algorithm=MD5, cnonce="raNBupUIEw5CyLL8MQbvYrk5txEd3Zep", qop=auth, nc=00000001    Content-Type: application/sdp    Content-Length:   482    v=0    o=- 3713765781 3713765781 IN IP4 192.168.178.110    s=pjmedia    b=AS:84    t=0 0    a=X-nat:0    m=audio 4000 RTP/AVP 98 97 99 104 3 0 8 9 96    c=IN IP4 192.168.178.110    b=TIAS:64000    a=rtcp:4001 IN IP4 192.168.178.110    a=sendrecv    a=rtpmap:98 speex/16000    a=rtpmap:97 speex/8000    a=rtpmap:99 speex/32000    a=rtpmap:104 iLBC/8000    a=fmtp:104 mode=30    a=rtpmap:3 GSM/8000    a=rtpmap:0 PCMU/8000    a=rtpmap:8 PCMA/8000    a=rtpmap:9 G722/8000    a=rtpmap:96 telephone-event/8000    a=fmtp:96 0-16    ------------------------------------------------------------------------ send 380 bytes to tcp/[77.119.248.251]:60186 at 11:36:21.457978:    ------------------------------------------------------------------------    SIP/2.0 100 Trying    Via: SIP/2.0/TCP 192.168.178.110:60186;rport=60186;branch=z9hG4bKPj4mpXCeY7uNSlFOUvRXIvfftWSq9PQE-7;alias;received=77.119.248.251    From: sip:102 at telequest-occ.0049.org;tag=DDna03R3v1t0hJut1phv2uhb4OxBMYnu    To: sip:+436641540180 at telequest-occ.0049.org    Call-ID: VQlDTx2IUOlzd5QZuZHdLoqbyOGtmO1Z    CSeq: 17633 INVITE    User-Agent: FreeSWITCH    Content-Length: 0    ------------------------------------------------------------------------ send 1193 bytes to tcp/[77.119.248.251]:60186 at 11:36:24.966185:    ------------------------------------------------------------------------    SIP/2.0 183 Session Progress    Via: SIP/2.0/TCP 192.168.178.110:60186;rport=60186;branch=z9hG4bKPj4mpXCeY7uNSlFOUvRXIvfftWSq9PQE-7;alias;received=77.119.248.251    From: sip:102 at telequest-occ.0049.org;tag=DDna03R3v1t0hJut1phv2uhb4OxBMYnu    To: ;tag=1r9F72USBUrQj    Call-ID: VQlDTx2IUOlzd5QZuZHdLoqbyOGtmO1Z    CSeq: 17633 INVITE    Contact:    User-Agent: FreeSWITCH    Accept: application/sdp    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE    Supported: timer, path, replaces    Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer    Content-Type: application/sdp    Content-Disposition: session    Content-Length: 203    Remote-Party-ID: "+436641540180" ;party=calling;privacy=off;screen=no    v=0    o=FreeSWITCH 1504753876 1504753877 IN IP4 88.217.132.183    s=FreeSWITCH    c=IN IP4 88.217.132.183    t=0 0    m=audio 23108 RTP/AVP 3    a=rtpmap:3 GSM/8000    a=ptime:20    a=rtcp:23109 IN IP4 88.217.132.183    ------------------------------------------------------------------------ recv 401 bytes from tcp/[77.119.248.251]:60186 at 11:36:28.869708:    ------------------------------------------------------------------------    CANCEL sip:+436641540180 at telequest-occ.0049.org SIP/2.0    Via: SIP/2.0/TCP 192.168.178.110:5060;rport;branch=z9hG4bKPj4mpXCeY7uNSlFOUvRXIvfftWSq9PQE-7;alias    Max-Forwards: 70    From: sip:102 at telequest-occ.0049.org;tag=DDna03R3v1t0hJut1phv2uhb4OxBMYnu    To: sip:+436641540180 at telequest-occ.0049.org    Call-ID: VQlDTx2IUOlzd5QZuZHdLoqbyOGtmO1Z    CSeq: 17633 CANCEL    User-Agent: AdCall    Content-Length:  0    ------------------------------------------------------------------------ send 400 bytes to tcp/[77.119.248.251]:60186 at 11:36:28.869807:    ------------------------------------------------------------------------    SIP/2.0 481 Call/Transaction Does Not Exist    Via: SIP/2.0/TCP 192.168.178.110:5060;rport=60186;branch=z9hG4bKPj4mpXCeY7uNSlFOUvRXIvfftWSq9PQE-7;alias;received=77.119.248.251    From: sip:102 at telequest-occ.0049.org;tag=DDna03R3v1t0hJut1phv2uhb4OxBMYnu    To: ;tag=1r9F72USBUrQj    Call-ID: VQlDTx2IUOlzd5QZuZHdLoqbyOGtmO1Z    CSeq: 17633 CANCEL    Content-Length: 0    ------------------------------------------------------------------------ Best regards, Thorsten From lists at telefaks.de Thu Sep 7 11:40:25 2017 From: lists at telefaks.de (Peter Steinbach) Date: Thu, 07 Sep 2017 13:40:25 +0200 Subject: [Freeswitch-users] SIP-Client / IOS / PJSIP / Error when CANCEL In-Reply-To: References: Message-ID: <59B13029.7060908@telefaks.de> Hello Thorsten this is often caused by changed Contact: header in your SIP dialog. In your case the 2 Invites have: Contact: Contact: So this will not work. Best regards from Germany Peter On 09/07/17 13:19, Thorsten Göllner wrote: > Hi, > > currently we are developing a SIP-Client for iOS (using PJSIP/PJSUA with > the current version). We use Freeswitch 1.6.18 (64 bit) on a bare metal > server. Windows-Clients (such as Blink or Zoiper) work fine. And our > iOS-App works fine also. BUT when we initialze an outgoing call, wait > for ringing on the other hand and hangup up when it's ringing, we face > an error. Freeswitch gives us "481 Call/Transaction Does Not Exist". > > I attached the corresponding SIP-Trace and you can see the error at the > last block. > > The interesting thing: If we activate PJSIP with TCP and(!) UDP we face > the error. BUT when we use only UDP it works fine. > > Is it a bug within PJSIP? Or anyone see an hint wihtin the SIP-Trace? > > Thanks in advance! > > recv 1247 bytes from udp/[77.119.248.251]:5060 at 11:36:21.330337: > ------------------------------------------------------------------------ > INVITE sip:+436641540180 at telequest-occ.0049.org SIP/2.0 > Via: SIP/2.0/UDP > 77.119.248.251:5060;branch=z9hG4bK4b69df210aa918eb70770f8f50681c51 > Via: SIP/2.0/UDP > 192.168.178.110:5060;rport;branch=z9hG4bKPjBP3AvciR4ifGE8V3KBSuxInUNOQN8VLh > From: > ;tag=DDna03R3v1t0hJut1phv2uhb4OxBMYnu > To: > Call-ID: VQlDTx2IUOlzd5QZuZHdLoqbyOGtmO1Z > CSeq: 17632 INVITE > Contact: > max-forwards: 69 > supported: replaces > supported: 100rel > supported: timer > supported: norefersub > session-expires: 1800 > min-se: 90 > user-agent: AdCall > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, > NOTIFY, REFER, MESSAGE, OPTIONS > Content-Type: application/sdp > Content-Length: 480 > > v=0 > o=- 3713765781 3713765781 IN IP4 77.119.248.251 > s=pjmedia > b=AS:84 > t=0 0 > a=X-nat:0 > m=audio 7070 RTP/AVP 98 97 99 104 3 0 8 9 96 > c=IN IP4 77.119.248.251 > b=TIAS:64000 > a=rtcp:4001 IN IP4 192.168.178.110 > a=sendrecv > a=rtpmap:98 speex/16000 > a=rtpmap:97 speex/8000 > a=rtpmap:99 speex/32000 > a=rtpmap:104 iLBC/8000 > a=fmtp:104 mode=30 > a=rtpmap:3 GSM/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:9 G722/8000 > a=rtpmap:96 telephone-event/8000 > a=fmtp:96 0-16 > ------------------------------------------------------------------------ > send 432 bytes to udp/[77.119.248.251]:5060 at 11:36:21.330679: > ------------------------------------------------------------------------ > SIP/2.0 100 Trying > Via: SIP/2.0/UDP > 77.119.248.251:5060;branch=z9hG4bK4b69df210aa918eb70770f8f50681c51 > Via: SIP/2.0/UDP > 192.168.178.110:5060;rport;branch=z9hG4bKPjBP3AvciR4ifGE8V3KBSuxInUNOQN8VLh > From: > ;tag=DDna03R3v1t0hJut1phv2uhb4OxBMYnu > To: > Call-ID: VQlDTx2IUOlzd5QZuZHdLoqbyOGtmO1Z > CSeq: 17632 INVITE > User-Agent: FreeSWITCH > Content-Length: 0 > > ------------------------------------------------------------------------ > send 945 bytes to udp/[77.119.248.251]:5060 at 11:36:21.331652: > ------------------------------------------------------------------------ > SIP/2.0 407 Proxy Authentication Required > Via: SIP/2.0/UDP > 77.119.248.251:5060;branch=z9hG4bK4b69df210aa918eb70770f8f50681c51 > Via: SIP/2.0/UDP > 192.168.178.110:5060;rport;branch=z9hG4bKPjBP3AvciR4ifGE8V3KBSuxInUNOQN8VLh > From: > ;tag=DDna03R3v1t0hJut1phv2uhb4OxBMYnu > To: ;tag=0FgQ57apej24p > Call-ID: VQlDTx2IUOlzd5QZuZHdLoqbyOGtmO1Z > CSeq: 17632 INVITE > User-Agent: FreeSWITCH > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, path, replaces > Allow-Events: talk, hold, conference, presence, as-feature-event, > dialog, line-seize, call-info, sla, include-session-description, > presence.winfo, message-summary, refer > Proxy-Authenticate: Digest realm="telequest-occ.0049.org", > nonce="01e39fee-93b0-11e7-9b0b-45e8f1c3918a", algorithm=MD5, qop="auth" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 475 bytes from udp/[77.119.248.251]:5060 at 11:36:21.409776: > ------------------------------------------------------------------------ > ACK sip:+436641540180 at telequest-occ.0049.org SIP/2.0 > Via: SIP/2.0/UDP > 77.119.248.251:5060;branch=z9hG4bK4b69df210aa918eb70770f8f50681c51 > Via: SIP/2.0/UDP > 192.168.178.110:5060;rport;branch=z9hG4bKPjBP3AvciR4ifGE8V3KBSuxInUNOQN8VLh > From: > ;tag=DDna03R3v1t0hJut1phv2uhb4OxBMYnu > To: ;tag=0FgQ57apej24p > Call-ID: VQlDTx2IUOlzd5QZuZHdLoqbyOGtmO1Z > CSeq: 17632 ACK > max-forwards: 69 > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 1439 bytes from tcp/[77.119.248.251]:60186 at 11:36:21.457742: > ------------------------------------------------------------------------ > INVITE sip:+436641540180 at telequest-occ.0049.org SIP/2.0 > Via: SIP/2.0/TCP > 192.168.178.110:60186;rport;branch=z9hG4bKPj4mpXCeY7uNSlFOUvRXIvfftWSq9PQE-7;alias > Max-Forwards: 70 > From: sip:102 at telequest-occ.0049.org;tag=DDna03R3v1t0hJut1phv2uhb4OxBMYnu > To: sip:+436641540180 at telequest-occ.0049.org > Contact: > Call-ID: VQlDTx2IUOlzd5QZuZHdLoqbyOGtmO1Z > CSeq: 17633 INVITE > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, > NOTIFY, REFER, MESSAGE, OPTIONS > Supported: replaces, 100rel, timer, norefersub > Session-Expires: 1800 > Min-SE: 90 > User-Agent: AdCall > Proxy-Authorization: Digest username="102", > realm="telequest-occ.0049.org", > nonce="01e39fee-93b0-11e7-9b0b-45e8f1c3918a", > uri="sip:+436641540180 at telequest-occ.0049.org", > response="c7fef167bd3ecc40bdaeaaef9c9b8085", algorithm=MD5, > cnonce="raNBupUIEw5CyLL8MQbvYrk5txEd3Zep", qop=auth, nc=00000001 > Content-Type: application/sdp > Content-Length: 482 > > v=0 > o=- 3713765781 3713765781 IN IP4 192.168.178.110 > s=pjmedia > b=AS:84 > t=0 0 > a=X-nat:0 > m=audio 4000 RTP/AVP 98 97 99 104 3 0 8 9 96 > c=IN IP4 192.168.178.110 > b=TIAS:64000 > a=rtcp:4001 IN IP4 192.168.178.110 > a=sendrecv > a=rtpmap:98 speex/16000 > a=rtpmap:97 speex/8000 > a=rtpmap:99 speex/32000 > a=rtpmap:104 iLBC/8000 > a=fmtp:104 mode=30 > a=rtpmap:3 GSM/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:9 G722/8000 > a=rtpmap:96 telephone-event/8000 > a=fmtp:96 0-16 > ------------------------------------------------------------------------ > send 380 bytes to tcp/[77.119.248.251]:60186 at 11:36:21.457978: > ------------------------------------------------------------------------ > SIP/2.0 100 Trying > Via: SIP/2.0/TCP > 192.168.178.110:60186;rport=60186;branch=z9hG4bKPj4mpXCeY7uNSlFOUvRXIvfftWSq9PQE-7;alias;received=77.119.248.251 > From: sip:102 at telequest-occ.0049.org;tag=DDna03R3v1t0hJut1phv2uhb4OxBMYnu > To: sip:+436641540180 at telequest-occ.0049.org > Call-ID: VQlDTx2IUOlzd5QZuZHdLoqbyOGtmO1Z > CSeq: 17633 INVITE > User-Agent: FreeSWITCH > Content-Length: 0 > > ------------------------------------------------------------------------ > send 1193 bytes to tcp/[77.119.248.251]:60186 at 11:36:24.966185: > ------------------------------------------------------------------------ > SIP/2.0 183 Session Progress > Via: SIP/2.0/TCP > 192.168.178.110:60186;rport=60186;branch=z9hG4bKPj4mpXCeY7uNSlFOUvRXIvfftWSq9PQE-7;alias;received=77.119.248.251 > From: sip:102 at telequest-occ.0049.org;tag=DDna03R3v1t0hJut1phv2uhb4OxBMYnu > To: ;tag=1r9F72USBUrQj > Call-ID: VQlDTx2IUOlzd5QZuZHdLoqbyOGtmO1Z > CSeq: 17633 INVITE > Contact: > User-Agent: FreeSWITCH > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, path, replaces > Allow-Events: talk, hold, conference, presence, as-feature-event, > dialog, line-seize, call-info, sla, include-session-description, > presence.winfo, message-summary, refer > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 203 > Remote-Party-ID: "+436641540180" > ;party=calling;privacy=off;screen=no > > v=0 > o=FreeSWITCH 1504753876 1504753877 IN IP4 88.217.132.183 > s=FreeSWITCH > c=IN IP4 88.217.132.183 > t=0 0 > m=audio 23108 RTP/AVP 3 > a=rtpmap:3 GSM/8000 > a=ptime:20 > a=rtcp:23109 IN IP4 88.217.132.183 > ------------------------------------------------------------------------ > recv 401 bytes from tcp/[77.119.248.251]:60186 at 11:36:28.869708: > ------------------------------------------------------------------------ > CANCEL sip:+436641540180 at telequest-occ.0049.org SIP/2.0 > Via: SIP/2.0/TCP > 192.168.178.110:5060;rport;branch=z9hG4bKPj4mpXCeY7uNSlFOUvRXIvfftWSq9PQE-7;alias > Max-Forwards: 70 > From: sip:102 at telequest-occ.0049.org;tag=DDna03R3v1t0hJut1phv2uhb4OxBMYnu > To: sip:+436641540180 at telequest-occ.0049.org > Call-ID: VQlDTx2IUOlzd5QZuZHdLoqbyOGtmO1Z > CSeq: 17633 CANCEL > User-Agent: AdCall > Content-Length: 0 > > ------------------------------------------------------------------------ > send 400 bytes to tcp/[77.119.248.251]:60186 at 11:36:28.869807: > ------------------------------------------------------------------------ > SIP/2.0 481 Call/Transaction Does Not Exist > Via: SIP/2.0/TCP > 192.168.178.110:5060;rport=60186;branch=z9hG4bKPj4mpXCeY7uNSlFOUvRXIvfftWSq9PQE-7;alias;received=77.119.248.251 > From: sip:102 at telequest-occ.0049.org;tag=DDna03R3v1t0hJut1phv2uhb4OxBMYnu > To: ;tag=1r9F72USBUrQj > Call-ID: VQlDTx2IUOlzd5QZuZHdLoqbyOGtmO1Z > CSeq: 17633 CANCEL > Content-Length: 0 > > ------------------------------------------------------------------------ > > Best regards, > Thorsten > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de From tg-maillistings at level5.de Thu Sep 7 13:43:14 2017 From: tg-maillistings at level5.de (=?UTF-8?Q?Thorsten_G=c3=b6llner?=) Date: Thu, 7 Sep 2017 15:43:14 +0200 Subject: [Freeswitch-users] SIP-Client / IOS / PJSIP / Error when CANCEL In-Reply-To: <59B13029.7060908@telefaks.de> References: <59B13029.7060908@telefaks.de> Message-ID: Hi Peter, thanks so far. But why is the Contact: header changed (by pjsip?) when switching from TCP/UDP to UDP-only? I am not sure for now but I don't think, that our developer changed the Contact: header explicitly. Best regards, Thorsten Am 07.09.2017 um 13:40 schrieb Peter Steinbach: > Hello Thorsten > > this is often caused by changed Contact: header in your SIP dialog. In > your case the 2 Invites have: > Contact: > Contact: > > So this will not work. > > Best regards from Germany > Peter > > > On 09/07/17 13:19, Thorsten Göllner wrote: >> Hi, >> >> currently we are developing a SIP-Client for iOS (using PJSIP/PJSUA with >> the current version). We use Freeswitch 1.6.18 (64 bit) on a bare metal >> server. Windows-Clients (such as Blink or Zoiper) work fine. And our >> iOS-App works fine also. BUT when we initialze an outgoing call, wait >> for ringing on the other hand and hangup up when it's ringing, we face >> an error. Freeswitch gives us "481 Call/Transaction Does Not Exist". >> >> I attached the corresponding SIP-Trace and you can see the error at the >> last block. >> >> The interesting thing: If we activate PJSIP with TCP and(!) UDP we face >> the error. BUT when we use only UDP it works fine. >> >> Is it a bug within PJSIP? Or anyone see an hint wihtin the SIP-Trace? >> >> Thanks in advance! >> >> recv 1247 bytes from udp/[77.119.248.251]:5060 at 11:36:21.330337: >> ------------------------------------------------------------------------ >> INVITE sip:+436641540180 at telequest-occ.0049.org SIP/2.0 >> Via: SIP/2.0/UDP >> 77.119.248.251:5060;branch=z9hG4bK4b69df210aa918eb70770f8f50681c51 >> Via: SIP/2.0/UDP >> 192.168.178.110:5060;rport;branch=z9hG4bKPjBP3AvciR4ifGE8V3KBSuxInUNOQN8VLh >> From: >> ;tag=DDna03R3v1t0hJut1phv2uhb4OxBMYnu >> To: >> Call-ID: VQlDTx2IUOlzd5QZuZHdLoqbyOGtmO1Z >> CSeq: 17632 INVITE >> Contact: >> max-forwards: 69 >> supported: replaces >> supported: 100rel >> supported: timer >> supported: norefersub >> session-expires: 1800 >> min-se: 90 >> user-agent: AdCall >> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, >> NOTIFY, REFER, MESSAGE, OPTIONS >> Content-Type: application/sdp >> Content-Length: 480 >> >> v=0 >> o=- 3713765781 3713765781 IN IP4 77.119.248.251 >> s=pjmedia >> b=AS:84 >> t=0 0 >> a=X-nat:0 >> m=audio 7070 RTP/AVP 98 97 99 104 3 0 8 9 96 >> c=IN IP4 77.119.248.251 >> b=TIAS:64000 >> a=rtcp:4001 IN IP4 192.168.178.110 >> a=sendrecv >> a=rtpmap:98 speex/16000 >> a=rtpmap:97 speex/8000 >> a=rtpmap:99 speex/32000 >> a=rtpmap:104 iLBC/8000 >> a=fmtp:104 mode=30 >> a=rtpmap:3 GSM/8000 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:9 G722/8000 >> a=rtpmap:96 telephone-event/8000 >> a=fmtp:96 0-16 >> ------------------------------------------------------------------------ >> send 432 bytes to udp/[77.119.248.251]:5060 at 11:36:21.330679: >> ------------------------------------------------------------------------ >> SIP/2.0 100 Trying >> Via: SIP/2.0/UDP >> 77.119.248.251:5060;branch=z9hG4bK4b69df210aa918eb70770f8f50681c51 >> Via: SIP/2.0/UDP >> 192.168.178.110:5060;rport;branch=z9hG4bKPjBP3AvciR4ifGE8V3KBSuxInUNOQN8VLh >> From: >> ;tag=DDna03R3v1t0hJut1phv2uhb4OxBMYnu >> To: >> Call-ID: VQlDTx2IUOlzd5QZuZHdLoqbyOGtmO1Z >> CSeq: 17632 INVITE >> User-Agent: FreeSWITCH >> Content-Length: 0 >> >> ------------------------------------------------------------------------ >> send 945 bytes to udp/[77.119.248.251]:5060 at 11:36:21.331652: >> ------------------------------------------------------------------------ >> SIP/2.0 407 Proxy Authentication Required >> Via: SIP/2.0/UDP >> 77.119.248.251:5060;branch=z9hG4bK4b69df210aa918eb70770f8f50681c51 >> Via: SIP/2.0/UDP >> 192.168.178.110:5060;rport;branch=z9hG4bKPjBP3AvciR4ifGE8V3KBSuxInUNOQN8VLh >> From: >> ;tag=DDna03R3v1t0hJut1phv2uhb4OxBMYnu >> To: ;tag=0FgQ57apej24p >> Call-ID: VQlDTx2IUOlzd5QZuZHdLoqbyOGtmO1Z >> CSeq: 17632 INVITE >> User-Agent: FreeSWITCH >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, path, replaces >> Allow-Events: talk, hold, conference, presence, as-feature-event, >> dialog, line-seize, call-info, sla, include-session-description, >> presence.winfo, message-summary, refer >> Proxy-Authenticate: Digest realm="telequest-occ.0049.org", >> nonce="01e39fee-93b0-11e7-9b0b-45e8f1c3918a", algorithm=MD5, qop="auth" >> Content-Length: 0 >> >> ------------------------------------------------------------------------ >> recv 475 bytes from udp/[77.119.248.251]:5060 at 11:36:21.409776: >> ------------------------------------------------------------------------ >> ACK sip:+436641540180 at telequest-occ.0049.org SIP/2.0 >> Via: SIP/2.0/UDP >> 77.119.248.251:5060;branch=z9hG4bK4b69df210aa918eb70770f8f50681c51 >> Via: SIP/2.0/UDP >> 192.168.178.110:5060;rport;branch=z9hG4bKPjBP3AvciR4ifGE8V3KBSuxInUNOQN8VLh >> From: >> ;tag=DDna03R3v1t0hJut1phv2uhb4OxBMYnu >> To: ;tag=0FgQ57apej24p >> Call-ID: VQlDTx2IUOlzd5QZuZHdLoqbyOGtmO1Z >> CSeq: 17632 ACK >> max-forwards: 69 >> Content-Length: 0 >> >> ------------------------------------------------------------------------ >> recv 1439 bytes from tcp/[77.119.248.251]:60186 at 11:36:21.457742: >> ------------------------------------------------------------------------ >> INVITE sip:+436641540180 at telequest-occ.0049.org SIP/2.0 >> Via: SIP/2.0/TCP >> 192.168.178.110:60186;rport;branch=z9hG4bKPj4mpXCeY7uNSlFOUvRXIvfftWSq9PQE-7;alias >> Max-Forwards: 70 >> From: sip:102 at telequest-occ.0049.org;tag=DDna03R3v1t0hJut1phv2uhb4OxBMYnu >> To: sip:+436641540180 at telequest-occ.0049.org >> Contact: >> Call-ID: VQlDTx2IUOlzd5QZuZHdLoqbyOGtmO1Z >> CSeq: 17633 INVITE >> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, >> NOTIFY, REFER, MESSAGE, OPTIONS >> Supported: replaces, 100rel, timer, norefersub >> Session-Expires: 1800 >> Min-SE: 90 >> User-Agent: AdCall >> Proxy-Authorization: Digest username="102", >> realm="telequest-occ.0049.org", >> nonce="01e39fee-93b0-11e7-9b0b-45e8f1c3918a", >> uri="sip:+436641540180 at telequest-occ.0049.org", >> response="c7fef167bd3ecc40bdaeaaef9c9b8085", algorithm=MD5, >> cnonce="raNBupUIEw5CyLL8MQbvYrk5txEd3Zep", qop=auth, nc=00000001 >> Content-Type: application/sdp >> Content-Length: 482 >> >> v=0 >> o=- 3713765781 3713765781 IN IP4 192.168.178.110 >> s=pjmedia >> b=AS:84 >> t=0 0 >> a=X-nat:0 >> m=audio 4000 RTP/AVP 98 97 99 104 3 0 8 9 96 >> c=IN IP4 192.168.178.110 >> b=TIAS:64000 >> a=rtcp:4001 IN IP4 192.168.178.110 >> a=sendrecv >> a=rtpmap:98 speex/16000 >> a=rtpmap:97 speex/8000 >> a=rtpmap:99 speex/32000 >> a=rtpmap:104 iLBC/8000 >> a=fmtp:104 mode=30 >> a=rtpmap:3 GSM/8000 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:9 G722/8000 >> a=rtpmap:96 telephone-event/8000 >> a=fmtp:96 0-16 >> ------------------------------------------------------------------------ >> send 380 bytes to tcp/[77.119.248.251]:60186 at 11:36:21.457978: >> ------------------------------------------------------------------------ >> SIP/2.0 100 Trying >> Via: SIP/2.0/TCP >> 192.168.178.110:60186;rport=60186;branch=z9hG4bKPj4mpXCeY7uNSlFOUvRXIvfftWSq9PQE-7;alias;received=77.119.248.251 >> From: sip:102 at telequest-occ.0049.org;tag=DDna03R3v1t0hJut1phv2uhb4OxBMYnu >> To: sip:+436641540180 at telequest-occ.0049.org >> Call-ID: VQlDTx2IUOlzd5QZuZHdLoqbyOGtmO1Z >> CSeq: 17633 INVITE >> User-Agent: FreeSWITCH >> Content-Length: 0 >> >> ------------------------------------------------------------------------ >> send 1193 bytes to tcp/[77.119.248.251]:60186 at 11:36:24.966185: >> ------------------------------------------------------------------------ >> SIP/2.0 183 Session Progress >> Via: SIP/2.0/TCP >> 192.168.178.110:60186;rport=60186;branch=z9hG4bKPj4mpXCeY7uNSlFOUvRXIvfftWSq9PQE-7;alias;received=77.119.248.251 >> From: sip:102 at telequest-occ.0049.org;tag=DDna03R3v1t0hJut1phv2uhb4OxBMYnu >> To: ;tag=1r9F72USBUrQj >> Call-ID: VQlDTx2IUOlzd5QZuZHdLoqbyOGtmO1Z >> CSeq: 17633 INVITE >> Contact: >> User-Agent: FreeSWITCH >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, path, replaces >> Allow-Events: talk, hold, conference, presence, as-feature-event, >> dialog, line-seize, call-info, sla, include-session-description, >> presence.winfo, message-summary, refer >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 203 >> Remote-Party-ID: "+436641540180" >> ;party=calling;privacy=off;screen=no >> >> v=0 >> o=FreeSWITCH 1504753876 1504753877 IN IP4 88.217.132.183 >> s=FreeSWITCH >> c=IN IP4 88.217.132.183 >> t=0 0 >> m=audio 23108 RTP/AVP 3 >> a=rtpmap:3 GSM/8000 >> a=ptime:20 >> a=rtcp:23109 IN IP4 88.217.132.183 >> ------------------------------------------------------------------------ >> recv 401 bytes from tcp/[77.119.248.251]:60186 at 11:36:28.869708: >> ------------------------------------------------------------------------ >> CANCEL sip:+436641540180 at telequest-occ.0049.org SIP/2.0 >> Via: SIP/2.0/TCP >> 192.168.178.110:5060;rport;branch=z9hG4bKPj4mpXCeY7uNSlFOUvRXIvfftWSq9PQE-7;alias >> Max-Forwards: 70 >> From: sip:102 at telequest-occ.0049.org;tag=DDna03R3v1t0hJut1phv2uhb4OxBMYnu >> To: sip:+436641540180 at telequest-occ.0049.org >> Call-ID: VQlDTx2IUOlzd5QZuZHdLoqbyOGtmO1Z >> CSeq: 17633 CANCEL >> User-Agent: AdCall >> Content-Length: 0 >> >> ------------------------------------------------------------------------ >> send 400 bytes to tcp/[77.119.248.251]:60186 at 11:36:28.869807: >> ------------------------------------------------------------------------ >> SIP/2.0 481 Call/Transaction Does Not Exist >> Via: SIP/2.0/TCP >> 192.168.178.110:5060;rport=60186;branch=z9hG4bKPj4mpXCeY7uNSlFOUvRXIvfftWSq9PQE-7;alias;received=77.119.248.251 >> From: sip:102 at telequest-occ.0049.org;tag=DDna03R3v1t0hJut1phv2uhb4OxBMYnu >> To: ;tag=1r9F72USBUrQj >> Call-ID: VQlDTx2IUOlzd5QZuZHdLoqbyOGtmO1Z >> CSeq: 17633 CANCEL >> Content-Length: 0 >> >> ------------------------------------------------------------------------ >> >> Best regards, >> Thorsten >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > From lists at telefaks.de Thu Sep 7 14:05:02 2017 From: lists at telefaks.de (Peter Steinbach) Date: Thu, 07 Sep 2017 16:05:02 +0200 Subject: [Freeswitch-users] SIP-Client / IOS / PJSIP / Error when CANCEL In-Reply-To: References: <59B13029.7060908@telefaks.de> Message-ID: <59B1520E.4020901@telefaks.de> Hello Thorsten, I don't have any insight in PJSIP, but I see that the IP changed from an external to an internal IP. This may?? be caused by the switch from UDP to TCP, dependend on how the external IP is determined (STUN?). Best regards Peter On 09/07/17 15:43, Thorsten Göllner wrote: > Hi Peter, > > thanks so far. But why is the Contact: header changed (by pjsip?) when > switching from TCP/UDP to UDP-only? I am not sure for now but I don't > think, that our developer changed the Contact: header explicitly. > > Best regards, > Thorsten > > Am 07.09.2017 um 13:40 schrieb Peter Steinbach: >> Hello Thorsten >> >> this is often caused by changed Contact: header in your SIP dialog. In >> your case the 2 Invites have: >> Contact: >> Contact: >> >> So this will not work. >> >> Best regards from Germany >> Peter >> >> >> On 09/07/17 13:19, Thorsten Göllner wrote: >>> Hi, >>> >>> currently we are developing a SIP-Client for iOS (using PJSIP/PJSUA with >>> the current version). We use Freeswitch 1.6.18 (64 bit) on a bare metal >>> server. Windows-Clients (such as Blink or Zoiper) work fine. And our >>> iOS-App works fine also. BUT when we initialze an outgoing call, wait >>> for ringing on the other hand and hangup up when it's ringing, we face >>> an error. Freeswitch gives us "481 Call/Transaction Does Not Exist". >>> >>> I attached the corresponding SIP-Trace and you can see the error at the >>> last block. >>> >>> The interesting thing: If we activate PJSIP with TCP and(!) UDP we face >>> the error. BUT when we use only UDP it works fine. >>> >>> Is it a bug within PJSIP? Or anyone see an hint wihtin the SIP-Trace? >>> >>> Thanks in advance! >>> >>> recv 1247 bytes from udp/[77.119.248.251]:5060 at 11:36:21.330337: >>> ------------------------------------------------------------------------ >>> INVITE sip:+436641540180 at telequest-occ.0049.org SIP/2.0 >>> Via: SIP/2.0/UDP >>> 77.119.248.251:5060;branch=z9hG4bK4b69df210aa918eb70770f8f50681c51 >>> Via: SIP/2.0/UDP >>> 192.168.178.110:5060;rport;branch=z9hG4bKPjBP3AvciR4ifGE8V3KBSuxInUNOQN8VLh >>> From: >>> ;tag=DDna03R3v1t0hJut1phv2uhb4OxBMYnu >>> To: >>> Call-ID: VQlDTx2IUOlzd5QZuZHdLoqbyOGtmO1Z >>> CSeq: 17632 INVITE >>> Contact: >>> max-forwards: 69 >>> supported: replaces >>> supported: 100rel >>> supported: timer >>> supported: norefersub >>> session-expires: 1800 >>> min-se: 90 >>> user-agent: AdCall >>> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, >>> NOTIFY, REFER, MESSAGE, OPTIONS >>> Content-Type: application/sdp >>> Content-Length: 480 >>> >>> v=0 >>> o=- 3713765781 3713765781 IN IP4 77.119.248.251 >>> s=pjmedia >>> b=AS:84 >>> t=0 0 >>> a=X-nat:0 >>> m=audio 7070 RTP/AVP 98 97 99 104 3 0 8 9 96 >>> c=IN IP4 77.119.248.251 >>> b=TIAS:64000 >>> a=rtcp:4001 IN IP4 192.168.178.110 >>> a=sendrecv >>> a=rtpmap:98 speex/16000 >>> a=rtpmap:97 speex/8000 >>> a=rtpmap:99 speex/32000 >>> a=rtpmap:104 iLBC/8000 >>> a=fmtp:104 mode=30 >>> a=rtpmap:3 GSM/8000 >>> a=rtpmap:0 PCMU/8000 >>> a=rtpmap:8 PCMA/8000 >>> a=rtpmap:9 G722/8000 >>> a=rtpmap:96 telephone-event/8000 >>> a=fmtp:96 0-16 >>> ------------------------------------------------------------------------ >>> send 432 bytes to udp/[77.119.248.251]:5060 at 11:36:21.330679: >>> ------------------------------------------------------------------------ >>> SIP/2.0 100 Trying >>> Via: SIP/2.0/UDP >>> 77.119.248.251:5060;branch=z9hG4bK4b69df210aa918eb70770f8f50681c51 >>> Via: SIP/2.0/UDP >>> 192.168.178.110:5060;rport;branch=z9hG4bKPjBP3AvciR4ifGE8V3KBSuxInUNOQN8VLh >>> From: >>> ;tag=DDna03R3v1t0hJut1phv2uhb4OxBMYnu >>> To: >>> Call-ID: VQlDTx2IUOlzd5QZuZHdLoqbyOGtmO1Z >>> CSeq: 17632 INVITE >>> User-Agent: FreeSWITCH >>> Content-Length: 0 >>> >>> ------------------------------------------------------------------------ >>> send 945 bytes to udp/[77.119.248.251]:5060 at 11:36:21.331652: >>> ------------------------------------------------------------------------ >>> SIP/2.0 407 Proxy Authentication Required >>> Via: SIP/2.0/UDP >>> 77.119.248.251:5060;branch=z9hG4bK4b69df210aa918eb70770f8f50681c51 >>> Via: SIP/2.0/UDP >>> 192.168.178.110:5060;rport;branch=z9hG4bKPjBP3AvciR4ifGE8V3KBSuxInUNOQN8VLh >>> From: >>> ;tag=DDna03R3v1t0hJut1phv2uhb4OxBMYnu >>> To: ;tag=0FgQ57apej24p >>> Call-ID: VQlDTx2IUOlzd5QZuZHdLoqbyOGtmO1Z >>> CSeq: 17632 INVITE >>> User-Agent: FreeSWITCH >>> Accept: application/sdp >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, >>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>> Supported: timer, path, replaces >>> Allow-Events: talk, hold, conference, presence, as-feature-event, >>> dialog, line-seize, call-info, sla, include-session-description, >>> presence.winfo, message-summary, refer >>> Proxy-Authenticate: Digest realm="telequest-occ.0049.org", >>> nonce="01e39fee-93b0-11e7-9b0b-45e8f1c3918a", algorithm=MD5, qop="auth" >>> Content-Length: 0 >>> >>> ------------------------------------------------------------------------ >>> recv 475 bytes from udp/[77.119.248.251]:5060 at 11:36:21.409776: >>> ------------------------------------------------------------------------ >>> ACK sip:+436641540180 at telequest-occ.0049.org SIP/2.0 >>> Via: SIP/2.0/UDP >>> 77.119.248.251:5060;branch=z9hG4bK4b69df210aa918eb70770f8f50681c51 >>> Via: SIP/2.0/UDP >>> 192.168.178.110:5060;rport;branch=z9hG4bKPjBP3AvciR4ifGE8V3KBSuxInUNOQN8VLh >>> From: >>> ;tag=DDna03R3v1t0hJut1phv2uhb4OxBMYnu >>> To: ;tag=0FgQ57apej24p >>> Call-ID: VQlDTx2IUOlzd5QZuZHdLoqbyOGtmO1Z >>> CSeq: 17632 ACK >>> max-forwards: 69 >>> Content-Length: 0 >>> >>> ------------------------------------------------------------------------ >>> recv 1439 bytes from tcp/[77.119.248.251]:60186 at 11:36:21.457742: >>> ------------------------------------------------------------------------ >>> INVITE sip:+436641540180 at telequest-occ.0049.org SIP/2.0 >>> Via: SIP/2.0/TCP >>> 192.168.178.110:60186;rport;branch=z9hG4bKPj4mpXCeY7uNSlFOUvRXIvfftWSq9PQE-7;alias >>> Max-Forwards: 70 >>> From: sip:102 at telequest-occ.0049.org;tag=DDna03R3v1t0hJut1phv2uhb4OxBMYnu >>> To: sip:+436641540180 at telequest-occ.0049.org >>> Contact: >>> Call-ID: VQlDTx2IUOlzd5QZuZHdLoqbyOGtmO1Z >>> CSeq: 17633 INVITE >>> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, >>> NOTIFY, REFER, MESSAGE, OPTIONS >>> Supported: replaces, 100rel, timer, norefersub >>> Session-Expires: 1800 >>> Min-SE: 90 >>> User-Agent: AdCall >>> Proxy-Authorization: Digest username="102", >>> realm="telequest-occ.0049.org", >>> nonce="01e39fee-93b0-11e7-9b0b-45e8f1c3918a", >>> uri="sip:+436641540180 at telequest-occ.0049.org", >>> response="c7fef167bd3ecc40bdaeaaef9c9b8085", algorithm=MD5, >>> cnonce="raNBupUIEw5CyLL8MQbvYrk5txEd3Zep", qop=auth, nc=00000001 >>> Content-Type: application/sdp >>> Content-Length: 482 >>> >>> v=0 >>> o=- 3713765781 3713765781 IN IP4 192.168.178.110 >>> s=pjmedia >>> b=AS:84 >>> t=0 0 >>> a=X-nat:0 >>> m=audio 4000 RTP/AVP 98 97 99 104 3 0 8 9 96 >>> c=IN IP4 192.168.178.110 >>> b=TIAS:64000 >>> a=rtcp:4001 IN IP4 192.168.178.110 >>> a=sendrecv >>> a=rtpmap:98 speex/16000 >>> a=rtpmap:97 speex/8000 >>> a=rtpmap:99 speex/32000 >>> a=rtpmap:104 iLBC/8000 >>> a=fmtp:104 mode=30 >>> a=rtpmap:3 GSM/8000 >>> a=rtpmap:0 PCMU/8000 >>> a=rtpmap:8 PCMA/8000 >>> a=rtpmap:9 G722/8000 >>> a=rtpmap:96 telephone-event/8000 >>> a=fmtp:96 0-16 >>> ------------------------------------------------------------------------ >>> send 380 bytes to tcp/[77.119.248.251]:60186 at 11:36:21.457978: >>> ------------------------------------------------------------------------ >>> SIP/2.0 100 Trying >>> Via: SIP/2.0/TCP >>> 192.168.178.110:60186;rport=60186;branch=z9hG4bKPj4mpXCeY7uNSlFOUvRXIvfftWSq9PQE-7;alias;received=77.119.248.251 >>> From: sip:102 at telequest-occ.0049.org;tag=DDna03R3v1t0hJut1phv2uhb4OxBMYnu >>> To: sip:+436641540180 at telequest-occ.0049.org >>> Call-ID: VQlDTx2IUOlzd5QZuZHdLoqbyOGtmO1Z >>> CSeq: 17633 INVITE >>> User-Agent: FreeSWITCH >>> Content-Length: 0 >>> >>> ------------------------------------------------------------------------ >>> send 1193 bytes to tcp/[77.119.248.251]:60186 at 11:36:24.966185: >>> ------------------------------------------------------------------------ >>> SIP/2.0 183 Session Progress >>> Via: SIP/2.0/TCP >>> 192.168.178.110:60186;rport=60186;branch=z9hG4bKPj4mpXCeY7uNSlFOUvRXIvfftWSq9PQE-7;alias;received=77.119.248.251 >>> From: sip:102 at telequest-occ.0049.org;tag=DDna03R3v1t0hJut1phv2uhb4OxBMYnu >>> To: ;tag=1r9F72USBUrQj >>> Call-ID: VQlDTx2IUOlzd5QZuZHdLoqbyOGtmO1Z >>> CSeq: 17633 INVITE >>> Contact: >>> User-Agent: FreeSWITCH >>> Accept: application/sdp >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, >>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>> Supported: timer, path, replaces >>> Allow-Events: talk, hold, conference, presence, as-feature-event, >>> dialog, line-seize, call-info, sla, include-session-description, >>> presence.winfo, message-summary, refer >>> Content-Type: application/sdp >>> Content-Disposition: session >>> Content-Length: 203 >>> Remote-Party-ID: "+436641540180" >>> ;party=calling;privacy=off;screen=no >>> >>> v=0 >>> o=FreeSWITCH 1504753876 1504753877 IN IP4 88.217.132.183 >>> s=FreeSWITCH >>> c=IN IP4 88.217.132.183 >>> t=0 0 >>> m=audio 23108 RTP/AVP 3 >>> a=rtpmap:3 GSM/8000 >>> a=ptime:20 >>> a=rtcp:23109 IN IP4 88.217.132.183 >>> ------------------------------------------------------------------------ >>> recv 401 bytes from tcp/[77.119.248.251]:60186 at 11:36:28.869708: >>> ------------------------------------------------------------------------ >>> CANCEL sip:+436641540180 at telequest-occ.0049.org SIP/2.0 >>> Via: SIP/2.0/TCP >>> 192.168.178.110:5060;rport;branch=z9hG4bKPj4mpXCeY7uNSlFOUvRXIvfftWSq9PQE-7;alias >>> Max-Forwards: 70 >>> From: sip:102 at telequest-occ.0049.org;tag=DDna03R3v1t0hJut1phv2uhb4OxBMYnu >>> To: sip:+436641540180 at telequest-occ.0049.org >>> Call-ID: VQlDTx2IUOlzd5QZuZHdLoqbyOGtmO1Z >>> CSeq: 17633 CANCEL >>> User-Agent: AdCall >>> Content-Length: 0 >>> >>> ------------------------------------------------------------------------ >>> send 400 bytes to tcp/[77.119.248.251]:60186 at 11:36:28.869807: >>> ------------------------------------------------------------------------ >>> SIP/2.0 481 Call/Transaction Does Not Exist >>> Via: SIP/2.0/TCP >>> 192.168.178.110:5060;rport=60186;branch=z9hG4bKPj4mpXCeY7uNSlFOUvRXIvfftWSq9PQE-7;alias;received=77.119.248.251 >>> From: sip:102 at telequest-occ.0049.org;tag=DDna03R3v1t0hJut1phv2uhb4OxBMYnu >>> To: ;tag=1r9F72USBUrQj >>> Call-ID: VQlDTx2IUOlzd5QZuZHdLoqbyOGtmO1Z >>> CSeq: 17633 CANCEL >>> Content-Length: 0 >>> >>> ------------------------------------------------------------------------ >>> >>> Best regards, >>> Thorsten >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org > > -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de From josefu at gmail.com Thu Sep 7 15:02:28 2017 From: josefu at gmail.com (=?UTF-8?Q?Jose_Fco=2E_Irles_Dur=C3=A1?=) Date: Thu, 7 Sep 2017 17:02:28 +0200 Subject: [Freeswitch-users] Implement PushKit or CallKit to call iOS app closed In-Reply-To: References: Message-ID: We have implemented a push system to iOS/Android devices with FreeSWITCH following the next steps: 1. Incomming call to user bob in FreeSWITCH 2. FreeSWITCH park the call 3. Our manager software (xml_curl) sends a push to bob's iOS device 4. Notification arrives to iOS device and the softphone call to a uri sended in the push notification (this uri is the server that previously has parked the call) 5. FreeSWITCH bridge the two legs (before this, manage the "ringing" event) The FreeSWITCH part is developed in a simple script in lua. Best regards 2017-09-07 10:55 GMT+02:00 Jose David Jurado Alonso : > Hi, > > I can't find any information over how to implement PushKit or Callkit to > call an iOS mobile phone that has the SIP client application closed or in > the background. > > When the application is closed it is obviously that SIP account isn't > registered and the FS server no send the call. > > Has anyone worked with this or know how to implement it? > > I tried with " value="NDLB-connectile-dysfunction"/>" option and this temporally solved > when the application is alive in background. > > Thanks, > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Jose Fco. Irles Durá From mike at jerris.com Thu Sep 7 16:41:02 2017 From: mike at jerris.com (Michael Jerris) Date: Thu, 07 Sep 2017 16:41:02 +0000 Subject: [Freeswitch-users] Possible enhancement - load mod_logfile way earlier In-Reply-To: <760B20D3-CE69-4BD0-AAAE-CC860A3327AF@mgtech.com> References: <760B20D3-CE69-4BD0-AAAE-CC860A3327AF@mgtech.com> Message-ID: looking at our default config, it starts pretty early. What configs are you using where it starts late? https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/conf/vanilla/autoload_configs/modules.conf.xml On Mon, Aug 28, 2017 at 3:15 PM Mario G wrote: > While working on an issue, I realized I was missing a lot of the startup > log. The reason is that FreeSWITCH is started in the background with -NC > and only the log file it writes is all that’s available. Problem is > mod_logfile is loaded long after other things and all messages prior to > mod_logfile are lost. Loading mod_logfile as soon as possible at a startup > would prevent all the lost startup messages. Let me know if you want a jira. > Mario G > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Thu Sep 7 16:42:31 2017 From: mike at jerris.com (Michael Jerris) Date: Thu, 07 Sep 2017 16:42:31 +0000 Subject: [Freeswitch-users] v8 failing to build master branch In-Reply-To: References: Message-ID: file not found. Are you building in cowbuilder? On Tue, Aug 29, 2017 at 1:00 AM John Covici wrote: > Hi. I am trying to update to masterand v8 will not build and I have > no idea even what this error means. Here is the complete build log. > It is saying ninja subcommand failed. > > https://covici.com/owncloud/index.php/s/6MBIR4hnWdu8lWG > > Am I doing something wrong or should I file a bug? > > Thanks in advance for any suggestions. > > -- > Your life is like a penny. You're going to lose it. The question is: > How do > you spend it? > > John Covici > covici at ccs.covici.com > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Thu Sep 7 17:00:55 2017 From: mike at jerris.com (Michael Jerris) Date: Thu, 07 Sep 2017 17:00:55 +0000 Subject: [Freeswitch-users] Cisco IP Phones & Counterpath Softphones Crashing w/ mod_av Enabled In-Reply-To: <20501504686328@web53g.yandex.ru> References: <20501504686328@web53g.yandex.ru> Message-ID: I haven't tested all those end points but for sure we have used video with latest and beta bria and had it work. If clients are crashing when receiving video in sdp you should consult with those vendors to find out why. On Wed, Sep 6, 2017 at 3:31 AM roy j wrote: > Hi. Just interested, you say they crash, what exactly is happening, they > are rebooting? If so I would ask Cisco about it, it's so strange. > > 05.09.2017, 10:23, "Joseph Barrero" : > > Hi, I’m running Freeswitch v1.6.19. > > When mod_av is enabled, inbound calls to Cisco SPA504g and SPA514g ip > phones crash as they start to ring. The crash seems to occur immediately > after Freeswitch sends SIP Invite with the SDP information that includes > the video details. I don’t have any problems when I disable mod_av and > include PCMU, PCMA, and OPUS. The latest versions of X-Lite and Bria 5 > also crash with mod_av enabled. > > Bria 4 and other models of ip phones (Polycom, Yealink, Grandstream) don’t > seem to have a problem. > > Any ideas what could be wrong? > > - Joe Barrero > > , > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From mjlopez at smartic.es Thu Sep 7 17:59:23 2017 From: mjlopez at smartic.es (=?iso-8859-1?Q?Miguel_Jes=FAs_L=F3pez_Valverde?=) Date: Thu, 7 Sep 2017 19:59:23 +0200 Subject: [Freeswitch-users] use of information with query 'sofia status' and locks in internal interface Message-ID: <02f501d32803$0a4037b0$1ec0a710$@smartic.es> Hello, good afternoon. I send this email to see if anyone can give me information about the meaning of the values (2) after the RUNNING state obtained on the query sofia status. I searched for forums and books, but I find nothing explanatory. freeswitch at ip-xxx-xxx-xxx-xxx> sofia status Name Type Data State ============================================================================ ===================== external profile sip:mod_sofia at xxx.xxx.xxx.xxx:5080 RUNNING (0) external profile sip:mod_sofia at xxx.xxx.xxx.xxx:5081 RUNNING (0) (TLS) internal profile sip:mod_sofia at xxx.xxx.xxx.xxx:5060 RUNNING (2) internal profile sip:mod_sofia at xxx.xxx.xxx.xxx:5061 RUNNING (2) (TLS) ============================================================================ ===================== 2 profiles 0 aliases It happens sometimes that this value rises above 2 in the internal interface, I do not know if it is casual, but it blocks that interface and does not respond to sip requests. To identify this problem I have traversed the log files and nothing appears that indicates what may be happening and it becomes necessary to pull and lift the freeswitch service to re-handle the sip requests correctly. Also, from console, if the sofia status profile internal command is executed, no response is obtained and the query must be canceled. Thank you very much. Miguel J. López Valverde Dpto. Técnico. _____ Pº de la Castellana, 135 7ª pl 28046 Madrid España Tel: 900 900 368 | Móvil: (+34) 667 772 911 E-mail: mjlopez at smartic.es | Web: http://www.smartic.es _____ Sus datos de carácter personal (nombre, apellidos, dirección postal y de correo electrónico, etc.) son tratados para la gestión de su relación con la Entidad, así como para el envío de información sobre nuestra actividad y la de terceros relacionadas con la actividad de Consulting Smartic Solutions, S.L., CIF: B85130037, Pº de la Castellana, 135 7ª pl 28046 Madrid. Usted puede ejercer sus derechos de acceso, rectificación, cancelación y oposición dirigiéndose por escrito, con copia de un documento que acredite su identidad, a la dirección info (arroba) smartic.es. Este mensaje puede contener información confidencial. Si usted no es su destinatario, no debe leerlo, copiarlo, distribuirlo, ni hacer uso de la información que contiene. En este caso, por favor, llámenos o comuníquenoslo por escrito y borre este mensaje de su sistema --- El software de antivirus Avast ha analizado este correo electrónico en busca de virus. https://www.avast.com/antivirus -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.png Type: image/png Size: 4475 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image002.jpg Type: image/jpeg Size: 1217 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image003.jpg Type: image/jpeg Size: 1201 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image004.jpg Type: image/jpeg Size: 1184 bytes Desc: not available URL: From rick at magicmail.mooo.com Thu Sep 7 20:01:14 2017 From: rick at magicmail.mooo.com (Rick Jarvis) Date: Thu, 7 Sep 2017 21:01:14 +0100 Subject: [Freeswitch-users] Verto Message-ID: <03E75515-780B-451B-A30F-62C99020D8A7@magicmail.mooo.com> Looking to get into Verto, in particular handling voice calls with JS. Going through the source, I’m wondering: 1. What’s the difference between the Verto source and the Verto Communicator source? 2. What’s the best way to start from the bottom up - by this I mean that it seems hugely comprehensive, but rather than just use grunt to set it all up, I’d like to start simply with the basics… is there for instance a list of the bare minimum scripts / file structure to use? Apologies if this is a silly question, I’m still relatively new to JS and I don’t want to blow my mind in one go ;) Rick From covici at ccs.covici.com Thu Sep 7 20:11:59 2017 From: covici at ccs.covici.com (John Covici) Date: Thu, 07 Sep 2017 16:11:59 -0400 Subject: [Freeswitch-users] v8 failing to build master branch In-Reply-To: References: Message-ID: This has been solved -- I had to update my compile of ncurses to create the tinfo library. On Thu, 07 Sep 2017 12:42:31 -0400, Michael Jerris wrote: > > [1 ] > [1.1 ] > [1.2 ] > file not found. Are you building in cowbuilder? > > On Tue, Aug 29, 2017 at 1:00 AM John Covici wrote: > > Hi. I am trying to update to masterand v8 will not build and I have > no idea even what this error means. Here is the complete build log. > It is saying ninja subcommand failed. > > https://covici.com/owncloud/index.php/s/6MBIR4hnWdu8lWG > > Am I doing something wrong or should I file a bug? > > Thanks in advance for any suggestions. > > -- > Your life is like a penny. You're going to lose it. The question is: > How do > you spend it? > > John Covici > covici at ccs.covici.com > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > [2 ] > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici at ccs.covici.com From italo at freeswitch.org Thu Sep 7 20:11:55 2017 From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=) Date: Thu, 07 Sep 2017 20:11:55 +0000 Subject: [Freeswitch-users] Verto In-Reply-To: <03E75515-780B-451B-A30F-62C99020D8A7@magicmail.mooo.com> References: <03E75515-780B-451B-A30F-62C99020D8A7@magicmail.mooo.com> Message-ID: Verto communicator is a example implementation on what can be done with Verto. Check our tutorial to understand how to build a minimum app: https://evoluxbr.github.io/verto-docs/ Em qui, 7 de set de 2017 às 17:02, Rick Jarvis escreveu: > Looking to get into Verto, in particular handling voice calls with JS. > Going through the source, I’m wondering: > > 1. What’s the difference between the Verto source and the Verto > Communicator source? > > 2. What’s the best way to start from the bottom up - by this I mean that > it seems hugely comprehensive, but rather than just use grunt to set it all > up, I’d like to start simply with the basics… is there for instance a list > of the bare minimum scripts / file structure to use? Apologies if this is a > silly question, I’m still relatively new to JS and I don’t want to blow my > mind in one go ;) > > Rick > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From luis.azedo at factorlusitano.com Thu Sep 7 21:41:36 2017 From: luis.azedo at factorlusitano.com (Luis Azedo) Date: Thu, 7 Sep 2017 22:41:36 +0100 Subject: [Freeswitch-users] Cisco IP Phones & Counterpath Softphones Crashing w/ mod_av Enabled Message-ID: > > i noticed that freeswitch crashes with mod_av enabled in debian stretch > > ---------- Forwarded message ---------- > From: Michael Jerris > To: FreeSWITCH Users Help > Cc: > Bcc: > Date: Thu, 07 Sep 2017 17:00:55 +0000 > Subject: Re: [Freeswitch-users] Cisco IP Phones & Counterpath Softphones > Crashing w/ mod_av Enabled > I haven't tested all those end points but for sure we have used video with > latest and beta bria and had it work. If clients are crashing when > receiving video in sdp you should consult with those vendors to find out > why. > > On Wed, Sep 6, 2017 at 3:31 AM roy j wrote: > >> Hi. Just interested, you say they crash, what exactly is happening, they >> are rebooting? If so I would ask Cisco about it, it's so strange. >> >> 05.09.2017, 10:23, "Joseph Barrero" : >> >> Hi, I’m running Freeswitch v1.6.19. >> >> When mod_av is enabled, inbound calls to Cisco SPA504g and SPA514g ip >> phones crash as they start to ring. The crash seems to occur immediately >> after Freeswitch sends SIP Invite with the SDP information that includes >> the video details. I don’t have any problems when I disable mod_av and >> include PCMU, PCMA, and OPUS. The latest versions of X-Lite and Bria 5 >> also crash with mod_av enabled. >> >> Bria 4 and other models of ip phones (Polycom, Yealink, Grandstream) >> don’t seem to have a problem. >> >> Any ideas what could be wrong? >> >> - Joe Barrero >> >> , >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From clive at lansink.co.nz Thu Sep 7 22:03:32 2017 From: clive at lansink.co.nz (Clive Lansink) Date: Fri, 8 Sep 2017 10:03:32 +1200 Subject: [Freeswitch-users] PortAudio and Windows devices Message-ID: An embedded and charset-unspecified text was scrubbed... Name: not available URL: From rick at magicmail.mooo.com Thu Sep 7 23:21:03 2017 From: rick at magicmail.mooo.com (Rick Jarvis) Date: Fri, 8 Sep 2017 00:21:03 +0100 Subject: [Freeswitch-users] Verto In-Reply-To: References: <03E75515-780B-451B-A30F-62C99020D8A7@magicmail.mooo.com> Message-ID: <163FFD7C-BEA0-4370-B8AC-6DFE97CF6A50@magicmail.mooo.com> That looks really helpful, thank you. > On 7 Sep 2017, at 21:11, Ítalo Rossi wrote: > > Verto communicator is a example implementation on what can be done with Verto. > > Check our tutorial to understand how to build a minimum app: > > https://evoluxbr.github.io/verto-docs/ > Em qui, 7 de set de 2017 às 17:02, Rick Jarvis > escreveu: > Looking to get into Verto, in particular handling voice calls with JS. Going through the source, I’m wondering: > > 1. What’s the difference between the Verto source and the Verto Communicator source? > > 2. What’s the best way to start from the bottom up - by this I mean that it seems hugely comprehensive, but rather than just use grunt to set it all up, I’d like to start simply with the basics… is there for instance a list of the bare minimum scripts / file structure to use? Apologies if this is a silly question, I’m still relatively new to JS and I don’t want to blow my mind in one go ;) > > Rick > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From lists at pbaines.com Thu Sep 7 17:23:37 2017 From: lists at pbaines.com (Peter Baines (lists)) Date: Thu, 7 Sep 2017 18:23:37 +0100 Subject: [Freeswitch-users] Video Conferencing freezing and lag in v1.8 and master Message-ID: Hello, I'm trying to get a simple video conference between two endpoints. I'm constantly seeing freeze and lag but I don't see any issues around CPU/memory/network that may cause this, has anyone come across something similar? I do notice in the debug log it keeps changing to large resolutions: e.g. VPX reset encoder picture from 352x288 to 720x1280 2048 BW Is there away to define the max resolution ? These are the current settings I'm using, in vars.xml I have: This is the profile from conference.conf.xml (tried both transcode and mux): The dialplan bridges like: Regards, Peter -------------- next part -------------- An HTML attachment was scrubbed... URL: From joe at expert.net Thu Sep 7 22:15:32 2017 From: joe at expert.net (Joseph Barrero) Date: Thu, 7 Sep 2017 17:15:32 -0500 Subject: [Freeswitch-users] Cisco IP Phones & Counterpath Softphones Crashing w/ mod_av Enabled In-Reply-To: References: Message-ID: To clarify a few details: I'm running on Debian 8 (Jessie). The Cisco phones crash by lighting up, ringing continuously, and the website interface, dialpad, and buttons become unresponsive. The phone must be unplugged. The latest version of X-Lite and Bria 5 crash on Mac OS X crashes with the Error that displays: *Thread 6 Crashed:: PhoneInterface* 0 libsystem_kernel.dylib 0x00007fffd873ad42 __pthread_kill + 10 1 libsystem_pthread.dylib 0x00007fffd8828457 pthread_kill + 90 2 libsystem_c.dylib 0x00007fffd86a04bb __abort + 140 3 libsystem_c.dylib 0x00007fffd86a0d7e __stack_chk_fail + 205 4 counterpath.CPCAPI2 0x000000010821dd87 CPCAPI2::SipConversation::ReconConversationManagerImpl::onIncomingParticipant(unsigned int, resip::SipMessage const&, bool, bool, recon::ConversationManager::MediaSpecificMismatchInfo const&, recon::ConversationManager::MediaAttributes const&) + 3633 5 ??? 0x0d44667062384453 0 + 956001653779612755 On Thu, Sep 7, 2017 at 4:41 PM, Luis Azedo wrote: > i noticed that freeswitch crashes with mod_av enabled in debian stretch >> > > >> ---------- Forwarded message ---------- >> From: Michael Jerris >> To: FreeSWITCH Users Help >> Cc: >> Bcc: >> Date: Thu, 07 Sep 2017 17:00:55 +0000 >> Subject: Re: [Freeswitch-users] Cisco IP Phones & Counterpath Softphones >> Crashing w/ mod_av Enabled >> I haven't tested all those end points but for sure we have used video >> with latest and beta bria and had it work. If clients are crashing when >> receiving video in sdp you should consult with those vendors to find out >> why. >> >> On Wed, Sep 6, 2017 at 3:31 AM roy j wrote: >> >>> Hi. Just interested, you say they crash, what exactly is happening, they >>> are rebooting? If so I would ask Cisco about it, it's so strange. >>> >>> 05.09.2017, 10:23, "Joseph Barrero" : >>> >>> Hi, I’m running Freeswitch v1.6.19. >>> >>> When mod_av is enabled, inbound calls to Cisco SPA504g and SPA514g ip >>> phones crash as they start to ring. The crash seems to occur immediately >>> after Freeswitch sends SIP Invite with the SDP information that includes >>> the video details. I don’t have any problems when I disable mod_av and >>> include PCMU, PCMA, and OPUS. The latest versions of X-Lite and Bria 5 >>> also crash with mod_av enabled. >>> >>> Bria 4 and other models of ip phones (Polycom, Yealink, Grandstream) >>> don’t seem to have a problem. >>> >>> Any ideas what could be wrong? >>> >>> - Joe Barrero >>> >>> , >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Fri Sep 8 01:16:48 2017 From: mike at jerris.com (Michael Jerris) Date: Fri, 08 Sep 2017 01:16:48 +0000 Subject: [Freeswitch-users] Cisco IP Phones & Counterpath Softphones Crashing w/ mod_av Enabled In-Reply-To: References: Message-ID: Then please report that properly On Thu, Sep 7, 2017 at 5:45 PM Luis Azedo wrote: > i noticed that freeswitch crashes with mod_av enabled in debian stretch >> > > >> ---------- Forwarded message ---------- >> From: Michael Jerris >> To: FreeSWITCH Users Help >> Cc: >> Bcc: >> Date: Thu, 07 Sep 2017 17:00:55 +0000 >> Subject: Re: [Freeswitch-users] Cisco IP Phones & Counterpath Softphones >> Crashing w/ mod_av Enabled >> I haven't tested all those end points but for sure we have used video >> with latest and beta bria and had it work. If clients are crashing when >> receiving video in sdp you should consult with those vendors to find out >> why. >> >> On Wed, Sep 6, 2017 at 3:31 AM roy j wrote: >> >>> Hi. Just interested, you say they crash, what exactly is happening, they >>> are rebooting? If so I would ask Cisco about it, it's so strange. >>> >>> 05.09.2017, 10:23, "Joseph Barrero" : >>> >>> Hi, I’m running Freeswitch v1.6.19. >>> >>> When mod_av is enabled, inbound calls to Cisco SPA504g and SPA514g ip >>> phones crash as they start to ring. The crash seems to occur immediately >>> after Freeswitch sends SIP Invite with the SDP information that includes >>> the video details. I don’t have any problems when I disable mod_av and >>> include PCMU, PCMA, and OPUS. The latest versions of X-Lite and Bria 5 >>> also crash with mod_av enabled. >>> >>> Bria 4 and other models of ip phones (Polycom, Yealink, Grandstream) >>> don’t seem to have a problem. >>> >>> Any ideas what could be wrong? >>> >>> - Joe Barrero >>> >>> , >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Fri Sep 8 01:18:06 2017 From: mike at jerris.com (Michael Jerris) Date: Fri, 08 Sep 2017 01:18:06 +0000 Subject: [Freeswitch-users] Cisco IP Phones & Counterpath Softphones Crashing w/ mod_av Enabled In-Reply-To: References: Message-ID: As i said, if those clients are crashing you should contact those vendors. On Thu, Sep 7, 2017 at 9:15 PM Joseph Barrero wrote: > To clarify a few details: > > I'm running on Debian 8 (Jessie). The Cisco phones crash by lighting up, > ringing continuously, and the website interface, dialpad, and buttons > become unresponsive. The phone must be unplugged. The latest version of > X-Lite and Bria 5 crash on Mac OS X crashes with the Error that displays: > > *Thread 6 Crashed:: PhoneInterface* > 0 libsystem_kernel.dylib 0x00007fffd873ad42 __pthread_kill + 10 > 1 libsystem_pthread.dylib 0x00007fffd8828457 pthread_kill + 90 > 2 libsystem_c.dylib 0x00007fffd86a04bb __abort + 140 > 3 libsystem_c.dylib 0x00007fffd86a0d7e __stack_chk_fail + 205 > 4 counterpath.CPCAPI2 0x000000010821dd87 > CPCAPI2::SipConversation::ReconConversationManagerImpl::onIncomingParticipant(unsigned > int, resip::SipMessage const&, bool, bool, > recon::ConversationManager::MediaSpecificMismatchInfo const&, > recon::ConversationManager::MediaAttributes const&) + 3633 > 5 ??? 0x0d44667062384453 0 + 956001653779612755 > > > > On Thu, Sep 7, 2017 at 4:41 PM, Luis Azedo > wrote: > >> i noticed that freeswitch crashes with mod_av enabled in debian stretch >>> >> >> >>> ---------- Forwarded message ---------- >>> From: Michael Jerris >>> To: FreeSWITCH Users Help >>> Cc: >>> Bcc: >>> Date: Thu, 07 Sep 2017 17:00:55 +0000 >>> Subject: Re: [Freeswitch-users] Cisco IP Phones & Counterpath Softphones >>> Crashing w/ mod_av Enabled >>> I haven't tested all those end points but for sure we have used video >>> with latest and beta bria and had it work. If clients are crashing when >>> receiving video in sdp you should consult with those vendors to find out >>> why. >>> >>> On Wed, Sep 6, 2017 at 3:31 AM roy j wrote: >>> >>>> Hi. Just interested, you say they crash, what exactly is happening, >>>> they are rebooting? If so I would ask Cisco about it, it's so strange. >>>> >>>> 05.09.2017, 10:23, "Joseph Barrero" : >>>> >>>> Hi, I’m running Freeswitch v1.6.19. >>>> >>>> When mod_av is enabled, inbound calls to Cisco SPA504g and SPA514g ip >>>> phones crash as they start to ring. The crash seems to occur immediately >>>> after Freeswitch sends SIP Invite with the SDP information that includes >>>> the video details. I don’t have any problems when I disable mod_av and >>>> include PCMU, PCMA, and OPUS. The latest versions of X-Lite and Bria 5 >>>> also crash with mod_av enabled. >>>> >>>> Bria 4 and other models of ip phones (Polycom, Yealink, Grandstream) >>>> don’t seem to have a problem. >>>> >>>> Any ideas what could be wrong? >>>> >>>> - Joe Barrero >>>> >>>> , >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From federico.omoto at gmail.com Fri Sep 8 03:05:12 2017 From: federico.omoto at gmail.com (Federico Omoto) Date: Fri, 8 Sep 2017 00:05:12 -0300 Subject: [Freeswitch-users] mod_avmd doesn't work on parked channels? Message-ID: Hi! I'm trying to use mod_amvd to detect voicemail on a parked channel, but the beep is never detected. On the other hand, if I use mod_amvd on a bridged call it's always detected. Is mod_amvd usable on a parked channel? Thanks in advance and best regards. Fede -------------- next part -------------- An HTML attachment was scrubbed... URL: From tculjaga at gmail.com Fri Sep 8 08:40:16 2017 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Fri, 8 Sep 2017 10:40:16 +0200 Subject: [Freeswitch-users] use of information with query 'sofia status' and locks in internal interface In-Reply-To: <02f501d32803$0a4037b0$1ec0a710$@smartic.es> References: <02f501d32803$0a4037b0$1ec0a710$@smartic.es> Message-ID: it is sofia_priofile_t profile->inuse On 7 September 2017 at 19:59, Miguel Jesús López Valverde < mjlopez at smartic.es> wrote: > Hello, good afternoon. > > I send this email to see if anyone can give me information about the > meaning of the values (2) after the RUNNING state obtained on the query > sofia status. I searched for forums and books, but I find nothing > explanatory. > > > > freeswitch at ip-xxx-xxx-xxx-xxx> sofia status > > Name Type > Data State > > ============================================================ > ===================================== > > external profile > sip:mod_sofia at xxx.xxx.xxx.xxx:5080 RUNNING (0) > > external profile > sip:mod_sofia at xxx.xxx.xxx.xxx:5081 RUNNING (0) (TLS) > > internal profile > sip:mod_sofia at xxx.xxx.xxx.xxx:5060 RUNNING (2) > > internal profile > sip:mod_sofia at xxx.xxx.xxx.xxx:5061 RUNNING (2) (TLS) > > ============================================================ > ===================================== > > 2 profiles 0 aliases > > > > It happens sometimes that this value rises above 2 in the internal > interface, I do not know if it is casual, but it blocks that interface and > does not respond to sip requests. > > > > To identify this problem I have traversed the log files and nothing > appears that indicates what may be happening and it becomes necessary to > pull and lift the freeswitch service to re-handle the sip requests > correctly. Also, from console, if the sofia status profile internal command > is executed, no response is obtained and the query must be canceled. > > > > Thank you very much. > > > > > > > > [image: cid:image007.png at 01D05CBE.43448510] > > *Miguel J. López Valverde* > Dpto. Técnico. > ------------------------------ > > Pº de la Castellana, 135 7ª pl > 28046 Madrid España > > *Tel:* 900 900 368 | *Móvil:* (+34) 667 772 911 > <+34%20667%2077%2029%2011> > > *E-mail:* mjlopez at smartic.es | *Web:* http://www.smartic.es > ------------------------------ > > *[image: cid:image002.png at 01CFEA0F.BA85E5A0]* > *[image: > cid:image003.png at 01CFEA0F.BA85E5A0]* *[image: > cid:image004.png at 01CFEA0F.BA85E5A0]* > > > > Sus datos de carácter personal (nombre, apellidos, dirección postal y de > correo electrónico, etc.) son tratados para la gestión de su relación con > la Entidad, así como para el envío de información sobre nuestra actividad y > la de terceros relacionadas con la actividad de Consulting Smartic > Solutions, S.L., CIF: B85130037, Pº de la Castellana, 135 7ª pl 28046 > Madrid. Usted puede ejercer sus derechos de acceso, rectificación, > cancelación y oposición dirigiéndose por escrito, con copia de un documento > que acredite su identidad, a la dirección info (arroba) smartic.es. Este > mensaje puede contener información confidencial. Si usted no es su > destinatario, no debe leerlo, copiarlo, distribuirlo, ni hacer uso de la > información que contiene. En este caso, por favor, llámenos o > comuníquenoslo por escrito y borre este mensaje de su sistema > > > > > Libre > de virus. www.avast.com > > <#m_3132687556219528161_DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... 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Name: image003.jpg Type: image/jpeg Size: 1201 bytes Desc: not available URL: From tculjaga at gmail.com Fri Sep 8 08:48:53 2017 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Fri, 8 Sep 2017 10:48:53 +0200 Subject: [Freeswitch-users] Multiple network interfaces In-Reply-To: <46B8D610-1A05-43B2-874F-49286A4804FD@tm.net.uk> References: <9438E820-1B36-47F1-9963-522961F4852F@tm.net.uk> <46B8D610-1A05-43B2-874F-49286A4804FD@tm.net.uk> Message-ID: FS has internal and external profiles .. check that under conf/sip_profiles/ Both xml files (internal.xml and external.xml) are heavily commented and you will find its easy to bind the IP address:port properly. regards, Tihomir. On 7 September 2017 at 12:09, Joseph Waite wrote: > To set the bind ip on the sofia profile you would edit > conf/sip_profiles/external.xml or internal.xml depending on which profile, > or both if you want both profiles to use the external ip. > > You then need to find the line and > change the value to be the internal IP. > You may also need to change the if > you need to ensure the media stream comes from the same IP, which is likely > in your use case. > > Regards > > > On 6 Sep 2017, at 18:16, Caio Assis wrote: > > Could you give me an XML file example? > > > ------------------------------ > *From:* FreeSWITCH-users > on behalf of Jospeh Waite > *Sent:* Wednesday, September 6, 2017 11:58 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Multiple network interfaces > > Depending on your use case, You could create a separate Sofia profile and > bind that to your internal address, then in the dial plan specify that > profile when placing the call to that specific gateway. > Alternatively if you wanted Freeswitch to only use the internal IP for > everything you could specify that all your Sofia profiles bind to the > internal address. > > Regards > > On 6 Sep 2017, at 15:26, Caio Assis wrote: > > Good morning. > > On my freeswitch system, I have 2 network interfaces, one with a public > IP address and one with a internal IP address. The issue is that when I > make a call, freeswitch uses the public address, but I have to use the > internal IP address. Only the internal IP address can connect to a > specific gateway, but all attempts try to use the public address. With > Asterisk I didn't have that issue, for I used the 'bindaddr' param. Can you > help me? > > Thanks. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From tg-maillistings at level5.de Fri Sep 8 09:15:32 2017 From: tg-maillistings at level5.de (=?UTF-8?Q?Thorsten_G=c3=b6llner?=) Date: Fri, 8 Sep 2017 11:15:32 +0200 Subject: [Freeswitch-users] Implement PushKit or CallKit to call iOS app closed In-Reply-To: References: Message-ID: Hi, we currently try to implement it via a hook. When a new channel is created, we call a lua script which checks, if a notification has to be done and returns after finishing. After that Freeswitch can handle the call as normal. Take a look at lua.conf.xml : This lua script queries a database to determine, if a push notification has to be sent and does so if necessary. This database is previously filled with data (such as the needed devicetoken) when a ios device registeres to our FS. Currently we are in developing process and I am not sure, if it is a stable solution. Best regards, Thorsten Am 07.09.2017 um 17:02 schrieb Jose Fco. Irles Durá: > We have implemented a push system to iOS/Android devices with > FreeSWITCH following the next steps: > > 1. Incomming call to user bob in FreeSWITCH > 2. FreeSWITCH park the call > 3. Our manager software (xml_curl) sends a push to bob's iOS device > 4. Notification arrives to iOS device and the softphone call to a uri > sended in the push notification (this uri is the server that > previously has parked the call) > 5. FreeSWITCH bridge the two legs (before this, manage the "ringing" event) > > The FreeSWITCH part is developed in a simple script in lua. > > Best regards > > > > 2017-09-07 10:55 GMT+02:00 Jose David Jurado Alonso : >> Hi, >> >> I can't find any information over how to implement PushKit or Callkit to >> call an iOS mobile phone that has the SIP client application closed or in >> the background. >> >> When the application is closed it is obviously that SIP account isn't >> registered and the FS server no send the call. >> >> Has anyone worked with this or know how to implement it? >> >> I tried with "> value="NDLB-connectile-dysfunction"/>" option and this temporally solved >> when the application is alive in background. >> >> Thanks, >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > From luis.azedo at factorlusitano.com Fri Sep 8 09:20:57 2017 From: luis.azedo at factorlusitano.com (Luis Azedo) Date: Fri, 8 Sep 2017 10:20:57 +0100 Subject: [Freeswitch-users] Cisco IP Phones & Counterpath Softphones Crashing w/ mod_av Enabled Message-ID: Hi Michael, i didn't report because stretch is not supported yet. and the crash seemed related to external libraries. Best -------------- next part -------------- An HTML attachment was scrubbed... URL: From s.safarov at gmail.com Fri Sep 8 09:24:17 2017 From: s.safarov at gmail.com (Sergey Safarov) Date: Fri, 08 Sep 2017 09:24:17 +0000 Subject: [Freeswitch-users] Multiple network interfaces In-Reply-To: References: Message-ID: Please look https://freeswitch.org/jira/browse/FS-10653 This is related to your case Sergey ср, 6 сент. 2017 г. в 17:49, Caio Assis : > Good morning. > > > On my freeswitch system, I have 2 network interfaces, one with a public > IP address and one with a internal IP address. The issue is that when I > make a call, freeswitch uses the public address, but I have to use the > internal IP address. Only the internal IP address can connect to a > specific gateway, but all attempts try to use the public address. With > Asterisk I didn't have that issue, for I used the 'bindaddr' param. Can you > help me? > > > Thanks. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From italo at freeswitch.org Fri Sep 8 12:04:06 2017 From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=) Date: Fri, 8 Sep 2017 09:04:06 -0300 Subject: [Freeswitch-users] Greenswitch ESL for Python In-Reply-To: <1501491856.437006.1057954392.6A534CBA@webmail.messagingengine.com> References: <1501456653.311283.1057571504.5FE3D6C2@webmail.messagingengine.com> <1501491856.437006.1057954392.6A534CBA@webmail.messagingengine.com> Message-ID: Outbound socket is being implemented at https://github.com/EvoluxBR/greenswitch/tree/outboundsocket Feel free to try it and post feedbacks there. On Mon, Jul 31, 2017 at 6:04 AM, Michael Avers wrote: > Code-wise they both work in a similar fashion, but I think where > Greenswitch shines is in its use of Gevent. > > Mike > > > On Sun, Jul 30, 2017, at 11:42 PM, Joel Serrano wrote: > > I'm actually in the process of testing https://github.com/sjthomason/PyESL from > Spencer Thomason, he shared it with the list around a year ago: > > http://lists.freeswitch.org/pipermail/freeswitch-users/ > 2016-August/122148.html > > I didn't know about GreenSWITCH so I'll check it out too. > > Joel. > > > > > > > > On Sun, Jul 30, 2017 at 4:17 PM, Michael Avers > wrote: > > Hello, > > I'm looking at greenswitch as an alternative Python inbound ESL connector. > Other than Italo's company (thanks for releasing this BTW), anyone else has > seen success using it in a moderate to heavy call environment? > > Project is at https://github.com/EvoluxBR/greenswitch > > Thanks > Mike > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > *_________________________________________________________________________* > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Ítalo Rossi italo at freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Fri Sep 8 15:38:14 2017 From: mike at jerris.com (Michael Jerris) Date: Fri, 8 Sep 2017 11:38:14 -0400 Subject: [Freeswitch-users] mod_avmd doesn't work on parked channels? In-Reply-To: References: Message-ID: I would expect the behavior you are describing. May work if you are playing media to the channel, even if its silence. > On Sep 7, 2017, at 11:05 PM, Federico Omoto wrote: > > Hi! > > I'm trying to use mod_amvd to detect voicemail on a parked channel, but the beep is never detected. On the other hand, if I use mod_amvd on a bridged call it's always detected. > > Is mod_amvd usable on a parked channel? > > Thanks in advance and best regards. > > Fede From josedavid at zennio.com Fri Sep 8 08:47:08 2017 From: josedavid at zennio.com (Jose David Jurado Alonso) Date: Fri, 8 Sep 2017 10:47:08 +0200 Subject: [Freeswitch-users] Implement PushKit or CallKit to call iOS app closed In-Reply-To: References: Message-ID: Hi Jose, Thank you very much for commenting your experience with this case. I don't find much information about it. Can you explain me a bit more in detail how you parked the call? It will do the LUA script I suppose. If you could show or send me the LUA script I would be of great help ... Thanks, 2017-09-07 17:02 GMT+02:00 Jose Fco. Irles Durá : > We have implemented a push system to iOS/Android devices with > FreeSWITCH following the next steps: > > 1. Incomming call to user bob in FreeSWITCH > 2. FreeSWITCH park the call > 3. Our manager software (xml_curl) sends a push to bob's iOS device > 4. Notification arrives to iOS device and the softphone call to a uri > sended in the push notification (this uri is the server that > previously has parked the call) > 5. FreeSWITCH bridge the two legs (before this, manage the "ringing" event) > > The FreeSWITCH part is developed in a simple script in lua. > > Best regards > > > > 2017-09-07 10:55 GMT+02:00 Jose David Jurado Alonso >: > > Hi, > > > > I can't find any information over how to implement PushKit or Callkit to > > call an iOS mobile phone that has the SIP client application closed or in > > the background. > > > > When the application is closed it is obviously that SIP account isn't > > registered and the FS server no send the call. > > > > Has anyone worked with this or know how to implement it? > > > > I tried with " > value="NDLB-connectile-dysfunction"/>" option and this temporally solved > > when the application is alive in background. > > > > Thanks, > > > > > > > > ____________________________________________________________ > _____________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > -- > Jose Fco. Irles Durá > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From mario_fs at mgtech.com Fri Sep 8 16:14:14 2017 From: mario_fs at mgtech.com (Mario G) Date: Fri, 8 Sep 2017 09:14:14 -0700 Subject: [Freeswitch-users] Possible enhancement - load mod_logfile way earlier In-Reply-To: References: <760B20D3-CE69-4BD0-AAAE-CC860A3327AF@mgtech.com> Message-ID: I attached the console and free switch logs in https://freeswitch.org/jira/browse/FS-10621 where you can see over 100 startup messages missing from freeswitch.log. Mario G > On Sep 7, 2017, at 9:41 AM, Michael Jerris wrote: > > looking at our default config, it starts pretty early. What configs are you using where it starts late? > > https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/conf/vanilla/autoload_configs/modules.conf.xml > > On Mon, Aug 28, 2017 at 3:15 PM Mario G > wrote: > While working on an issue, I realized I was missing a lot of the startup log. The reason is that FreeSWITCH is started in the background with -NC and only the log file it writes is all that’s available. Problem is mod_logfile is loaded long after other things and all messages prior to mod_logfile are lost. Loading mod_logfile as soon as possible at a startup would prevent all the lost startup messages. Let me know if you want a jira. > Mario G > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From rocketsbay at gmail.com Fri Sep 8 15:40:13 2017 From: rocketsbay at gmail.com (Bayani Carbone) Date: Fri, 08 Sep 2017 15:40:13 +0000 Subject: [Freeswitch-users] xmlrpc originate failure processing Message-ID: Hi all, I'm using xmlrpc with php to send the originate command to freeswitch to initiate a call to leg-a. This command also specifies the dialplan in which I use the bridge command to make a call to leg-b and bridge both legs. xmlrpc command: originate {originate_timeout=120,origination_caller_id_name='+xxxxx',origination_caller_id_number=+xxxxxx,call_ref='xxxxxxx',bCallerIDName='+xxxxxxx',bCallerID='+xxxxxx',bnumber='+xxxxxxx'}sofia/gateway/siplb-01/+xxxxx|sofia/gateway/siplb-02/+xxxxxx call XML public dialplan: My problem is that if the originate command results in a call setup failure, I don't enter the dialplan and I can't retrieve the failure in php either. I tried to add the following to the originate command: transfer_on_fail='auto_cause error XML public' With corresponding dialplan entry but it did not work either. I need to be able to execute eoc.lua even on failure of leg-A. Any help is welcome! Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: From federico.omoto at gmail.com Fri Sep 8 22:05:11 2017 From: federico.omoto at gmail.com (Federico Omoto) Date: Fri, 8 Sep 2017 19:05:11 -0300 Subject: [Freeswitch-users] mod_avmd doesn't work on parked channels? In-Reply-To: References: Message-ID: Thanks for the confirmation! If I play media to the channel it'd be detected as long as the beep occurs during the playback, right? Regards, Fede El 8 sept. 2017 12:38 PM, "Michael Jerris" escribió: I would expect the behavior you are describing. May work if you are playing media to the channel, even if its silence. > On Sep 7, 2017, at 11:05 PM, Federico Omoto wrote: > > Hi! > > I'm trying to use mod_amvd to detect voicemail on a parked channel, but the beep is never detected. On the other hand, if I use mod_amvd on a bridged call it's always detected. > > Is mod_amvd usable on a parked channel? > > Thanks in advance and best regards. > > Fede _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Fri Sep 8 22:07:52 2017 From: mike at jerris.com (Michael Jerris) Date: Fri, 8 Sep 2017 18:07:52 -0400 Subject: [Freeswitch-users] mod_avmd doesn't work on parked channels? In-Reply-To: References: Message-ID: <581CB3D9-A6B6-4C6A-AA66-32B312576006@jerris.com> unclear on why you’d want it parked at this point… what exactly are you trying to do. > On Sep 8, 2017, at 6:05 PM, Federico Omoto wrote: > > Thanks for the confirmation! > If I play media to the channel it'd be detected as long as the beep occurs during the playback, right? > > Regards, > Fede > > > El 8 sept. 2017 12:38 PM, "Michael Jerris" > escribió: > I would expect the behavior you are describing. May work if you are playing media to the channel, even if its silence. > > > On Sep 7, 2017, at 11:05 PM, Federico Omoto > wrote: > > > > Hi! > > > > I'm trying to use mod_amvd to detect voicemail on a parked channel, but the beep is never detected. On the other hand, if I use mod_amvd on a bridged call it's always detected. > > > > Is mod_amvd usable on a parked channel? > > > > Thanks in advance and best regards. > > > > Fede > -------------- next part -------------- An HTML attachment was scrubbed... URL: From federico.omoto at gmail.com Fri Sep 8 22:16:08 2017 From: federico.omoto at gmail.com (Federico Omoto) Date: Fri, 8 Sep 2017 19:16:08 -0300 Subject: [Freeswitch-users] mod_avmd doesn't work on parked channels? In-Reply-To: <581CB3D9-A6B6-4C6A-AA66-32B312576006@jerris.com> References: <581CB3D9-A6B6-4C6A-AA66-32B312576006@jerris.com> Message-ID: I have an ESL app that calls a customer, parks the channel and reproduces a short audio; then, another call is originated to one of our agents. When the agent answers, the two channel are bridged. El 8 sept. 2017 7:08 PM, "Michael Jerris" escribió: unclear on why you’d want it parked at this point… what exactly are you trying to do. On Sep 8, 2017, at 6:05 PM, Federico Omoto wrote: Thanks for the confirmation! If I play media to the channel it'd be detected as long as the beep occurs during the playback, right? Regards, Fede El 8 sept. 2017 12:38 PM, "Michael Jerris" escribió: I would expect the behavior you are describing. May work if you are playing media to the channel, even if its silence. > On Sep 7, 2017, at 11:05 PM, Federico Omoto wrote: > > Hi! > > I'm trying to use mod_amvd to detect voicemail on a parked channel, but the beep is never detected. On the other hand, if I use mod_amvd on a bridged call it's always detected. > > Is mod_amvd usable on a parked channel? > > Thanks in advance and best regards. > > Fede _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From saumar at uol.com.br Fri Sep 8 23:52:11 2017 From: saumar at uol.com.br (Saumar Hajjar) Date: Fri, 8 Sep 2017 20:52:11 -0300 Subject: [Freeswitch-users] Brazilian Portuguese ASR Message-ID: Hi, I'm looking for a FreeSWITCH supported ASR engine that works well with Brazilian Portuguese - preferably tested in a production system. Any ideas? Thanks From nandy1925 at gmail.com Sat Sep 9 13:35:02 2017 From: nandy1925 at gmail.com (Nandy Dagondon) Date: Sat, 9 Sep 2017 13:35:02 +0000 Subject: [Freeswitch-users] TDM-to-TDM Pass-through Media Message-ID: Hello folks, I have a setup wherein I inserted FS in-between a PSTN exchange and a PABX linked via E1 R2. My TDM card has dual E1 ports - port A is connected to the exchange and port B to the PABX. If there's a call that pass-through FS (EXCH > PABX and vice-versa), I want incoming media from one port to immediately sent to the other port. What mode should I set in the dialplan? proxy_media or bypass_media? I have read the Wiki on proxy_media. It's very clear to me re handling of TDM media. Thanks in advance for the inputs. /Nandy -------------- next part -------------- An HTML attachment was scrubbed... URL: From tayeb.meftah at gmail.com Sat Sep 9 15:43:53 2017 From: tayeb.meftah at gmail.com (Tayeb Meftah) Date: Sat, 9 Sep 2017 16:43:53 +0100 Subject: [Freeswitch-users] TDM-to-TDM Pass-through Media In-Reply-To: References: Message-ID: according to my knoledges, you cant bypass media in tdm, at freeswitch is handling Analog to digital conversion and so on, its interlink bypassmedia / proxymedia is for rtp based setup at i could get thanks On 9/9/17, Nandy Dagondon wrote: > Hello folks, > > I have a setup wherein I inserted FS in-between a PSTN exchange and a PABX > linked via E1 R2. My TDM card has dual E1 ports - port A is connected to > the exchange and port B to the PABX. If there's a call that pass-through > FS (EXCH > PABX and vice-versa), I want incoming media from one port to > immediately sent to the other port. What mode should I set in the > dialplan? proxy_media or bypass_media? I have read the Wiki on > proxy_media. It's very clear to me re handling of TDM media. > > Thanks in advance for the inputs. > > /Nandy > From krice at freeswitch.org Sat Sep 9 22:03:22 2017 From: krice at freeswitch.org (Ken Rice) Date: Sat, 9 Sep 2017 17:03:22 -0500 Subject: [Freeswitch-users] TDM-to-TDM Pass-through Media In-Reply-To: References: Message-ID: bypass and proxy media is only useful for rtp based calls (read sip to sip) you still have yo copy frames from on t1/e1 port to the other port. Sent from my iPhone > On Sep 9, 2017, at 08:35, Nandy Dagondon wrote: > > Hello folks, > > I have a setup wherein I inserted FS in-between a PSTN exchange and a PABX linked via E1 R2. My TDM card has dual E1 ports - port A is connected to the exchange and port B to the PABX. If there's a call that pass-through FS (EXCH > PABX and vice-versa), I want incoming media from one port to immediately sent to the other port. What mode should I set in the dialplan? proxy_media or bypass_media? I have read the Wiki on proxy_media. It's very clear to me re handling of TDM media. > > Thanks in advance for the inputs. > > /Nandy > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From clive at lansink.co.nz Sun Sep 10 01:04:41 2017 From: clive at lansink.co.nz (Clive Lansink) Date: Sun, 10 Sep 2017 13:04:41 +1200 Subject: [Freeswitch-users] PortAudio and Windows devices Message-ID: An embedded and charset-unspecified text was scrubbed... Name: not available URL: From gregor at infomedia.si Sun Sep 10 05:25:55 2017 From: gregor at infomedia.si (Gregor Nanger) Date: Sun, 10 Sep 2017 07:25:55 +0200 Subject: [Freeswitch-users] PortAudio and Windows devices In-Reply-To: <59b49038.cac66b0a.8882.218cSMTPIN_ADDED_MISSING@mx.google.com> References: <59b49038.cac66b0a.8882.218cSMTPIN_ADDED_MISSING@mx.google.com> Message-ID: Hi, in VS go to Build/Configuration Manager and see if mod_portaudio project is selected for build. Maybe it builds without it? Or post errors you get while building solution. 2017-09-10 3:04 GMT+02:00 Clive Lansink : > Hi again everyone. > > OK I appreciate that nobody has yet answered this but now I have more info > that may help. > > I succeeded in building the Freeswitch executable for Windows from the > source. I git cloned version 1.6.19 which I understand is the most recent > stable release. I am using Visual Studio 2015 Express. > > I first built the debug version 64-bit. There were three errors but at > least there was an executable so I ran that. Now the pa devlist command > lists my sound devices so that works. Also the debug information indicates > that it is using DirectSound. I actually haven't made a call because of > another issue but at least I can confirm that mod_portaudio sees my audio > devices in this situation. > > Then I cleaned the solution and built the release version. Again I got > three errors but again there was an executable so I ran that. Now the pa > devlist command produces no devices. > > This suggests to me that something is amiss with the process for > generating the release version for Windows which also makes the Windows > installer, that impacts on PortAudio. In that case, the result In my > experience is that PortAudio just doesn't work because it doesn't see any > audio devices. > > It's beyond me I'm afraid to figure out the difference here. I'm just not > experienced with Visual Studio to understand what is going on with these > two scenarios. > > Hope this helps. > > > Clive Lansink > Email: Clive at Lansink.Co.NZ > Phone: +64 9 520-4242 > Mobile: +64 21 663-999 > Fax: +64 21 789-150 > > -----Original message----- > From: Clive Lansink > To: "Freeswitch users list" > Subject: [Freeswitch-users] PortAudio and Windows devices > Reply-to: FreeSWITCH Users Help > Date: Fri, 8 Sep 2017 10:03:32 +1200 > > > Hi everyone. Can someone please help with the following problem? I have > spent several hours on this today, just googling around, and although I > think I am not the only one with this problem, I haven't actually found the > solution. > > Problem is PortAudio reports there are no input and output devices. I am > running Freeswitch on Windows 10 64-bit. The Freeswitch installer is > FreeSWITCH-1.6.18-x64-Release.msi which I downloaded recently from > files.freeswitch.org. I should explain that I have been using Freeswitch > for a number of years and PortAudio used to work, even on this Windows 10 > PC. The problem seems to be with the more recent versions of Freeswitch. > > We know that Freeswitch for Windows includes PortAudio. That's good > because I don't know how to build Freeswitch for Windows. I just know how > to install and use it. > > I think I read somewhere that the Freeswitch PortAudio is built for > DirectEx. I came across the Dxdiag tool which tells me about DirectEx > devices. I have several input and output devices on this machine, but > Dxdiag only lists one output device and it lists no input devices. It tells > me I have DirectEx version 12. > > I do a lot of programming in Python and that includes a Python module that > provides bindings for PortAudio. That code still works. > > I read that PortAudio by default is built for MME, or maybe it is PyAudio > that is built for MME. Whatever the case, that code works. > > But if Freeswitch's PortAudio module is built for DirectEx, could there be > a problem with DirectEx on Windows 10? I just get the impression that > Microsoft is favouring newer Windows sound APIs. Issuing the pa devlist > command just lists nothing so it is as if it is just not seeing any > devices, but they are certainly there. > > I think this has been a problem for several years because I have > Freeswitch version 1.5.15 on another Windows 10 64-bit PC and PortAudio on > that PC has the same problem. I just haven't had to use it on that PC. > > By the way, I did try running Freeswitch in console mode with > administrator privileges and it made no difference. > > I hope someone can suggest something to help me fix this problem. > > > Clive Lansink > Email: Clive at Lansink.Co.NZ > Phone: +64 9 520-4242 > Mobile: +64 21 663-999 > Fax: +64 21 789-150 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Gregor Nanger *CTO* t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia • www.infomedia.si -------------- next part -------------- An HTML attachment was scrubbed... URL: From nandy1925 at gmail.com Sun Sep 10 10:12:42 2017 From: nandy1925 at gmail.com (Nandy Dagondon) Date: Sun, 10 Sep 2017 10:12:42 +0000 Subject: [Freeswitch-users] TDM-to-TDM Pass-through Media In-Reply-To: References: Message-ID: Yes. The frames need to be copied from one port to the other. However, FS can still snoop on the media like detecting DTMF like an incoming caller responding to an auto-attendant/IVR. Will using "bypass_media" parameter on the dialplan, the frames will just go from one endpoint (TDM channel) to the other endpoint (TDM channel on the other port) in like manner? Or there are TDM parameters I need to know? This is easy to visualize in SIP because packets just pass through the switches/routers - not through FS. Appreciate for more additional info. /Nandy On Sat, Sep 9, 2017 at 10:03 PM, Ken Rice wrote: > bypass and proxy media is only useful for rtp based calls (read sip to > sip) you still have yo copy frames from on t1/e1 port to the other port. > > > > Sent from my iPhone > > > On Sep 9, 2017, at 08:35, Nandy Dagondon wrote: > > > > Hello folks, > > > > I have a setup wherein I inserted FS in-between a PSTN exchange and a > PABX linked via E1 R2. My TDM card has dual E1 ports - port A is connected > to the exchange and port B to the PABX. If there's a call that > pass-through FS (EXCH > PABX and vice-versa), I want incoming media from > one port to immediately sent to the other port. What mode should I set in > the dialplan? proxy_media or bypass_media? I have read the Wiki on > proxy_media. It's very clear to me re handling of TDM media. > > > > Thanks in advance for the inputs. > > > > /Nandy > > > > ____________________________________________________________ > _____________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From rbetancor at gmail.com Sun Sep 10 10:21:38 2017 From: rbetancor at gmail.com (=?UTF-8?Q?Ra=C3=BAl_Alexis_Betancor_Santana?=) Date: Sun, 10 Sep 2017 11:21:38 +0100 Subject: [Freeswitch-users] TDM-to-TDM Pass-through Media In-Reply-To: References: Message-ID: But in this case, TDM frames are ALLWAYS going throught FS, and ALLWAYS means ALLWAYS, every single frame of the TDM-TDM call. You don't need to use any param on the dialplan, as you could not offload FS of the TDM-frames proccessing. 2017-09-10 11:12 GMT+01:00 Nandy Dagondon : > Yes. The frames need to be copied from one port to the other. However, FS > can still snoop on the media like detecting DTMF like an incoming caller > responding to an auto-attendant/IVR. Will using "bypass_media" parameter > on the dialplan, the frames will just go from one endpoint (TDM channel) to > the other endpoint (TDM channel on the other port) in like manner? Or > there are TDM parameters I need to know? > > This is easy to visualize in SIP because packets just pass through the > switches/routers - not through FS. > > Appreciate for more additional info. > > /Nandy > > > > > > On Sat, Sep 9, 2017 at 10:03 PM, Ken Rice wrote: > >> bypass and proxy media is only useful for rtp based calls (read sip to >> sip) you still have yo copy frames from on t1/e1 port to the other port. >> >> >> >> Sent from my iPhone >> >> > On Sep 9, 2017, at 08:35, Nandy Dagondon wrote: >> > >> > Hello folks, >> > >> > I have a setup wherein I inserted FS in-between a PSTN exchange and a >> PABX linked via E1 R2. My TDM card has dual E1 ports - port A is connected >> to the exchange and port B to the PABX. If there's a call that >> pass-through FS (EXCH > PABX and vice-versa), I want incoming media from >> one port to immediately sent to the other port. What mode should I set in >> the dialplan? proxy_media or bypass_media? I have read the Wiki on >> proxy_media. It's very clear to me re handling of TDM media. >> > >> > Thanks in advance for the inputs. >> > >> > /Nandy >> > >> > ____________________________________________________________ >> _____________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://confluence.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ >> freeswitch-users >> > http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From nandy1925 at gmail.com Sun Sep 10 10:45:03 2017 From: nandy1925 at gmail.com (Nandy Dagondon) Date: Sun, 10 Sep 2017 10:45:03 +0000 Subject: [Freeswitch-users] TDM-to-TDM Pass-through Media In-Reply-To: References: Message-ID: Hi Raul, I know because freetdm drivers run in the FS hardware. So, frames always passthrough the server. I also notice these FreeTDM applications: enable_dtmf, disable_dtmf and disable_ec. Perhaps I can test them. Tks, /Nandy On Sun, Sep 10, 2017 at 10:21 AM, Raúl Alexis Betancor Santana < rbetancor at gmail.com> wrote: > But in this case, TDM frames are ALLWAYS going throught FS, and ALLWAYS > means ALLWAYS, every single frame of the TDM-TDM call. > > You don't need to use any param on the dialplan, as you could not offload > FS of the TDM-frames proccessing. > > 2017-09-10 11:12 GMT+01:00 Nandy Dagondon : > >> Yes. The frames need to be copied from one port to the other. However, FS >> can still snoop on the media like detecting DTMF like an incoming caller >> responding to an auto-attendant/IVR. Will using "bypass_media" parameter >> on the dialplan, the frames will just go from one endpoint (TDM channel) to >> the other endpoint (TDM channel on the other port) in like manner? Or >> there are TDM parameters I need to know? >> >> This is easy to visualize in SIP because packets just pass through the >> switches/routers - not through FS. >> >> Appreciate for more additional info. >> >> /Nandy >> >> >> >> >> >> On Sat, Sep 9, 2017 at 10:03 PM, Ken Rice wrote: >> >>> bypass and proxy media is only useful for rtp based calls (read sip to >>> sip) you still have yo copy frames from on t1/e1 port to the other port. >>> >>> >>> >>> Sent from my iPhone >>> >>> > On Sep 9, 2017, at 08:35, Nandy Dagondon wrote: >>> > >>> > Hello folks, >>> > >>> > I have a setup wherein I inserted FS in-between a PSTN exchange and a >>> PABX linked via E1 R2. My TDM card has dual E1 ports - port A is connected >>> to the exchange and port B to the PABX. If there's a call that >>> pass-through FS (EXCH > PABX and vice-versa), I want incoming media from >>> one port to immediately sent to the other port. What mode should I set in >>> the dialplan? proxy_media or bypass_media? I have read the Wiki on >>> proxy_media. It's very clear to me re handling of TDM media. >>> > >>> > Thanks in advance for the inputs. >>> > >>> > /Nandy >>> > >>> > ____________________________________________________________ >>> _____________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting at freeswitch.org >>> > http://www.freeswitchsolutions.com >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://confluence.freeswitch.org >>> > http://www.cluecon.com >>> > >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>> switch-users >>> > http://www.freeswitch.org >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From krice at freeswitch.org Sun Sep 10 13:25:51 2017 From: krice at freeswitch.org (Ken Rice) Date: Sun, 10 Sep 2017 08:25:51 -0500 Subject: [Freeswitch-users] TDM-to-TDM Pass-through Media In-Reply-To: References: Message-ID: again there is no bypass or proxy media for TDM. bypass media does only works with sofia it is a special mode. proxy media is also only for sofia. Sent from my iPhone > On Sep 10, 2017, at 05:12, Nandy Dagondon wrote: > > Yes. The frames need to be copied from one port to the other. However, FS can still snoop on the media like detecting DTMF like an incoming caller responding to an auto-attendant/IVR. Will using "bypass_media" parameter on the dialplan, the frames will just go from one endpoint (TDM channel) to the other endpoint (TDM channel on the other port) in like manner? Or there are TDM parameters I need to know? > > This is easy to visualize in SIP because packets just pass through the switches/routers - not through FS. > > Appreciate for more additional info. > > /Nandy > > > > > >> On Sat, Sep 9, 2017 at 10:03 PM, Ken Rice wrote: >> bypass and proxy media is only useful for rtp based calls (read sip to sip) you still have yo copy frames from on t1/e1 port to the other port. >> >> >> >> Sent from my iPhone >> >> > On Sep 9, 2017, at 08:35, Nandy Dagondon wrote: >> > >> > Hello folks, >> > >> > I have a setup wherein I inserted FS in-between a PSTN exchange and a PABX linked via E1 R2. My TDM card has dual E1 ports - port A is connected to the exchange and port B to the PABX. If there's a call that pass-through FS (EXCH > PABX and vice-versa), I want incoming media from one port to immediately sent to the other port. What mode should I set in the dialplan? proxy_media or bypass_media? I have read the Wiki on proxy_media. It's very clear to me re handling of TDM media. >> > >> > Thanks in advance for the inputs. >> > >> > /Nandy >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://confluence.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From nandy1925 at gmail.com Mon Sep 11 03:46:16 2017 From: nandy1925 at gmail.com (Nandy Dagondon) Date: Mon, 11 Sep 2017 03:46:16 +0000 Subject: [Freeswitch-users] TDM-to-TDM Pass-through Media In-Reply-To: References: Message-ID: Okay Ken. Just believe the guru. :-) Thanks On Sun, Sep 10, 2017 at 1:25 PM, Ken Rice wrote: > again there is no bypass or proxy media for TDM. bypass media does only > works with sofia it is a special mode. proxy media is also only for sofia. > > Sent from my iPhone > > On Sep 10, 2017, at 05:12, Nandy Dagondon wrote: > > Yes. The frames need to be copied from one port to the other. However, FS > can still snoop on the media like detecting DTMF like an incoming caller > responding to an auto-attendant/IVR. Will using "bypass_media" parameter > on the dialplan, the frames will just go from one endpoint (TDM channel) to > the other endpoint (TDM channel on the other port) in like manner? Or > there are TDM parameters I need to know? > > This is easy to visualize in SIP because packets just pass through the > switches/routers - not through FS. > > Appreciate for more additional info. > > /Nandy > > > > > > On Sat, Sep 9, 2017 at 10:03 PM, Ken Rice wrote: > >> bypass and proxy media is only useful for rtp based calls (read sip to >> sip) you still have yo copy frames from on t1/e1 port to the other port. >> >> >> >> Sent from my iPhone >> >> > On Sep 9, 2017, at 08:35, Nandy Dagondon wrote: >> > >> > Hello folks, >> > >> > I have a setup wherein I inserted FS in-between a PSTN exchange and a >> PABX linked via E1 R2. My TDM card has dual E1 ports - port A is connected >> to the exchange and port B to the PABX. If there's a call that >> pass-through FS (EXCH > PABX and vice-versa), I want incoming media from >> one port to immediately sent to the other port. What mode should I set in >> the dialplan? proxy_media or bypass_media? I have read the Wiki on >> proxy_media. It's very clear to me re handling of TDM media. >> > >> > Thanks in advance for the inputs. >> > >> > /Nandy >> > >> > ____________________________________________________________ >> _____________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://confluence.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ >> freeswitch-users >> > http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From yu at yu-boot.ru Mon Sep 11 04:46:00 2017 From: yu at yu-boot.ru (yu at yu-boot.ru) Date: Mon, 11 Sep 2017 07:46:00 +0300 Subject: [Freeswitch-users] =?utf-8?q?TDM-to-TDM_Pass-through_Media?= In-Reply-To: References: Message-ID: <1505105160.200609772@f14.my.com> You can do exactly what you want with hardware TDM gateways like Mediant or dirt-cheap old Cisco routers like 28xx or 26xx with T1/E1 boards. -- sent from myMail for Android понедельник, 11 сентября 2017г., 06:46 +03:00 от Nandy Dagondon : Okay Ken.  Just believe the guru.  :-) Thanks On Sun, Sep 10, 2017 at 1:25 PM, Ken Rice wrote: >again there is no bypass or proxy media for TDM. bypass media does only works with sofia it is a special mode. proxy media is also only for sofia.  >Sent from my iPhone >On Sep 10, 2017, at 05:12, Nandy Dagondon wrote: >>Yes. The frames need to be copied from one port to the other. However, FS can still snoop on the media like detecting DTMF like an incoming caller responding to an auto-attendant/IVR.  Will using "bypass_media" parameter on the dialplan, the frames will just go from one endpoint (TDM channel) to the other endpoint (TDM channel on the other port) in like manner?  Or there are TDM parameters I need to know? >>This is easy to visualize in SIP because packets just pass through the switches/routers - not through FS.  >>Appreciate for more additional info. >>/Nandy >>On Sat, Sep 9, 2017 at 10:03 PM, Ken Rice wrote: >>>bypass and proxy media is only useful for rtp based calls (read sip to sip) you still have yo copy frames from on t1/e1 port to the other port. >>> Sent from my iPhone >>> > On Sep 9, 2017, at 08:35, Nandy Dagondon wrote: >>> > >>> > Hello folks, >>> > >>> > I have a setup wherein I inserted FS in-between a PSTN exchange and a PABX linked via E1 R2.  My TDM card has dual E1 ports - port A is connected to the exchange and port B to the PABX.  If there's a call that pass-through FS (EXCH > PABX and vice-versa), I want incoming media from one port to immediately sent to the other port.  What mode should I set in the dialplan?  proxy_media or bypass_media?  I have read the Wiki on proxy_media. It's very clear to me re handling of TDM media. >>> -------------- next part -------------- An HTML attachment was scrubbed... URL: From kkothari157 at gmail.com Mon Sep 11 05:30:40 2017 From: kkothari157 at gmail.com (Ketan Kothari) Date: Mon, 11 Sep 2017 11:00:40 +0530 Subject: [Freeswitch-users] feature access code for call transfer Message-ID: Hello there, In Asterisk i'm using *2 as call transfer and its working fine. *Is there any feature access code for call transfer for manually?* I have tested call transfer using button which available in soft-phone and its working fine. But i want to transfer manually without using button. Any suggestion would be helpful. -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Mon Sep 11 06:18:05 2017 From: david.villasmil.work at gmail.com (David Villasmil) Date: Mon, 11 Sep 2017 08:18:05 +0200 Subject: [Freeswitch-users] xmlrpc originate failure processing In-Reply-To: References: Message-ID: Have you tried hangup_after_bridge to false? Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 On Fri, Sep 8, 2017 at 5:40 PM, Bayani Carbone wrote: > Hi all, > > I'm using xmlrpc with php to send the originate command to freeswitch to > initiate a call to leg-a. This command also specifies the dialplan in which > I use the bridge command to make a call to leg-b and bridge both legs. > > xmlrpc command: > originate {originate_timeout=120,origination_caller_id_name='+ > xxxxx',origination_caller_id_number=+xxxxxx,call_ref=' > xxxxxxx',bCallerIDName='+xxxxxxx',bCallerID='+xxxxxx', > bnumber='+xxxxxxx'}sofia/gateway/siplb-01/+xxxxx|sofia/gateway/siplb-02/+xxxxxx > call XML public > > dialplan: > > > > > > > > My problem is that if the originate command results in a call setup > failure, I don't enter the dialplan and I can't retrieve the failure in php > either. > > I tried to add the following to the originate command: > transfer_on_fail='auto_cause error XML public' > > With corresponding dialplan entry but it did not work either. > > I need to be able to execute eoc.lua even on failure of leg-A. > > Any help is welcome! > > Thanks > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > ᐧ -------------- next part -------------- An HTML attachment was scrubbed... URL: From covici at ccs.covici.com Mon Sep 11 06:19:09 2017 From: covici at ccs.covici.com (John Covici) Date: Mon, 11 Sep 2017 02:19:09 -0400 Subject: [Freeswitch-users] feature access code for call transfer In-Reply-To: References: Message-ID: How about bind_digit_action? On Mon, 11 Sep 2017 01:30:40 -0400, Ketan Kothari wrote: > > [1 ] > [1.1 ] > [1.2 ] > Hello there, > > In Asterisk i'm using *2 as call transfer and its working fine. > > Is there any feature access code for call transfer for manually? > I have tested call transfer using button which available in soft-phone and its working fine. But i want to transfer manually without using button. > > Any suggestion would be helpful. > [2 ] > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici at ccs.covici.com From d.mordovin at dwide.com Mon Sep 11 07:04:55 2017 From: d.mordovin at dwide.com (Dmitry Mordovin) Date: Mon, 11 Sep 2017 10:04:55 +0300 Subject: [Freeswitch-users] Strange FROM in SIP header Message-ID: Hello, I set ANI number like this: [origination_caller_id_name="1234567",origination_caller_id_number=1234567] in bridge. Then in trace I see Invite message where From looks like /From: "1234567" ;tag=ypt0Kt3Q0em7j/ Anybody knows why *sip:FreeSWITCH* instead *sip:1234567* ? BR Dmitry -------------- next part -------------- An HTML attachment was scrubbed... URL: From mjlopez at smartic.es Mon Sep 11 07:32:15 2017 From: mjlopez at smartic.es (=?UTF-8?Q?Miguel_Jes=C3=BAs_L=C3=B3pez_Valverde?=) Date: Mon, 11 Sep 2017 09:32:15 +0200 Subject: [Freeswitch-users] use of information with query 'sofia status' and locks in internal interface In-Reply-To: References: <02f501d32803$0a4037b0$1ec0a710$@smartic.es> Message-ID: <011d01d32ad0$183fb9b0$48bf2d10$@smartic.es> Thank you very much for your answer. Do you know that indicates the Running state with the values 0 to X? Regards Miguel J. López Valverde Dpto. Técnico. _____ Pº de la Castellana, 135 7ª pl 28046 Madrid España Tel: 900 900 368 | Móvil: (+34) 667 772 911 E-mail: mjlopez at smartic.es | Web: http://www.smartic.es _____ Sus datos de carácter personal (nombre, apellidos, dirección postal y de correo electrónico, etc.) son tratados para la gestión de su relación con la Entidad, así como para el envío de información sobre nuestra actividad y la de terceros relacionadas con la actividad de Consulting Smartic Solutions, S.L., CIF: B85130037, Pº de la Castellana, 135 7ª pl 28046 Madrid. Usted puede ejercer sus derechos de acceso, rectificación, cancelación y oposición dirigiéndose por escrito, con copia de un documento que acredite su identidad, a la dirección info (arroba) smartic.es. Este mensaje puede contener información confidencial. Si usted no es su destinatario, no debe leerlo, copiarlo, distribuirlo, ni hacer uso de la información que contiene. En este caso, por favor, llámenos o comuníquenoslo por escrito y borre este mensaje de su sistema De: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] En nombre de Tihomir Culjaga Enviado el: viernes, 08 de septiembre de 2017 10:40 Para: FreeSWITCH Users Help Asunto: Re: [Freeswitch-users] use of information with query 'sofia status' and locks in internal interface it is sofia_priofile_t profile->inuse On 7 September 2017 at 19:59, Miguel Jesús López Valverde > wrote: Hello, good afternoon. I send this email to see if anyone can give me information about the meaning of the values (2) after the RUNNING state obtained on the query sofia status. I searched for forums and books, but I find nothing explanatory. freeswitch at ip-xxx-xxx-xxx-xxx> sofia status Name Type Data State ================================================================================================= external profile sip:mod_sofia at xxx.xxx.xxx.xxx:5080 RUNNING (0) external profile sip:mod_sofia at xxx.xxx.xxx.xxx:5081 RUNNING (0) (TLS) internal profile sip:mod_sofia at xxx.xxx.xxx.xxx:5060 RUNNING (2) internal profile sip:mod_sofia at xxx.xxx.xxx.xxx:5061 RUNNING (2) (TLS) ================================================================================================= 2 profiles 0 aliases It happens sometimes that this value rises above 2 in the internal interface, I do not know if it is casual, but it blocks that interface and does not respond to sip requests. To identify this problem I have traversed the log files and nothing appears that indicates what may be happening and it becomes necessary to pull and lift the freeswitch service to re-handle the sip requests correctly. Also, from console, if the sofia status profile internal command is executed, no response is obtained and the query must be canceled. Thank you very much. Miguel J. López Valverde Dpto. Técnico. _____ Pº de la Castellana, 135 7ª pl 28046 Madrid España Tel: 900 900 368 | Móvil: (+34) 667 772 911 E-mail: mjlopez at smartic.es | Web: http://www.smartic.es _____ Sus datos de carácter personal (nombre, apellidos, dirección postal y de correo electrónico, etc.) son tratados para la gestión de su relación con la Entidad, así como para el envío de información sobre nuestra actividad y la de terceros relacionadas con la actividad de Consulting Smartic Solutions, S.L., CIF: B85130037, Pº de la Castellana, 135 7ª pl 28046 Madrid. Usted puede ejercer sus derechos de acceso, rectificación, cancelación y oposición dirigiéndose por escrito, con copia de un documento que acredite su identidad, a la dirección info (arroba) smartic.es . Este mensaje puede contener información confidencial. Si usted no es su destinatario, no debe leerlo, copiarlo, distribuirlo, ni hacer uso de la información que contiene. En este caso, por favor, llámenos o comuníquenoslo por escrito y borre este mensaje de su sistema Libre de virus. www.avast.com _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org --- El software de antivirus Avast ha analizado este correo electrónico en busca de virus. https://www.avast.com/antivirus -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... 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Name: image004.jpg Type: image/jpeg Size: 1184 bytes Desc: not available URL: From alexandr.popov at iqoption.com Mon Sep 11 08:54:27 2017 From: alexandr.popov at iqoption.com (Alexandr Popov) Date: Mon, 11 Sep 2017 11:54:27 +0300 Subject: [Freeswitch-users] use of information with query 'sofia status' and locks in internal interface In-Reply-To: <011d01d32ad0$183fb9b0$48bf2d10$@smartic.es> References: <02f501d32803$0a4037b0$1ec0a710$@smartic.es> <011d01d32ad0$183fb9b0$48bf2d10$@smartic.es> Message-ID: amount of channels 2017-09-11 10:32 GMT+03:00 Miguel Jesús López Valverde : > Thank you very much for your answer. > > > > Do you know that indicates the Running state with the values 0 to X? > > > > Regards > > > > > > [image: cid:image007.png at 01D05CBE.43448510] > > *Miguel J. López Valverde* > Dpto. Técnico. > ------------------------------ > > Pº de la Castellana, 135 7ª pl > 28046 Madrid España > > *Tel:* 900 900 368 | *Móvil:* (+34) 667 772 911 > > *E-mail:* mjlopez at smartic.es | *Web:* http://www.smartic.es > ------------------------------ > > *[image: cid:image002.png at 01CFEA0F.BA85E5A0]* > *[image: > cid:image003.png at 01CFEA0F.BA85E5A0]* *[image: > cid:image004.png at 01CFEA0F.BA85E5A0]* > > > > Sus datos de carácter personal (nombre, apellidos, dirección postal y de > correo electrónico, etc.) son tratados para la gestión de su relación con > la Entidad, así como para el envío de información sobre nuestra actividad y > la de terceros relacionadas con la actividad de Consulting Smartic > Solutions, S.L., CIF: B85130037, Pº de la Castellana, 135 7ª pl 28046 > Madrid. Usted puede ejercer sus derechos de acceso, rectificación, > cancelación y oposición dirigiéndose por escrito, con copia de un documento > que acredite su identidad, a la dirección info (arroba) smartic.es. Este > mensaje puede contener información confidencial. Si usted no es su > destinatario, no debe leerlo, copiarlo, distribuirlo, ni hacer uso de la > información que contiene. En este caso, por favor, llámenos o > comuníquenoslo por escrito y borre este mensaje de su sistema > > > > *De:* FreeSWITCH-users [mailto:freeswitch-users- > bounces at lists.freeswitch.org] *En nombre de *Tihomir Culjaga > *Enviado el:* viernes, 08 de septiembre de 2017 10:40 > *Para:* FreeSWITCH Users Help > *Asunto:* Re: [Freeswitch-users] use of information with query 'sofia > status' and locks in internal interface > > > > it is sofia_priofile_t profile->inuse > > > > > > On 7 September 2017 at 19:59, Miguel Jesús López Valverde < > mjlopez at smartic.es> wrote: > > Hello, good afternoon. > > I send this email to see if anyone can give me information about the > meaning of the values (2) after the RUNNING state obtained on the query > sofia status. I searched for forums and books, but I find nothing > explanatory. > > > > freeswitch at ip-xxx-xxx-xxx-xxx> sofia status > > Name Type > Data State > > ============================================================ > ===================================== > > external profile > sip:mod_sofia at xxx.xxx.xxx.xxx:5080 RUNNING (0) > > external profile > sip:mod_sofia at xxx.xxx.xxx.xxx:5081 RUNNING (0) (TLS) > > internal profile > sip:mod_sofia at xxx.xxx.xxx.xxx:5060 RUNNING (2) > > internal profile > sip:mod_sofia at xxx.xxx.xxx.xxx:5061 RUNNING (2) (TLS) > > ============================================================ > ===================================== > > 2 profiles 0 aliases > > > > It happens sometimes that this value rises above 2 in the internal > interface, I do not know if it is casual, but it blocks that interface and > does not respond to sip requests. > > > > To identify this problem I have traversed the log files and nothing > appears that indicates what may be happening and it becomes necessary to > pull and lift the freeswitch service to re-handle the sip requests > correctly. Also, from console, if the sofia status profile internal command > is executed, no response is obtained and the query must be canceled. > > > > Thank you very much. > > > > > > > > [image: cid:image007.png at 01D05CBE.43448510] > > *Miguel J. López Valverde* > Dpto. Técnico. > ------------------------------ > > Pº de la Castellana, 135 7ª pl > 28046 Madrid España > > *Tel:* 900 900 368 | *Móvil:* (+34) 667 772 911 > <+34%20667%2077%2029%2011> > > *E-mail:* mjlopez at smartic.es | *Web:* http://www.smartic.es > ------------------------------ > > *[image: cid:image002.png at 01CFEA0F.BA85E5A0]* > *[image: > cid:image003.png at 01CFEA0F.BA85E5A0]* *[image: > cid:image004.png at 01CFEA0F.BA85E5A0]* > > > > Sus datos de carácter personal (nombre, apellidos, dirección postal y de > correo electrónico, etc.) son tratados para la gestión de su relación con > la Entidad, así como para el envío de información sobre nuestra actividad y > la de terceros relacionadas con la actividad de Consulting Smartic > Solutions, S.L., CIF: B85130037, Pº de la Castellana, 135 7ª pl 28046 > Madrid. Usted puede ejercer sus derechos de acceso, rectificación, > cancelación y oposición dirigiéndose por escrito, con copia de un documento > que acredite su identidad, a la dirección info (arroba) smartic.es. Este > mensaje puede contener información confidencial. Si usted no es su > destinatario, no debe leerlo, copiarlo, distribuirlo, ni hacer uso de la > información que contiene. En este caso, por favor, llámenos o > comuníquenoslo por escrito y borre este mensaje de su sistema > > > > > > > > > Libre de virus. www.avast.com > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... 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Name: image004.jpg Type: image/jpeg Size: 1184 bytes Desc: not available URL: From kkothari157 at gmail.com Mon Sep 11 09:32:03 2017 From: kkothari157 at gmail.com (Ketan Kothari) Date: Mon, 11 Sep 2017 15:02:03 +0530 Subject: [Freeswitch-users] feature access code for call transfer In-Reply-To: References: Message-ID: Hello John, I mean is there *any kind of access code for call transfer* which we can use while manual call transfer? -------------- next part -------------- An HTML attachment was scrubbed... URL: From covici at ccs.covici.com Mon Sep 11 09:43:45 2017 From: covici at ccs.covici.com (John Covici) Date: Mon, 11 Sep 2017 05:43:45 -0400 Subject: [Freeswitch-users] feature access code for call transfer In-Reply-To: References: Message-ID: You can use bind_digit_action to make one yourself. On Mon, 11 Sep 2017 05:32:03 -0400, Ketan Kothari wrote: > > [1 ] > [1.1 ] > [1.2 ] > Hello John, > > I mean is there any kind of access code for call transfer which we can use while manual call transfer? > [2 ] > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici at ccs.covici.com From mjlopez at smartic.es Mon Sep 11 10:08:09 2017 From: mjlopez at smartic.es (=?UTF-8?Q?Miguel_Jes=C3=BAs_L=C3=B3pez_Valverde?=) Date: Mon, 11 Sep 2017 12:08:09 +0200 Subject: [Freeswitch-users] use of information with query 'sofia status' and locks in internal interface In-Reply-To: References: <02f501d32803$0a4037b0$1ec0a710$@smartic.es> <011d01d32ad0$183fb9b0$48bf2d10$@smartic.es> Message-ID: <019001d32ae5$df16ff20$9d44fd60$@smartic.es> Ok, tank you! Miguel J. López Valverde Dpto. Técnico. _____ Pº de la Castellana, 135 7ª pl 28046 Madrid España Tel: 900 900 368 | Móvil: (+34) 667 772 911 E-mail: mjlopez at smartic.es | Web: http://www.smartic.es _____ Sus datos de carácter personal (nombre, apellidos, dirección postal y de correo electrónico, etc.) son tratados para la gestión de su relación con la Entidad, así como para el envío de información sobre nuestra actividad y la de terceros relacionadas con la actividad de Consulting Smartic Solutions, S.L., CIF: B85130037, Pº de la Castellana, 135 7ª pl 28046 Madrid. Usted puede ejercer sus derechos de acceso, rectificación, cancelación y oposición dirigiéndose por escrito, con copia de un documento que acredite su identidad, a la dirección info (arroba) smartic.es. Este mensaje puede contener información confidencial. Si usted no es su destinatario, no debe leerlo, copiarlo, distribuirlo, ni hacer uso de la información que contiene. En este caso, por favor, llámenos o comuníquenoslo por escrito y borre este mensaje de su sistema De: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] En nombre de Alexandr Popov Enviado el: lunes, 11 de septiembre de 2017 10:54 Para: FreeSWITCH Users Help Asunto: Re: [Freeswitch-users] use of information with query 'sofia status' and locks in internal interface amount of channels 2017-09-11 10:32 GMT+03:00 Miguel Jesús López Valverde >: Thank you very much for your answer. Do you know that indicates the Running state with the values 0 to X? Regards Miguel J. López Valverde Dpto. Técnico. _____ Pº de la Castellana, 135 7ª pl 28046 Madrid España Tel: 900 900 368 | Móvil: (+34) 667 772 911 E-mail: mjlopez at smartic.es | Web: http://www.smartic.es _____ Sus datos de carácter personal (nombre, apellidos, dirección postal y de correo electrónico, etc.) son tratados para la gestión de su relación con la Entidad, así como para el envío de información sobre nuestra actividad y la de terceros relacionadas con la actividad de Consulting Smartic Solutions, S.L., CIF: B85130037, Pº de la Castellana, 135 7ª pl 28046 Madrid. Usted puede ejercer sus derechos de acceso, rectificación, cancelación y oposición dirigiéndose por escrito, con copia de un documento que acredite su identidad, a la dirección info (arroba) smartic.es . Este mensaje puede contener información confidencial. Si usted no es su destinatario, no debe leerlo, copiarlo, distribuirlo, ni hacer uso de la información que contiene. En este caso, por favor, llámenos o comuníquenoslo por escrito y borre este mensaje de su sistema De: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org ] En nombre de Tihomir Culjaga Enviado el: viernes, 08 de septiembre de 2017 10:40 Para: FreeSWITCH Users Help > Asunto: Re: [Freeswitch-users] use of information with query 'sofia status' and locks in internal interface it is sofia_priofile_t profile->inuse On 7 September 2017 at 19:59, Miguel Jesús López Valverde > wrote: Hello, good afternoon. I send this email to see if anyone can give me information about the meaning of the values (2) after the RUNNING state obtained on the query sofia status. I searched for forums and books, but I find nothing explanatory. freeswitch at ip-xxx-xxx-xxx-xxx> sofia status Name Type Data State ================================================================================================= external profile sip:mod_sofia at xxx.xxx.xxx.xxx:5080 RUNNING (0) external profile sip:mod_sofia at xxx.xxx.xxx.xxx:5081 RUNNING (0) (TLS) internal profile sip:mod_sofia at xxx.xxx.xxx.xxx:5060 RUNNING (2) internal profile sip:mod_sofia at xxx.xxx.xxx.xxx:5061 RUNNING (2) (TLS) ================================================================================================= 2 profiles 0 aliases It happens sometimes that this value rises above 2 in the internal interface, I do not know if it is casual, but it blocks that interface and does not respond to sip requests. To identify this problem I have traversed the log files and nothing appears that indicates what may be happening and it becomes necessary to pull and lift the freeswitch service to re-handle the sip requests correctly. Also, from console, if the sofia status profile internal command is executed, no response is obtained and the query must be canceled. Thank you very much. Miguel J. López Valverde Dpto. Técnico. _____ Pº de la Castellana, 135 7ª pl 28046 Madrid España Tel: 900 900 368 | Móvil: (+34) 667 772 911 E-mail: mjlopez at smartic.es | Web: http://www.smartic.es _____ Sus datos de carácter personal (nombre, apellidos, dirección postal y de correo electrónico, etc.) son tratados para la gestión de su relación con la Entidad, así como para el envío de información sobre nuestra actividad y la de terceros relacionadas con la actividad de Consulting Smartic Solutions, S.L., CIF: B85130037, Pº de la Castellana, 135 7ª pl 28046 Madrid. Usted puede ejercer sus derechos de acceso, rectificación, cancelación y oposición dirigiéndose por escrito, con copia de un documento que acredite su identidad, a la dirección info (arroba) smartic.es . Este mensaje puede contener información confidencial. Si usted no es su destinatario, no debe leerlo, copiarlo, distribuirlo, ni hacer uso de la información que contiene. En este caso, por favor, llámenos o comuníquenoslo por escrito y borre este mensaje de su sistema Libre de virus. www.avast.com _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org --- El software de antivirus Avast ha analizado este correo electrónico en busca de virus. https://www.avast.com/antivirus -------------- next part -------------- An HTML attachment was scrubbed... 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Name: image004.jpg Type: image/jpeg Size: 1184 bytes Desc: not available URL: From lists at pbaines.com Mon Sep 11 11:28:41 2017 From: lists at pbaines.com (Peter Baines) Date: Mon, 11 Sep 2017 12:28:41 +0100 Subject: [Freeswitch-users] 2 person video conference layout Message-ID: Hello, I am trying to create a conference layout for a two person conference only where they only see each other. i.e. Person A sees Person B Person B sees Person A Is this possible with the conference layout system? Regards, Peter -------------- next part -------------- An HTML attachment was scrubbed... URL: From mahdi_shirazi at yahoo.com Sun Sep 10 07:31:34 2017 From: mahdi_shirazi at yahoo.com (Mehdi Shirazi) Date: Sun, 10 Sep 2017 07:31:34 +0000 (UTC) Subject: [Freeswitch-users] TDM-to-TDM Pass-through Media References: <1598190912.6253044.1505028694077.ref@mail.yahoo.com> Message-ID: <1598190912.6253044.1505028694077@mail.yahoo.com> Hi Regarding Freeswitch shares same DAHDI and libpri with Asterisk, if there was no reason to call comes inside server ( like codec conversion ,inband DTMF processing , call waiting)... Theoretically this two case should be possible: 1-In DAHDI-driven cards supports direct timeslot bridging (interchange)(E1/T1s in same span cards) the audio won't even pass across the system bus at all. http://lists.digium.com/pipermail/asterisk-dev/2010-March/043056.html 2-With another interesting service = 2BCT(Two B-Channel Transfer)(ITU version ECT = Explicit Call Transfer) even it is possible to free both B channels and transfer Call to Local switch( if both sides was on same E1/T1 span) http://support.digium.com/Answers?id=kA080000000GwmUCAS Regards M.Shirazi >bypass and proxy media is only useful for rtp based calls (read sip to sip) you still have yo copy frames >from on t1/e1 port to the other port. >Sent from my iPhone >> On Sep 9, 2017, at 08:35, Nandy Dagondon wrote: >> >> Hello folks, >> >> I have a setup wherein I inserted FS in-between a PSTN exchange and a PABX linked via E1 R2. My TDM card >>has dual E1 ports - port A is connected to the exchange and port B to the PABX. If there's a call that pass->>through FS (EXCH > PABX and vice-versa), I want incoming media from one port to immediately sent to the >>other port. What mode should I set in the dialplan? proxy_media or bypass_media? I have read the Wiki on >>proxy_media. It's very clear to me re handling of TDM media. >> >> Thanks in advance for the inputs. >> >> /Nandy From npaleari at phonecallsrl.com.ar Sat Sep 9 13:16:27 2017 From: npaleari at phonecallsrl.com.ar (Nicolas Paleari) Date: Sat, 9 Sep 2017 10:16:27 -0300 Subject: [Freeswitch-users] Stop packet radius In-Reply-To: References: <814EFC40-F9E1-438B-A2FD-CA8E92984490@tm.net.uk> <609287ea-c63f-9655-9d4b-a0c238797033@phonecallsrl.com.ar> <6888FB36-04AC-43A2-8077-151BB1925244@tm.net.uk> Message-ID: Dear Joseph, thanks for the reply, the password is not real, but you're right, I should have deleted the field, it is not necessary. As for the solution I will try and answer the results, thanks El 9 sep. 2017 9:38 AM, "Joseph Waite" escribió: Ok Firstly, you should always redact and sensitive parts of config files before posting to a public user group. And Hacker now has the access credential for your radius server, I would advise changing the password immediately. This line should have had the IP and password blanked out to look like Hope this helps. Regards On 6 Sep 2017, at 18:39, Phonecall SRL - Nicolas Paleari < npaleari at phonecallsrl.com.ar> wrote: Hi, copy the contents of the file: Best regards El 6/9/2017 a las 11:59 a. m., Jospeh Waite escribió: Could you supply a copy of your xml_radius.conf.xml? Might have a clue in there. On 6 Sep 2017, at 12:03, Nicolas Paleari wrote: Hello, change does Freeswitch, I understand why it does a coding, I need to receive #, this is how the carrier calls waiting El 6 sep. 2017 7:08 AM, "Joseph Waite" escribió: > I'm your radius config are you using the same variable to set both? > > If not and they will always be the same, may be worth trying changing to > use the same. That way can figure out if it's the radius module or > freeswitch making the change. > > Joe Waite > > On 5 Sep 2017, at 14:22, Nicolas Paleari > wrote: > > Friends, I need help, I get incorrect information in the stop packet of > raduis that sends freeswitch, in the field dst-number-out the number has > %23 instead of #, I do not understand why it replaces it,dst-number-out should > be equal to field dst-number-in = 492482 # 541151995330 > Send stop package where you see the problem: > > > [2017-09-01 19:03:46,119] DEBUG WORKER 38 - [pkt#175515/ACCT-STOP] > --- START: [pkt#175515/ACCT-STOP] -------------------------------------- > pdd-time => 0 > called-station-id => 492482%23541151995330 > nas-port => 0 > dst-number-out => 492482%23541151995330 > h323-conf-id => cd0f2a0d-4b7f-414c-b5be-d1b937a414db > h323-setup-time => 2017-09-01T15:03:36.328927-0400 > acct-status-type => Stop > h323-disconnect-cause => 10 > dst-gw-ip => 190.210.240.37 > h323-disconnect-time => 2017-09-01T15:03:43.929039-0400 > h323-call-id => 7a34a7be-d643-4f82-a66a-cd1eb0658b6c > src-gw-ip => 190.210.240.37 > dst-number-in => 492482#541151995330 > h323-call-origin => originate > nas-ip-address => 107.170.35.75 > src-number-in => 7680858053 > dst-gw-name => 492482#541151995330 > h323-connect-time => 2017-09-01T15:03:38.268917-0400 > src-gw-name => 7680858053 > acct-session-time => 5 > acct-delay-time => 0 > src-number-out => 7680858053 > calling-station-id => 7680858053 > --- END: [pkt#175515/ACCT-STOP] -------------------------------------- > > I hope you can help me > Thank you > > > Libre > de virus. www.avast.com > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.com Mon Sep 11 14:37:19 2017 From: brian at freeswitch.com (Brian West) Date: Mon, 11 Sep 2017 09:37:19 -0500 Subject: [Freeswitch-users] 2 person video conference layout In-Reply-To: References: Message-ID: Why are you using a conference for a bridged call? You can just call A to B. /b On Mon, Sep 11, 2017 at 6:28 AM, Peter Baines wrote: > Hello, > > I am trying to create a conference layout for a two person conference only > where they only see each other. > > i.e. > > Person A sees Person B > Person B sees Person A > > Is this possible with the conference layout system? > > Regards, > Peter > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.com *Twitter: @FreeSWITCH , @cluecon* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) -------------- next part -------------- An HTML attachment was scrubbed... URL: From igorolhovskiy at gmail.com Mon Sep 11 14:41:13 2017 From: igorolhovskiy at gmail.com (Igor Olhovskiy) Date: Mon, 11 Sep 2017 17:41:13 +0300 Subject: [Freeswitch-users] Strange FROM in SIP header In-Reply-To: References: Message-ID: gateway parameter callerid-in-from might help you Regards, Igor On 11 сент. 2017 г., 10:05 +0300, Dmitry Mordovin , wrote: > Hello, > I set ANI number like this: [origination_caller_id_name="1234567",origination_caller_id_number=1234567] in bridge. > Then in trace I see Invite message where From looks like > From: "1234567" ;tag=ypt0Kt3Q0em7j > > Anybody knows why sip:FreeSWITCH instead sip:1234567 ? > > BR > Dmitry > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Mon Sep 11 16:23:12 2017 From: david.villasmil.work at gmail.com (David Villasmil) Date: Mon, 11 Sep 2017 18:23:12 +0200 Subject: [Freeswitch-users] Careful out there, guys.. Message-ID: I got this on a kamailio box: From: a'or'3=3--;tag=cdecd76f Of course, nothing happened, just pointing it out. Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 ᐧ -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Mon Sep 11 17:35:49 2017 From: mike at jerris.com (Michael Jerris) Date: Mon, 11 Sep 2017 13:35:49 -0400 Subject: [Freeswitch-users] Careful out there, guys.. In-Reply-To: References: Message-ID: We’ve been seeing this attack for 4-5 months now… its out there in the wild for sure. > On Sep 11, 2017, at 12:23 PM, David Villasmil wrote: > > I got this on a kamailio box: > > From: a'or'3=3-->;tag=cdecd76f > > Of course, nothing happened, just pointing it out. -------------- next part -------------- An HTML attachment was scrubbed... URL: From saumar at uol.com.br Mon Sep 11 19:25:47 2017 From: saumar at uol.com.br (Saumar Hajjar) Date: Mon, 11 Sep 2017 16:25:47 -0300 Subject: [Freeswitch-users] Brazilian Portuguese ASR In-Reply-To: <020601d32b2f$37cee6c0$a76cb440$@pontimax.com> References: <020601d32b2f$37cee6c0$a76cb440$@pontimax.com> Message-ID: <048ae756-90a0-4381-28d3-9713185cd6e3@uol.com.br> Good to know there's a server based license. I'm currently evaluating unimrcp server + Google SR plugin. I'll contact Pontimax and request a trial license. Thank you. Saumar Em 11/09/2017 15:53, Ray Keating escreveu: > Hi Hajjar, > > Pontimax's MRCP/SP11-STT MRCPV1 ASR server is working in production with FreeSWITCH (and Asterisk). Since our speech recognition engine is Microsoft's Speech Platform 11, all of the twenty six Microsoft language modules are freely available with use with it, including Portuguese-Brazil (pr-BR). > > We have a production user in Brazil (an Asterisk user) that has reported excellent results using the Portuguese-Brazil language module. I can provide that individual's e-mail upon request, by PM. > > Our latest 1.3 version of MRCP/SP11-STT is just being released for download from www.pontimax.com; a thirty day free trial key can be requested so that you can evaluate it for yourself. A permanent license is only $1,000USD, per machine. > > Regards, > > Ray > > Pontimax’s mrcpSP11-STT- the lowest cost, by far, highest recognition accuracy MRCP server available > > -----Original Message----- > From: Saumar Hajjar [mailto:saumar at uol.com.br] > Sent: Friday, September 8, 2017 7:52 PM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] Brazilian Portuguese ASR > > Hi, > > I'm looking for a FreeSWITCH supported ASR engine that works well with Brazilian Portuguese - preferably tested in a production system. > Any ideas? > > Thanks > > > > > From rayk at pontimax.com Mon Sep 11 19:23:37 2017 From: rayk at pontimax.com (Ray Keating) Date: Mon, 11 Sep 2017 15:23:37 -0400 Subject: [Freeswitch-users] PortAudio and Windows devices In-Reply-To: References: <59b49038.cac66b0a.8882.218cSMTPIN_ADDED_MISSING@mx.google.com> Message-ID: <020901d32b33$7a4783f0$6ed68bd0$@pontimax.com> Completely separate from FS, I’ve recently encountered sound driver issues on Windows 10. You might look in the Event Viewer under Custom Views, Server Roles, Administrative Events and see if there are any Device Manager reports, such as ‘Failed to migrate (driver to Win 10)’. Besides providing drivers that fail to install under Windows 10, there have been some Windows updates that have introduced some annoying driver glitches, such as for the virtual NIC, if you’re a Hyper-V VM user. -Ray Pontimax’s mrcpSP11-STT- the lowest cost, by far, highest recognition accuracy MRCP server available From: Gregor Nanger [mailto:gregor at infomedia.si] Sent: Sunday, September 10, 2017 1:26 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] PortAudio and Windows devices Hi, in VS go to Build/Configuration Manager and see if mod_portaudio project is selected for build. Maybe it builds without it? Or post errors you get while building solution. 2017-09-10 3:04 GMT+02:00 Clive Lansink >: Hi again everyone. OK I appreciate that nobody has yet answered this but now I have more info that may help. I succeeded in building the Freeswitch executable for Windows from the source. I git cloned version 1.6.19 which I understand is the most recent stable release. I am using Visual Studio 2015 Express. I first built the debug version 64-bit. There were three errors but at least there was an executable so I ran that. Now the pa devlist command lists my sound devices so that works. Also the debug information indicates that it is using DirectSound. I actually haven't made a call because of another issue but at least I can confirm that mod_portaudio sees my audio devices in this situation. Then I cleaned the solution and built the release version. Again I got three errors but again there was an executable so I ran that. Now the pa devlist command produces no devices. This suggests to me that something is amiss with the process for generating the release version for Windows which also makes the Windows installer, that impacts on PortAudio. In that case, the result In my experience is that PortAudio just doesn't work because it doesn't see any audio devices. It's beyond me I'm afraid to figure out the difference here. I'm just not experienced with Visual Studio to understand what is going on with these two scenarios. Hope this helps. Clive Lansink Email: Clive at Lansink.Co.NZ Phone: +64 9 520-4242 Mobile: +64 21 663-999 Fax: +64 21 789-150 -----Original message----- From: Clive Lansink > To: "Freeswitch users list" > Subject: [Freeswitch-users] PortAudio and Windows devices Reply-to: FreeSWITCH Users Help > Date: Fri, 8 Sep 2017 10:03:32 +1200 Hi everyone. Can someone please help with the following problem? I have spent several hours on this today, just googling around, and although I think I am not the only one with this problem, I haven't actually found the solution. Problem is PortAudio reports there are no input and output devices. I am running Freeswitch on Windows 10 64-bit. The Freeswitch installer is FreeSWITCH-1.6.18-x64-Release.msi which I downloaded recently from files.freeswitch.org . I should explain that I have been using Freeswitch for a number of years and PortAudio used to work, even on this Windows 10 PC. The problem seems to be with the more recent versions of Freeswitch. We know that Freeswitch for Windows includes PortAudio. That's good because I don't know how to build Freeswitch for Windows. I just know how to install and use it. I think I read somewhere that the Freeswitch PortAudio is built for DirectEx. I came across the Dxdiag tool which tells me about DirectEx devices. I have several input and output devices on this machine, but Dxdiag only lists one output device and it lists no input devices. It tells me I have DirectEx version 12. I do a lot of programming in Python and that includes a Python module that provides bindings for PortAudio. That code still works. I read that PortAudio by default is built for MME, or maybe it is PyAudio that is built for MME. Whatever the case, that code works. But if Freeswitch's PortAudio module is built for DirectEx, could there be a problem with DirectEx on Windows 10? I just get the impression that Microsoft is favouring newer Windows sound APIs. Issuing the pa devlist command just lists nothing so it is as if it is just not seeing any devices, but they are certainly there. I think this has been a problem for several years because I have Freeswitch version 1.5.15 on another Windows 10 64-bit PC and PortAudio on that PC has the same problem. I just haven't had to use it on that PC. By the way, I did try running Freeswitch in console mode with administrator privileges and it made no difference. I hope someone can suggest something to help me fix this problem. Clive Lansink Email: Clive at Lansink.Co.NZ Phone: +64 9 520-4242 Mobile: +64 21 663-999 Fax: +64 21 789-150 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Gregor Nanger CTO t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia • www.infomedia.si -------------- next part -------------- An HTML attachment was scrubbed... URL: From rayk at pontimax.com Mon Sep 11 18:53:07 2017 From: rayk at pontimax.com (Ray Keating) Date: Mon, 11 Sep 2017 14:53:07 -0400 Subject: [Freeswitch-users] Brazilian Portuguese ASR In-Reply-To: References: Message-ID: <020601d32b2f$37cee6c0$a76cb440$@pontimax.com> Hi Hajjar, Pontimax's MRCP/SP11-STT MRCPV1 ASR server is working in production with FreeSWITCH (and Asterisk). Since our speech recognition engine is Microsoft's Speech Platform 11, all of the twenty six Microsoft language modules are freely available with use with it, including Portuguese-Brazil (pr-BR). We have a production user in Brazil (an Asterisk user) that has reported excellent results using the Portuguese-Brazil language module. I can provide that individual's e-mail upon request, by PM. Our latest 1.3 version of MRCP/SP11-STT is just being released for download from www.pontimax.com; a thirty day free trial key can be requested so that you can evaluate it for yourself. A permanent license is only $1,000USD, per machine. Regards, Ray Pontimax’s mrcpSP11-STT- the lowest cost, by far, highest recognition accuracy MRCP server available -----Original Message----- From: Saumar Hajjar [mailto:saumar at uol.com.br] Sent: Friday, September 8, 2017 7:52 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Brazilian Portuguese ASR Hi, I'm looking for a FreeSWITCH supported ASR engine that works well with Brazilian Portuguese - preferably tested in a production system. Any ideas? Thanks From rick at magicmail.mooo.com Mon Sep 11 19:46:20 2017 From: rick at magicmail.mooo.com (Rick Jarvis) Date: Mon, 11 Sep 2017 20:46:20 +0100 Subject: [Freeswitch-users] Verto In-Reply-To: References: <03E75515-780B-451B-A30F-62C99020D8A7@magicmail.mooo.com> Message-ID: <1334A7C4-01C0-47C4-9D5A-E1C3448E2080@magicmail.mooo.com> I have verto up and running (sort of), but I’m confused about whether authentication is working. Chrome’s console is showing ‘authentication required’ errors. Verto debug gives me the below (I have a context ‘vertobox’, which is working with a login to 201 at vertobox from an IP phone; certs / DNS etc are working ok). Is this looking good or not? I can’t see an active registration from the FS console, so it must be failing I guess? 2017-09-11 20:33:45.518527 [ALERT] mod_verto.c:1335 READ 10.1.1.3:63258 [{ "jsonrpc": "2.0", "method": "login", "params": { "sessid": "3462fb9e-5849-9560-a451-1adec02d958d" }, "id": 2 }] 2017-09-11 20:33:45.518527 [INFO] mod_verto.c:1232 10.1.1.3:63258 re-connecting session 3462fb9e-5849-9560-a451-1adec02d958d 2017-09-11 20:33:45.518527 [WARNING] mod_verto.c:1095 New connection for session 3462fb9e-5849-9560-a451-1adec02d958d dropping previous connection. 2017-09-11 20:33:45.518527 [ALERT] mod_verto.c:604 WRITE 10.1.1.3:63246 [{ "jsonrpc": "2.0", "id": 5, "method": "verto.punt", "params": { } }] 2017-09-11 20:33:45.518527 [ALERT] mod_verto.c:604 WRITE 10.1.1.3:63258 [{ "jsonrpc": "2.0", "id": 2, "error": { "code": -32000, "message": "Authentication Required" } }] 2017-09-11 20:33:45.538529 [DEBUG] mod_verto.c:1804 10.1.1.3:63246 Dropping Connection 2017-09-11 20:33:45.538529 [INFO] mod_verto.c:1910 10.1.1.3:63246 Ending client thread. 2017-09-11 20:33:45.538529 [INFO] mod_verto.c:1917 10.1.1.3:63246 Thread ended 2017-09-11 20:33:45.538529 [ALERT] mod_verto.c:1335 READ 10.1.1.3:63258 [{ "jsonrpc": "2.0", "method": "login", "params": { "login": “201 at vertobox", "passwd": "j4guar", "loginParams": { }, "userVariables": { }, "sessid": "3462fb9e-5849-9560-a451-1adec02d958d" }, "id": 3 }] 2017-09-11 20:33:45.538529 [ALERT] mod_verto.c:604 WRITE 10.1.1.3:63258 [{ "jsonrpc": "2.0", "id": 3, "result": { "message": "logged in", "sessid": "3462fb9e-5849-9560-a451-1adec02d958d" } }] > On 7 Sep 2017, at 21:11, Ítalo Rossi wrote: > > Verto communicator is a example implementation on what can be done with Verto. > > Check our tutorial to understand how to build a minimum app: > > https://evoluxbr.github.io/verto-docs/ > Em qui, 7 de set de 2017 às 17:02, Rick Jarvis > escreveu: > Looking to get into Verto, in particular handling voice calls with JS. Going through the source, I’m wondering: > > 1. What’s the difference between the Verto source and the Verto Communicator source? > > 2. What’s the best way to start from the bottom up - by this I mean that it seems hugely comprehensive, but rather than just use grunt to set it all up, I’d like to start simply with the basics… is there for instance a list of the bare minimum scripts / file structure to use? Apologies if this is a silly question, I’m still relatively new to JS and I don’t want to blow my mind in one go ;) > > Rick > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Mon Sep 11 19:52:03 2017 From: mike at jerris.com (Michael Jerris) Date: Mon, 11 Sep 2017 15:52:03 -0400 Subject: [Freeswitch-users] Verto In-Reply-To: <1334A7C4-01C0-47C4-9D5A-E1C3448E2080@magicmail.mooo.com> References: <03E75515-780B-451B-A30F-62C99020D8A7@magicmail.mooo.com> <1334A7C4-01C0-47C4-9D5A-E1C3448E2080@magicmail.mooo.com> Message-ID: <820C03C3-999D-48F5-865E-74E7467E2CDF@jerris.com> That is successfully authenticating from what i see there. > On Sep 11, 2017, at 3:46 PM, Rick Jarvis wrote: > > I have verto up and running (sort of), but I’m confused about whether authentication is working. Chrome’s console is showing ‘authentication required’ errors. Verto debug gives me the below (I have a context ‘vertobox’, which is working with a login to 201 at vertobox from an IP phone; certs / DNS etc are working ok). > > Is this looking good or not? I can’t see an active registration from the FS console, so it must be failing I guess? > > 2017-09-11 20:33:45.518527 [ALERT] mod_verto.c:1335 READ 10.1.1.3:63258 [{ > "jsonrpc": "2.0", > "method": "login", > "params": { > "sessid": "3462fb9e-5849-9560-a451-1adec02d958d" > }, > "id": 2 > }] > 2017-09-11 20:33:45.518527 [INFO] mod_verto.c:1232 10.1.1.3:63258 re-connecting session 3462fb9e-5849-9560-a451-1adec02d958d > 2017-09-11 20:33:45.518527 [WARNING] mod_verto.c:1095 New connection for session 3462fb9e-5849-9560-a451-1adec02d958d dropping previous connection. > 2017-09-11 20:33:45.518527 [ALERT] mod_verto.c:604 WRITE 10.1.1.3:63246 [{ > "jsonrpc": "2.0", > "id": 5, > "method": "verto.punt", > "params": { > } > }] > 2017-09-11 20:33:45.518527 [ALERT] mod_verto.c:604 WRITE 10.1.1.3:63258 [{ > "jsonrpc": "2.0", > "id": 2, > "error": { > "code": -32000, > "message": "Authentication Required" > } > }] > 2017-09-11 20:33:45.538529 [DEBUG] mod_verto.c:1804 10.1.1.3:63246 Dropping Connection > 2017-09-11 20:33:45.538529 [INFO] mod_verto.c:1910 10.1.1.3:63246 Ending client thread. > 2017-09-11 20:33:45.538529 [INFO] mod_verto.c:1917 10.1.1.3:63246 Thread ended > 2017-09-11 20:33:45.538529 [ALERT] mod_verto.c:1335 READ 10.1.1.3:63258 [{ > "jsonrpc": "2.0", > "method": "login", > "params": { > "login": “201 at vertobox", > "passwd": "j4guar", > "loginParams": { > }, > "userVariables": { > }, > "sessid": "3462fb9e-5849-9560-a451-1adec02d958d" > }, > "id": 3 > }] > 2017-09-11 20:33:45.538529 [ALERT] mod_verto.c:604 WRITE 10.1.1.3:63258 [{ > "jsonrpc": "2.0", > "id": 3, > "result": { > "message": "logged in", > "sessid": "3462fb9e-5849-9560-a451-1adec02d958d" > } > }] > > >> On 7 Sep 2017, at 21:11, Ítalo Rossi > wrote: >> >> Verto communicator is a example implementation on what can be done with Verto. >> >> Check our tutorial to understand how to build a minimum app: >> >> https://evoluxbr.github.io/verto-docs/ >> Em qui, 7 de set de 2017 às 17:02, Rick Jarvis > escreveu: >> Looking to get into Verto, in particular handling voice calls with JS. Going through the source, I’m wondering: >> >> 1. What’s the difference between the Verto source and the Verto Communicator source? >> >> 2. What’s the best way to start from the bottom up - by this I mean that it seems hugely comprehensive, but rather than just use grunt to set it all up, I’d like to start simply with the basics… is there for instance a list of the bare minimum scripts / file structure to use? Apologies if this is a silly question, I’m still relatively new to JS and I don’t want to blow my mind in one go ;) >> >> Rick -------------- next part -------------- An HTML attachment was scrubbed... URL: From rick at magicmail.mooo.com Mon Sep 11 19:59:40 2017 From: rick at magicmail.mooo.com (Rick Jarvis) Date: Mon, 11 Sep 2017 20:59:40 +0100 Subject: [Freeswitch-users] Verto In-Reply-To: <820C03C3-999D-48F5-865E-74E7467E2CDF@jerris.com> References: <03E75515-780B-451B-A30F-62C99020D8A7@magicmail.mooo.com> <1334A7C4-01C0-47C4-9D5A-E1C3448E2080@magicmail.mooo.com> <820C03C3-999D-48F5-865E-74E7467E2CDF@jerris.com> Message-ID: Should it not show up in ‘sofia status profile internal reg’ ? I can see the IP phone’s reg in there, and I have multiple registrations enabled (double checked that with another IP phone!)…? > On 11 Sep 2017, at 20:52, Michael Jerris wrote: > > That is successfully authenticating from what i see there. > >> On Sep 11, 2017, at 3:46 PM, Rick Jarvis > wrote: >> >> I have verto up and running (sort of), but I’m confused about whether authentication is working. Chrome’s console is showing ‘authentication required’ errors. Verto debug gives me the below (I have a context ‘vertobox’, which is working with a login to 201 at vertobox from an IP phone; certs / DNS etc are working ok). >> >> Is this looking good or not? I can’t see an active registration from the FS console, so it must be failing I guess? >> >> 2017-09-11 20:33:45.518527 [ALERT] mod_verto.c:1335 READ 10.1.1.3:63258 [{ >> "jsonrpc": "2.0", >> "method": "login", >> "params": { >> "sessid": "3462fb9e-5849-9560-a451-1adec02d958d" >> }, >> "id": 2 >> }] >> 2017-09-11 20:33:45.518527 [INFO] mod_verto.c:1232 10.1.1.3:63258 re-connecting session 3462fb9e-5849-9560-a451-1adec02d958d >> 2017-09-11 20:33:45.518527 [WARNING] mod_verto.c:1095 New connection for session 3462fb9e-5849-9560-a451-1adec02d958d dropping previous connection. >> 2017-09-11 20:33:45.518527 [ALERT] mod_verto.c:604 WRITE 10.1.1.3:63246 [{ >> "jsonrpc": "2.0", >> "id": 5, >> "method": "verto.punt", >> "params": { >> } >> }] >> 2017-09-11 20:33:45.518527 [ALERT] mod_verto.c:604 WRITE 10.1.1.3:63258 [{ >> "jsonrpc": "2.0", >> "id": 2, >> "error": { >> "code": -32000, >> "message": "Authentication Required" >> } >> }] >> 2017-09-11 20:33:45.538529 [DEBUG] mod_verto.c:1804 10.1.1.3:63246 Dropping Connection >> 2017-09-11 20:33:45.538529 [INFO] mod_verto.c:1910 10.1.1.3:63246 Ending client thread. >> 2017-09-11 20:33:45.538529 [INFO] mod_verto.c:1917 10.1.1.3:63246 Thread ended >> 2017-09-11 20:33:45.538529 [ALERT] mod_verto.c:1335 READ 10.1.1.3:63258 [{ >> "jsonrpc": "2.0", >> "method": "login", >> "params": { >> "login": “201 at vertobox", >> "passwd": "j4guar", >> "loginParams": { >> }, >> "userVariables": { >> }, >> "sessid": "3462fb9e-5849-9560-a451-1adec02d958d" >> }, >> "id": 3 >> }] >> 2017-09-11 20:33:45.538529 [ALERT] mod_verto.c:604 WRITE 10.1.1.3:63258 [{ >> "jsonrpc": "2.0", >> "id": 3, >> "result": { >> "message": "logged in", >> "sessid": "3462fb9e-5849-9560-a451-1adec02d958d" >> } >> }] >> >> >>> On 7 Sep 2017, at 21:11, Ítalo Rossi > wrote: >>> >>> Verto communicator is a example implementation on what can be done with Verto. >>> >>> Check our tutorial to understand how to build a minimum app: >>> >>> https://evoluxbr.github.io/verto-docs/ >>> Em qui, 7 de set de 2017 às 17:02, Rick Jarvis > escreveu: >>> Looking to get into Verto, in particular handling voice calls with JS. Going through the source, I’m wondering: >>> >>> 1. What’s the difference between the Verto source and the Verto Communicator source? >>> >>> 2. What’s the best way to start from the bottom up - by this I mean that it seems hugely comprehensive, but rather than just use grunt to set it all up, I’d like to start simply with the basics… is there for instance a list of the bare minimum scripts / file structure to use? Apologies if this is a silly question, I’m still relatively new to JS and I don’t want to blow my mind in one go ;) >>> >>> Rick > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Mon Sep 11 19:35:45 2017 From: mike at jerris.com (Michael Jerris) Date: Mon, 11 Sep 2017 15:35:45 -0400 Subject: [Freeswitch-users] PortAudio and Windows devices In-Reply-To: <59b48fcb.c536240a.776e.d6c5SMTPIN_ADDED_MISSING@mx.google.com> References: <59b48fcb.c536240a.776e.d6c5SMTPIN_ADDED_MISSING@mx.google.com> Message-ID: <57339532-56B7-4990-9653-850F95A5DB00@jerris.com> https://freeswitch.org/jira/browse/FS-10663 fixed in master now. Thanks Andrey! > On Sep 9, 2017, at 9:04 PM, Clive Lansink wrote: > > Hi again everyone. > > OK I appreciate that nobody has yet answered this but now I have more info that may help. > > I succeeded in building the Freeswitch executable for Windows from the source. I git cloned version 1.6.19 which I understand is the most recent stable release. I am using Visual Studio 2015 Express. > > I first built the debug version 64-bit. There were three errors but at least there was an executable so I ran that. Now the pa devlist command lists my sound devices so that works. Also the debug information indicates that it is using DirectSound. I actually haven't made a call because of another issue but at least I can confirm that mod_portaudio sees my audio devices in this situation. > > Then I cleaned the solution and built the release version. Again I got three errors but again there was an executable so I ran that. Now the pa devlist command produces no devices. > > This suggests to me that something is amiss with the process for generating the release version for Windows which also makes the Windows installer, that impacts on PortAudio. In that case, the result In my experience is that PortAudio just doesn't work because it doesn't see any audio devices. > > It's beyond me I'm afraid to figure out the difference here. I'm just not experienced with Visual Studio to understand what is going on with these two scenarios. > > Hope this helps. > > > Clive Lansink > Email: Clive at Lansink.Co.NZ > Phone: +64 9 520-4242 > Mobile: +64 21 663-999 > Fax: +64 21 789-150 > > -----Original message----- > From: Clive Lansink > To: "Freeswitch users list" > Subject: [Freeswitch-users] PortAudio and Windows devices > Reply-to: FreeSWITCH Users Help > Date: Fri, 8 Sep 2017 10:03:32 +1200 > > > Hi everyone. Can someone please help with the following problem? I have spent several hours on this today, just googling around, and although I think I am not the only one with this problem, I haven't actually found the solution. > > Problem is PortAudio reports there are no input and output devices. I am running Freeswitch on Windows 10 64-bit. The Freeswitch installer is FreeSWITCH-1.6.18-x64-Release.msi which I downloaded recently from files.freeswitch.org. I should explain that I have been using Freeswitch for a number of years and PortAudio used to work, even on this Windows 10 PC. The problem seems to be with the more recent versions of Freeswitch. > > We know that Freeswitch for Windows includes PortAudio. That's good because I don't know how to build Freeswitch for Windows. I just know how to install and use it. > > I think I read somewhere that the Freeswitch PortAudio is built for DirectEx. I came across the Dxdiag tool which tells me about DirectEx devices. I have several input and output devices on this machine, but Dxdiag only lists one output device and it lists no input devices. It tells me I have DirectEx version 12. > > I do a lot of programming in Python and that includes a Python module that provides bindings for PortAudio. That code still works. > > I read that PortAudio by default is built for MME, or maybe it is PyAudio that is built for MME. Whatever the case, that code works. > > But if Freeswitch's PortAudio module is built for DirectEx, could there be a problem with DirectEx on Windows 10? I just get the impression that Microsoft is favouring newer Windows sound APIs. Issuing the pa devlist command just lists nothing so it is as if it is just not seeing any devices, but they are certainly there. > > I think this has been a problem for several years because I have Freeswitch version 1.5.15 on another Windows 10 64-bit PC and PortAudio on that PC has the same problem. I just haven't had to use it on that PC. > > By the way, I did try running Freeswitch in console mode with administrator privileges and it made no difference. > > I hope someone can suggest something to help me fix this problem. > > > Clive Lansink > Email: Clive at Lansink.Co.NZ > Phone: +64 9 520-4242 > Mobile: +64 21 663-999 > Fax: +64 21 789-150 > -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Mon Sep 11 20:15:01 2017 From: mike at jerris.com (Michael Jerris) Date: Mon, 11 Sep 2017 16:15:01 -0400 Subject: [Freeswitch-users] Verto In-Reply-To: References: <03E75515-780B-451B-A30F-62C99020D8A7@magicmail.mooo.com> <1334A7C4-01C0-47C4-9D5A-E1C3448E2080@magicmail.mooo.com> <820C03C3-999D-48F5-865E-74E7467E2CDF@jerris.com> Message-ID: <010379EC-6F3C-46E9-A9DA-8519F523DAA1@jerris.com> verto has nothing at all to do with sofia > On Sep 11, 2017, at 3:59 PM, Rick Jarvis wrote: > > Should it not show up in ‘sofia status profile internal reg’ ? I can see the IP phone’s reg in there, and I have multiple registrations enabled (double checked that with another IP phone!)…? > >> On 11 Sep 2017, at 20:52, Michael Jerris > wrote: >> >> That is successfully authenticating from what i see there. >> >>> On Sep 11, 2017, at 3:46 PM, Rick Jarvis > wrote: >>> >>> I have verto up and running (sort of), but I’m confused about whether authentication is working. Chrome’s console is showing ‘authentication required’ errors. Verto debug gives me the below (I have a context ‘vertobox’, which is working with a login to 201 at vertobox from an IP phone; certs / DNS etc are working ok). >>> >>> Is this looking good or not? I can’t see an active registration from the FS console, so it must be failing I guess? >>> >>> 2017-09-11 20:33:45.518527 [ALERT] mod_verto.c:1335 READ 10.1.1.3:63258 [{ >>> "jsonrpc": "2.0", >>> "method": "login", >>> "params": { >>> "sessid": "3462fb9e-5849-9560-a451-1adec02d958d" >>> }, >>> "id": 2 >>> }] >>> 2017-09-11 20:33:45.518527 [INFO] mod_verto.c:1232 10.1.1.3:63258 re-connecting session 3462fb9e-5849-9560-a451-1adec02d958d >>> 2017-09-11 20:33:45.518527 [WARNING] mod_verto.c:1095 New connection for session 3462fb9e-5849-9560-a451-1adec02d958d dropping previous connection. >>> 2017-09-11 20:33:45.518527 [ALERT] mod_verto.c:604 WRITE 10.1.1.3:63246 [{ >>> "jsonrpc": "2.0", >>> "id": 5, >>> "method": "verto.punt", >>> "params": { >>> } >>> }] >>> 2017-09-11 20:33:45.518527 [ALERT] mod_verto.c:604 WRITE 10.1.1.3:63258 [{ >>> "jsonrpc": "2.0", >>> "id": 2, >>> "error": { >>> "code": -32000, >>> "message": "Authentication Required" >>> } >>> }] >>> 2017-09-11 20:33:45.538529 [DEBUG] mod_verto.c:1804 10.1.1.3:63246 Dropping Connection >>> 2017-09-11 20:33:45.538529 [INFO] mod_verto.c:1910 10.1.1.3:63246 Ending client thread. >>> 2017-09-11 20:33:45.538529 [INFO] mod_verto.c:1917 10.1.1.3:63246 Thread ended >>> 2017-09-11 20:33:45.538529 [ALERT] mod_verto.c:1335 READ 10.1.1.3:63258 [{ >>> "jsonrpc": "2.0", >>> "method": "login", >>> "params": { >>> "login": “201 at vertobox", >>> "passwd": "j4guar", >>> "loginParams": { >>> }, >>> "userVariables": { >>> }, >>> "sessid": "3462fb9e-5849-9560-a451-1adec02d958d" >>> }, >>> "id": 3 >>> }] >>> 2017-09-11 20:33:45.538529 [ALERT] mod_verto.c:604 WRITE 10.1.1.3:63258 [{ >>> "jsonrpc": "2.0", >>> "id": 3, >>> "result": { >>> "message": "logged in", >>> "sessid": "3462fb9e-5849-9560-a451-1adec02d958d" >>> } >>> }] >>> >>> >>>> On 7 Sep 2017, at 21:11, Ítalo Rossi > wrote: >>>> >>>> Verto communicator is a example implementation on what can be done with Verto. >>>> >>>> Check our tutorial to understand how to build a minimum app: >>>> >>>> https://evoluxbr.github.io/verto-docs/ >>>> Em qui, 7 de set de 2017 às 17:02, Rick Jarvis > escreveu: >>>> Looking to get into Verto, in particular handling voice calls with JS. Going through the source, I’m wondering: >>>> >>>> 1. What’s the difference between the Verto source and the Verto Communicator source? >>>> >>>> 2. What’s the best way to start from the bottom up - by this I mean that it seems hugely comprehensive, but rather than just use grunt to set it all up, I’d like to start simply with the basics… is there for instance a list of the bare minimum scripts / file structure to use? Apologies if this is a silly question, I’m still relatively new to JS and I don’t want to blow my mind in one go ;) >>>> >>>> Rick -------------- next part -------------- An HTML attachment was scrubbed... URL: From rick at magicmail.mooo.com Mon Sep 11 20:29:55 2017 From: rick at magicmail.mooo.com (Rick Jarvis) Date: Mon, 11 Sep 2017 21:29:55 +0100 Subject: [Freeswitch-users] Verto In-Reply-To: <010379EC-6F3C-46E9-A9DA-8519F523DAA1@jerris.com> References: <03E75515-780B-451B-A30F-62C99020D8A7@magicmail.mooo.com> <1334A7C4-01C0-47C4-9D5A-E1C3448E2080@magicmail.mooo.com> <820C03C3-999D-48F5-865E-74E7467E2CDF@jerris.com> <010379EC-6F3C-46E9-A9DA-8519F523DAA1@jerris.com> Message-ID: <2F114FDD-E440-409A-B756-9ECDCD0B1277@magicmail.mooo.com> Ah! I had thought that all registrations ended up at sofia one way or another, thanks Mike. R > On 11 Sep 2017, at 21:15, Michael Jerris wrote: > > verto has nothing at all to do with sofia > >> On Sep 11, 2017, at 3:59 PM, Rick Jarvis > wrote: >> >> Should it not show up in ‘sofia status profile internal reg’ ? I can see the IP phone’s reg in there, and I have multiple registrations enabled (double checked that with another IP phone!)…? >> >>> On 11 Sep 2017, at 20:52, Michael Jerris > wrote: >>> >>> That is successfully authenticating from what i see there. >>> >>>> On Sep 11, 2017, at 3:46 PM, Rick Jarvis > wrote: >>>> >>>> I have verto up and running (sort of), but I’m confused about whether authentication is working. Chrome’s console is showing ‘authentication required’ errors. Verto debug gives me the below (I have a context ‘vertobox’, which is working with a login to 201 at vertobox from an IP phone; certs / DNS etc are working ok). >>>> >>>> Is this looking good or not? I can’t see an active registration from the FS console, so it must be failing I guess? >>>> >>>> 2017-09-11 20:33:45.518527 [ALERT] mod_verto.c:1335 READ 10.1.1.3:63258 [{ >>>> "jsonrpc": "2.0", >>>> "method": "login", >>>> "params": { >>>> "sessid": "3462fb9e-5849-9560-a451-1adec02d958d" >>>> }, >>>> "id": 2 >>>> }] >>>> 2017-09-11 20:33:45.518527 [INFO] mod_verto.c:1232 10.1.1.3:63258 re-connecting session 3462fb9e-5849-9560-a451-1adec02d958d >>>> 2017-09-11 20:33:45.518527 [WARNING] mod_verto.c:1095 New connection for session 3462fb9e-5849-9560-a451-1adec02d958d dropping previous connection. >>>> 2017-09-11 20:33:45.518527 [ALERT] mod_verto.c:604 WRITE 10.1.1.3:63246 [{ >>>> "jsonrpc": "2.0", >>>> "id": 5, >>>> "method": "verto.punt", >>>> "params": { >>>> } >>>> }] >>>> 2017-09-11 20:33:45.518527 [ALERT] mod_verto.c:604 WRITE 10.1.1.3:63258 [{ >>>> "jsonrpc": "2.0", >>>> "id": 2, >>>> "error": { >>>> "code": -32000, >>>> "message": "Authentication Required" >>>> } >>>> }] >>>> 2017-09-11 20:33:45.538529 [DEBUG] mod_verto.c:1804 10.1.1.3:63246 Dropping Connection >>>> 2017-09-11 20:33:45.538529 [INFO] mod_verto.c:1910 10.1.1.3:63246 Ending client thread. >>>> 2017-09-11 20:33:45.538529 [INFO] mod_verto.c:1917 10.1.1.3:63246 Thread ended >>>> 2017-09-11 20:33:45.538529 [ALERT] mod_verto.c:1335 READ 10.1.1.3:63258 [{ >>>> "jsonrpc": "2.0", >>>> "method": "login", >>>> "params": { >>>> "login": “201 at vertobox", >>>> "passwd": "j4guar", >>>> "loginParams": { >>>> }, >>>> "userVariables": { >>>> }, >>>> "sessid": "3462fb9e-5849-9560-a451-1adec02d958d" >>>> }, >>>> "id": 3 >>>> }] >>>> 2017-09-11 20:33:45.538529 [ALERT] mod_verto.c:604 WRITE 10.1.1.3:63258 [{ >>>> "jsonrpc": "2.0", >>>> "id": 3, >>>> "result": { >>>> "message": "logged in", >>>> "sessid": "3462fb9e-5849-9560-a451-1adec02d958d" >>>> } >>>> }] >>>> >>>> >>>>> On 7 Sep 2017, at 21:11, Ítalo Rossi > wrote: >>>>> >>>>> Verto communicator is a example implementation on what can be done with Verto. >>>>> >>>>> Check our tutorial to understand how to build a minimum app: >>>>> >>>>> https://evoluxbr.github.io/verto-docs/ >>>>> Em qui, 7 de set de 2017 às 17:02, Rick Jarvis > escreveu: >>>>> Looking to get into Verto, in particular handling voice calls with JS. Going through the source, I’m wondering: >>>>> >>>>> 1. What’s the difference between the Verto source and the Verto Communicator source? >>>>> >>>>> 2. What’s the best way to start from the bottom up - by this I mean that it seems hugely comprehensive, but rather than just use grunt to set it all up, I’d like to start simply with the basics… is there for instance a list of the bare minimum scripts / file structure to use? Apologies if this is a silly question, I’m still relatively new to JS and I don’t want to blow my mind in one go ;) >>>>> >>>>> Rick > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From nandy1925 at gmail.com Mon Sep 11 21:22:32 2017 From: nandy1925 at gmail.com (Nandy Dagondon) Date: Mon, 11 Sep 2017 21:22:32 +0000 Subject: [Freeswitch-users] TDM-to-TDM Pass-through Media In-Reply-To: <1505105160.200609772@f14.my.com> References: <1505105160.200609772@f14.my.com> Message-ID: Yah! Why not? Thanks for this tip. On Mon, Sep 11, 2017 at 4:46 AM, wrote: > You can do exactly what you want with hardware TDM gateways like Mediant > or dirt-cheap old Cisco routers like 28xx or 26xx with T1/E1 boards. > > -- > sent from myMail for Android > > понедельник, 11 сентября 2017г., 06:46 +03:00 от Nandy Dagondon < > nandy1925 at gmail.com>: > > Okay Ken. Just believe the guru. :-) Thanks > On Sun, Sep 10, 2017 at 1:25 PM, Ken Rice wrote: > >again there is no bypass or proxy media for TDM. bypass media does only > works with sofia it is a special mode. proxy media is also only for sofia. > >Sent from my iPhone > >On Sep 10, 2017, at 05:12, Nandy Dagondon wrote: > >>Yes. The frames need to be copied from one port to the other. However, > FS can still snoop on the media like detecting DTMF like an incoming caller > responding to an auto-attendant/IVR. Will using "bypass_media" parameter > on the dialplan, the frames will just go from one endpoint (TDM channel) to > the other endpoint (TDM channel on the other port) in like manner? Or > there are TDM parameters I need to know? > >>This is easy to visualize in SIP because packets just pass through the > switches/routers - not through FS. > >>Appreciate for more additional info. > >>/Nandy > >>On Sat, Sep 9, 2017 at 10:03 PM, Ken Rice wrote: > >>>bypass and proxy media is only useful for rtp based calls (read sip to > sip) you still have yo copy frames from on t1/e1 port to the other port. > >>> > Sent from my iPhone > >>> > > On Sep 9, 2017, at 08:35, Nandy Dagondon wrote: > >>> > > > >>> > > Hello folks, > >>> > > > >>> > > I have a setup wherein I inserted FS in-between a PSTN exchange and a > PABX linked via E1 R2. My TDM card has dual E1 ports - port A is connected > to the exchange and port B to the PABX. If there's a call that > pass-through FS (EXCH > PABX and vice-versa), I want incoming media from > one port to immediately sent to the other port. What mode should I set in > the dialplan? proxy_media or bypass_media? I have read the Wiki on > proxy_media. It's very clear to me re handling of TDM media. > >>> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From rick at magicmail.mooo.com Mon Sep 11 23:12:02 2017 From: rick at magicmail.mooo.com (Rick Jarvis) Date: Tue, 12 Sep 2017 00:12:02 +0100 Subject: [Freeswitch-users] Verto In-Reply-To: <2F114FDD-E440-409A-B756-9ECDCD0B1277@magicmail.mooo.com> References: <03E75515-780B-451B-A30F-62C99020D8A7@magicmail.mooo.com> <1334A7C4-01C0-47C4-9D5A-E1C3448E2080@magicmail.mooo.com> <820C03C3-999D-48F5-865E-74E7467E2CDF@jerris.com> <010379EC-6F3C-46E9-A9DA-8519F523DAA1@jerris.com> <2F114FDD-E440-409A-B756-9ECDCD0B1277@magicmail.mooo.com> Message-ID: Confused now. ‘show registrations’ - should this give me *all*, or just the non-sofia ones? ATM I have 2 x ip phone registrations, and sofia is showing both, but ‘show registrations’ gives me one - despite having disconnected Chrome, and leaving the 2 ip phones connected…? > On 11 Sep 2017, at 21:29, Rick Jarvis wrote: > > Ah! I had thought that all registrations ended up at sofia one way or another, thanks Mike. > > R > >> On 11 Sep 2017, at 21:15, Michael Jerris > wrote: >> >> verto has nothing at all to do with sofia >> >>> On Sep 11, 2017, at 3:59 PM, Rick Jarvis > wrote: >>> >>> Should it not show up in ‘sofia status profile internal reg’ ? I can see the IP phone’s reg in there, and I have multiple registrations enabled (double checked that with another IP phone!)…? >>> >>>> On 11 Sep 2017, at 20:52, Michael Jerris > wrote: >>>> >>>> That is successfully authenticating from what i see there. >>>> >>>>> On Sep 11, 2017, at 3:46 PM, Rick Jarvis > wrote: >>>>> >>>>> I have verto up and running (sort of), but I’m confused about whether authentication is working. Chrome’s console is showing ‘authentication required’ errors. Verto debug gives me the below (I have a context ‘vertobox’, which is working with a login to 201 at vertobox from an IP phone; certs / DNS etc are working ok). >>>>> >>>>> Is this looking good or not? I can’t see an active registration from the FS console, so it must be failing I guess? >>>>> >>>>> 2017-09-11 20:33:45.518527 [ALERT] mod_verto.c:1335 READ 10.1.1.3:63258 [{ >>>>> "jsonrpc": "2.0", >>>>> "method": "login", >>>>> "params": { >>>>> "sessid": "3462fb9e-5849-9560-a451-1adec02d958d" >>>>> }, >>>>> "id": 2 >>>>> }] >>>>> 2017-09-11 20:33:45.518527 [INFO] mod_verto.c:1232 10.1.1.3:63258 re-connecting session 3462fb9e-5849-9560-a451-1adec02d958d >>>>> 2017-09-11 20:33:45.518527 [WARNING] mod_verto.c:1095 New connection for session 3462fb9e-5849-9560-a451-1adec02d958d dropping previous connection. >>>>> 2017-09-11 20:33:45.518527 [ALERT] mod_verto.c:604 WRITE 10.1.1.3:63246 [{ >>>>> "jsonrpc": "2.0", >>>>> "id": 5, >>>>> "method": "verto.punt", >>>>> "params": { >>>>> } >>>>> }] >>>>> 2017-09-11 20:33:45.518527 [ALERT] mod_verto.c:604 WRITE 10.1.1.3:63258 [{ >>>>> "jsonrpc": "2.0", >>>>> "id": 2, >>>>> "error": { >>>>> "code": -32000, >>>>> "message": "Authentication Required" >>>>> } >>>>> }] >>>>> 2017-09-11 20:33:45.538529 [DEBUG] mod_verto.c:1804 10.1.1.3:63246 Dropping Connection >>>>> 2017-09-11 20:33:45.538529 [INFO] mod_verto.c:1910 10.1.1.3:63246 Ending client thread. >>>>> 2017-09-11 20:33:45.538529 [INFO] mod_verto.c:1917 10.1.1.3:63246 Thread ended >>>>> 2017-09-11 20:33:45.538529 [ALERT] mod_verto.c:1335 READ 10.1.1.3:63258 [{ >>>>> "jsonrpc": "2.0", >>>>> "method": "login", >>>>> "params": { >>>>> "login": “201 at vertobox", >>>>> "passwd": "j4guar", >>>>> "loginParams": { >>>>> }, >>>>> "userVariables": { >>>>> }, >>>>> "sessid": "3462fb9e-5849-9560-a451-1adec02d958d" >>>>> }, >>>>> "id": 3 >>>>> }] >>>>> 2017-09-11 20:33:45.538529 [ALERT] mod_verto.c:604 WRITE 10.1.1.3:63258 [{ >>>>> "jsonrpc": "2.0", >>>>> "id": 3, >>>>> "result": { >>>>> "message": "logged in", >>>>> "sessid": "3462fb9e-5849-9560-a451-1adec02d958d" >>>>> } >>>>> }] >>>>> >>>>> >>>>>> On 7 Sep 2017, at 21:11, Ítalo Rossi > wrote: >>>>>> >>>>>> Verto communicator is a example implementation on what can be done with Verto. >>>>>> >>>>>> Check our tutorial to understand how to build a minimum app: >>>>>> >>>>>> https://evoluxbr.github.io/verto-docs/ >>>>>> Em qui, 7 de set de 2017 às 17:02, Rick Jarvis > escreveu: >>>>>> Looking to get into Verto, in particular handling voice calls with JS. Going through the source, I’m wondering: >>>>>> >>>>>> 1. What’s the difference between the Verto source and the Verto Communicator source? >>>>>> >>>>>> 2. What’s the best way to start from the bottom up - by this I mean that it seems hugely comprehensive, but rather than just use grunt to set it all up, I’d like to start simply with the basics… is there for instance a list of the bare minimum scripts / file structure to use? Apologies if this is a silly question, I’m still relatively new to JS and I don’t want to blow my mind in one go ;) >>>>>> >>>>>> Rick >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From rick at magicmail.mooo.com Mon Sep 11 23:22:49 2017 From: rick at magicmail.mooo.com (Rick Jarvis) Date: Tue, 12 Sep 2017 00:22:49 +0100 Subject: [Freeswitch-users] Verto In-Reply-To: References: <03E75515-780B-451B-A30F-62C99020D8A7@magicmail.mooo.com> <1334A7C4-01C0-47C4-9D5A-E1C3448E2080@magicmail.mooo.com> <820C03C3-999D-48F5-865E-74E7467E2CDF@jerris.com> <010379EC-6F3C-46E9-A9DA-8519F523DAA1@jerris.com> <2F114FDD-E440-409A-B756-9ECDCD0B1277@magicmail.mooo.com> Message-ID: Ok, now logged back on with the same verto code, and same debug output as below, but ‘show registrations’ gives me zero. Sorry guys, I know this is me. I’ll get it eventually. > On 12 Sep 2017, at 00:12, Rick Jarvis wrote: > > Confused now. > > ‘show registrations’ - should this give me *all*, or just the non-sofia ones? > > ATM I have 2 x ip phone registrations, and sofia is showing both, but ‘show registrations’ gives me one - despite having disconnected Chrome, and leaving the 2 ip phones connected…? > > > >> On 11 Sep 2017, at 21:29, Rick Jarvis > wrote: >> >> Ah! I had thought that all registrations ended up at sofia one way or another, thanks Mike. >> >> R >> >>> On 11 Sep 2017, at 21:15, Michael Jerris > wrote: >>> >>> verto has nothing at all to do with sofia >>> >>>> On Sep 11, 2017, at 3:59 PM, Rick Jarvis > wrote: >>>> >>>> Should it not show up in ‘sofia status profile internal reg’ ? I can see the IP phone’s reg in there, and I have multiple registrations enabled (double checked that with another IP phone!)…? >>>> >>>>> On 11 Sep 2017, at 20:52, Michael Jerris > wrote: >>>>> >>>>> That is successfully authenticating from what i see there. >>>>> >>>>>> On Sep 11, 2017, at 3:46 PM, Rick Jarvis > wrote: >>>>>> >>>>>> I have verto up and running (sort of), but I’m confused about whether authentication is working. Chrome’s console is showing ‘authentication required’ errors. Verto debug gives me the below (I have a context ‘vertobox’, which is working with a login to 201 at vertobox from an IP phone; certs / DNS etc are working ok). >>>>>> >>>>>> Is this looking good or not? I can’t see an active registration from the FS console, so it must be failing I guess? >>>>>> >>>>>> 2017-09-11 20:33:45.518527 [ALERT] mod_verto.c:1335 READ 10.1.1.3:63258 [{ >>>>>> "jsonrpc": "2.0", >>>>>> "method": "login", >>>>>> "params": { >>>>>> "sessid": "3462fb9e-5849-9560-a451-1adec02d958d" >>>>>> }, >>>>>> "id": 2 >>>>>> }] >>>>>> 2017-09-11 20:33:45.518527 [INFO] mod_verto.c:1232 10.1.1.3:63258 re-connecting session 3462fb9e-5849-9560-a451-1adec02d958d >>>>>> 2017-09-11 20:33:45.518527 [WARNING] mod_verto.c:1095 New connection for session 3462fb9e-5849-9560-a451-1adec02d958d dropping previous connection. >>>>>> 2017-09-11 20:33:45.518527 [ALERT] mod_verto.c:604 WRITE 10.1.1.3:63246 [{ >>>>>> "jsonrpc": "2.0", >>>>>> "id": 5, >>>>>> "method": "verto.punt", >>>>>> "params": { >>>>>> } >>>>>> }] >>>>>> 2017-09-11 20:33:45.518527 [ALERT] mod_verto.c:604 WRITE 10.1.1.3:63258 [{ >>>>>> "jsonrpc": "2.0", >>>>>> "id": 2, >>>>>> "error": { >>>>>> "code": -32000, >>>>>> "message": "Authentication Required" >>>>>> } >>>>>> }] >>>>>> 2017-09-11 20:33:45.538529 [DEBUG] mod_verto.c:1804 10.1.1.3:63246 Dropping Connection >>>>>> 2017-09-11 20:33:45.538529 [INFO] mod_verto.c:1910 10.1.1.3:63246 Ending client thread. >>>>>> 2017-09-11 20:33:45.538529 [INFO] mod_verto.c:1917 10.1.1.3:63246 Thread ended >>>>>> 2017-09-11 20:33:45.538529 [ALERT] mod_verto.c:1335 READ 10.1.1.3:63258 [{ >>>>>> "jsonrpc": "2.0", >>>>>> "method": "login", >>>>>> "params": { >>>>>> "login": “201 at vertobox", >>>>>> "passwd": "j4guar", >>>>>> "loginParams": { >>>>>> }, >>>>>> "userVariables": { >>>>>> }, >>>>>> "sessid": "3462fb9e-5849-9560-a451-1adec02d958d" >>>>>> }, >>>>>> "id": 3 >>>>>> }] >>>>>> 2017-09-11 20:33:45.538529 [ALERT] mod_verto.c:604 WRITE 10.1.1.3:63258 [{ >>>>>> "jsonrpc": "2.0", >>>>>> "id": 3, >>>>>> "result": { >>>>>> "message": "logged in", >>>>>> "sessid": "3462fb9e-5849-9560-a451-1adec02d958d" >>>>>> } >>>>>> }] >>>>>> >>>>>> >>>>>>> On 7 Sep 2017, at 21:11, Ítalo Rossi > wrote: >>>>>>> >>>>>>> Verto communicator is a example implementation on what can be done with Verto. >>>>>>> >>>>>>> Check our tutorial to understand how to build a minimum app: >>>>>>> >>>>>>> https://evoluxbr.github.io/verto-docs/ >>>>>>> Em qui, 7 de set de 2017 às 17:02, Rick Jarvis > escreveu: >>>>>>> Looking to get into Verto, in particular handling voice calls with JS. Going through the source, I’m wondering: >>>>>>> >>>>>>> 1. What’s the difference between the Verto source and the Verto Communicator source? >>>>>>> >>>>>>> 2. What’s the best way to start from the bottom up - by this I mean that it seems hugely comprehensive, but rather than just use grunt to set it all up, I’d like to start simply with the basics… is there for instance a list of the bare minimum scripts / file structure to use? Apologies if this is a silly question, I’m still relatively new to JS and I don’t want to blow my mind in one go ;) >>>>>>> >>>>>>> Rick >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Tue Sep 12 00:19:00 2017 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Tue, 12 Sep 2017 09:19:00 +0900 Subject: [Freeswitch-users] Verto In-Reply-To: References: <03E75515-780B-451B-A30F-62C99020D8A7@magicmail.mooo.com> <1334A7C4-01C0-47C4-9D5A-E1C3448E2080@magicmail.mooo.com> <820C03C3-999D-48F5-865E-74E7467E2CDF@jerris.com> <010379EC-6F3C-46E9-A9DA-8519F523DAA1@jerris.com> <2F114FDD-E440-409A-B756-9ECDCD0B1277@magicmail.mooo.com> Message-ID: >From fs_cli: verto status sent from mobile cell: +39 347 266 56 18 Giovanni Maruzzelli OpenTelecom.IT On Sep 12, 2017 01:23, "Rick Jarvis" wrote: > Ok, now logged back on with the same verto code, and same debug output as > below, but ‘show registrations’ gives me zero. > > Sorry guys, I know this is me. I’ll get it eventually. > > On 12 Sep 2017, at 00:12, Rick Jarvis wrote: > > Confused now. > > ‘show registrations’ - should this give me *all*, or just the non-sofia > ones? > > ATM I have 2 x ip phone registrations, and sofia is showing both, but > ‘show registrations’ gives me one - despite having disconnected Chrome, and > leaving the 2 ip phones connected…? > > > > On 11 Sep 2017, at 21:29, Rick Jarvis wrote: > > Ah! I had thought that all registrations ended up at sofia one way or > another, thanks Mike. > > R > > On 11 Sep 2017, at 21:15, Michael Jerris wrote: > > verto has nothing at all to do with sofia > > On Sep 11, 2017, at 3:59 PM, Rick Jarvis wrote: > > Should it not show up in ‘sofia status profile internal reg’ ? I can see > the IP phone’s reg in there, and I have multiple registrations enabled > (double checked that with another IP phone!)…? > > On 11 Sep 2017, at 20:52, Michael Jerris wrote: > > That is successfully authenticating from what i see there. > > On Sep 11, 2017, at 3:46 PM, Rick Jarvis wrote: > > I have verto up and running (sort of), but I’m confused about whether > authentication is working. Chrome’s console is showing ‘authentication > required’ errors. Verto debug gives me the below (I have a context > ‘vertobox’, which is working with a login to 201 at vertobox from an IP > phone; certs / DNS etc are working ok). > > Is this looking good or not? I can’t see an active registration from the > FS console, so it must be failing I guess? > > 2017-09-11 20:33:45.518527 [ALERT] mod_verto.c:1335 READ 10.1.1.3:63258 [{ > "jsonrpc": "2.0", > "method": "login", > "params": { > "sessid": "3462fb9e-5849-9560-a451-1adec02d958d" > }, > "id": 2 > }] > 2017-09-11 20:33:45.518527 [INFO] mod_verto.c:1232 10.1.1.3:63258 > re-connecting session 3462fb9e-5849-9560-a451-1adec02d958d > 2017-09-11 20:33:45.518527 [WARNING] mod_verto.c:1095 New connection for > session 3462fb9e-5849-9560-a451-1adec02d958d dropping previous connection. > 2017-09-11 20:33:45.518527 [ALERT] mod_verto.c:604 WRITE 10.1.1.3:63246 [{ > "jsonrpc": "2.0", > "id": 5, > "method": "verto.punt", > "params": { > } > }] > 2017-09-11 20:33:45.518527 [ALERT] mod_verto.c:604 WRITE 10.1.1.3:63258 [{ > "jsonrpc": "2.0", > "id": 2, > "error": { > "code": -32000, > "message": "Authentication Required" > } > }] > 2017-09-11 20:33:45.538529 [DEBUG] mod_verto.c:1804 10.1.1.3:63246 > Dropping Connection > 2017-09-11 20:33:45.538529 [INFO] mod_verto.c:1910 10.1.1.3:63246 Ending > client thread. > 2017-09-11 20:33:45.538529 [INFO] mod_verto.c:1917 10.1.1.3:63246 Thread > ended > 2017-09-11 20:33:45.538529 [ALERT] mod_verto.c:1335 READ 10.1.1.3:63258 [{ > "jsonrpc": "2.0", > "method": "login", > "params": { > "login": “201 at vertobox", > "passwd": "j4guar", > "loginParams": { > }, > "userVariables": { > }, > "sessid": "3462fb9e-5849-9560-a451-1adec02d958d" > }, > "id": 3 > }] > 2017-09-11 20:33:45.538529 [ALERT] mod_verto.c:604 WRITE 10.1.1.3:63258 [{ > "jsonrpc": "2.0", > "id": 3, > "result": { > "message": "logged in", > "sessid": "3462fb9e-5849-9560-a451-1adec02d958d" > } > }] > > > On 7 Sep 2017, at 21:11, Ítalo Rossi wrote: > > Verto communicator is a example implementation on what can be done with > Verto. > > Check our tutorial to understand how to build a minimum app: > > https://evoluxbr.github.io/verto-docs/ > Em qui, 7 de set de 2017 às 17:02, Rick Jarvis > escreveu: > >> Looking to get into Verto, in particular handling voice calls with JS. >> Going through the source, I’m wondering: >> >> 1. What’s the difference between the Verto source and the Verto >> Communicator source? >> >> 2. What’s the best way to start from the bottom up - by this I mean that >> it seems hugely comprehensive, but rather than just use grunt to set it all >> up, I’d like to start simply with the basics… is there for instance a list >> of the bare minimum scripts / file structure to use? Apologies if this is a >> silly question, I’m still relatively new to JS and I don’t want to blow my >> mind in one go ;) >> >> Rick >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From alexandr.popov at iqoption.com Tue Sep 12 08:35:50 2017 From: alexandr.popov at iqoption.com (Alexandr Popov) Date: Tue, 12 Sep 2017 11:35:50 +0300 Subject: [Freeswitch-users] Verto In-Reply-To: <010379EC-6F3C-46E9-A9DA-8519F523DAA1@jerris.com> References: <03E75515-780B-451B-A30F-62C99020D8A7@magicmail.mooo.com> <1334A7C4-01C0-47C4-9D5A-E1C3448E2080@magicmail.mooo.com> <820C03C3-999D-48F5-865E-74E7467E2CDF@jerris.com> <010379EC-6F3C-46E9-A9DA-8519F523DAA1@jerris.com> Message-ID: not correct -- verto using domain aliases made by sofia from directory. 2017-09-11 23:15 GMT+03:00 Michael Jerris : > verto has nothing at all to do with sofia > > On Sep 11, 2017, at 3:59 PM, Rick Jarvis wrote: > > Should it not show up in ‘sofia status profile internal reg’ ? I can see > the IP phone’s reg in there, and I have multiple registrations enabled > (double checked that with another IP phone!)…? > > On 11 Sep 2017, at 20:52, Michael Jerris wrote: > > That is successfully authenticating from what i see there. > > On Sep 11, 2017, at 3:46 PM, Rick Jarvis wrote: > > I have verto up and running (sort of), but I’m confused about whether > authentication is working. Chrome’s console is showing ‘authentication > required’ errors. Verto debug gives me the below (I have a context > ‘vertobox’, which is working with a login to 201 at vertobox from an IP > phone; certs / DNS etc are working ok). > > Is this looking good or not? I can’t see an active registration from the > FS console, so it must be failing I guess? > > 2017-09-11 20:33:45.518527 [ALERT] mod_verto.c:1335 READ 10.1.1.3:63258 [{ > "jsonrpc": "2.0", > "method": "login", > "params": { > "sessid": "3462fb9e-5849-9560-a451-1adec02d958d" > }, > "id": 2 > }] > 2017-09-11 20:33:45.518527 [INFO] mod_verto.c:1232 10.1.1.3:63258 > re-connecting session 3462fb9e-5849-9560-a451-1adec02d958d > 2017-09-11 20:33:45.518527 [WARNING] mod_verto.c:1095 New connection for > session 3462fb9e-5849-9560-a451-1adec02d958d dropping previous connection. > 2017-09-11 20:33:45.518527 [ALERT] mod_verto.c:604 WRITE 10.1.1.3:63246 [{ > "jsonrpc": "2.0", > "id": 5, > "method": "verto.punt", > "params": { > } > }] > 2017-09-11 20:33:45.518527 [ALERT] mod_verto.c:604 WRITE 10.1.1.3:63258 [{ > "jsonrpc": "2.0", > "id": 2, > "error": { > "code": -32000, > "message": "Authentication Required" > } > }] > 2017-09-11 20:33:45.538529 [DEBUG] mod_verto.c:1804 10.1.1.3:63246 > Dropping Connection > 2017-09-11 20:33:45.538529 [INFO] mod_verto.c:1910 10.1.1.3:63246 Ending > client thread. > 2017-09-11 20:33:45.538529 [INFO] mod_verto.c:1917 10.1.1.3:63246 Thread > ended > 2017-09-11 20:33:45.538529 [ALERT] mod_verto.c:1335 READ 10.1.1.3:63258 [{ > "jsonrpc": "2.0", > "method": "login", > "params": { > "login": “201 at vertobox", > "passwd": "j4guar", > "loginParams": { > }, > "userVariables": { > }, > "sessid": "3462fb9e-5849-9560-a451-1adec02d958d" > }, > "id": 3 > }] > 2017-09-11 20:33:45.538529 [ALERT] mod_verto.c:604 WRITE 10.1.1.3:63258 [{ > "jsonrpc": "2.0", > "id": 3, > "result": { > "message": "logged in", > "sessid": "3462fb9e-5849-9560-a451-1adec02d958d" > } > }] > > > On 7 Sep 2017, at 21:11, Ítalo Rossi wrote: > > Verto communicator is a example implementation on what can be done with > Verto. > > Check our tutorial to understand how to build a minimum app: > > https://evoluxbr.github.io/verto-docs/ > Em qui, 7 de set de 2017 às 17:02, Rick Jarvis > escreveu: > >> Looking to get into Verto, in particular handling voice calls with JS. >> Going through the source, I’m wondering: >> >> 1. What’s the difference between the Verto source and the Verto >> Communicator source? >> >> 2. What’s the best way to start from the bottom up - by this I mean that >> it seems hugely comprehensive, but rather than just use grunt to set it all >> up, I’d like to start simply with the basics… is there for instance a list >> of the bare minimum scripts / file structure to use? Apologies if this is a >> silly question, I’m still relatively new to JS and I don’t want to blow my >> mind in one go ;) >> >> Rick >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From julf at julf.com Tue Sep 12 13:37:21 2017 From: julf at julf.com (Johan Helsingius) Date: Tue, 12 Sep 2017 15:37:21 +0200 Subject: [Freeswitch-users] FreeSWITCH on Raspberry Pi In-Reply-To: References: Message-ID: Hi, Seems the last mention of the Raspberry Pi on this list is from 2014. Are there any up-to-date / working installation instructions for the RPi (raspbian)? Julf From wsimon at stratusvideo.com Tue Sep 12 14:24:55 2017 From: wsimon at stratusvideo.com (William Simon) Date: Tue, 12 Sep 2017 14:24:55 +0000 Subject: [Freeswitch-users] 2 person video conference layout In-Reply-To: References: Message-ID: <5CA6AE5A-360F-4D9C-BBE7-FBF8AF217F28@stratusvideo.com> You can use the conference option "video bridge first two". You can also do this by enabling personal canvas but at resource penalty (the FS team also considers it experimental to some degree). > On Sep 11, 2017, at 7:28 AM, Peter Baines wrote: > > Hello, > > I am trying to create a conference layout for a two person conference only where they only see each other. > > i.e. > > Person A sees Person B > Person B sees Person A > > Is this possible with the conference layout system? > > Regards, > Peter > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/pkcs7-signature Size: 3872 bytes Desc: not available URL: From bipin at xbipin.com Tue Sep 12 14:29:37 2017 From: bipin at xbipin.com (Bipin Patel) Date: Tue, 12 Sep 2017 18:29:37 +0400 Subject: [Freeswitch-users] FreeSWITCH on Raspberry Pi In-Reply-To: References: Message-ID: hi, yes it works fine on raspbian jessie and we use it in production Regards, Bipin ------------------------------------------------------------------------ -------- Original Message -------- Subject: [Freeswitch-users] FreeSWITCH on Raspberry Pi From: Johan Helsingius To: freeswitch-users at lists.freeswitch.org Date: 9/12/2017, 5:37:21 PM > Hi, > > Seems the last mention of the Raspberry Pi on this list is from 2014. > Are there any up-to-date / working installation instructions for the > RPi (raspbian)? > > Julf > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From rick at magicmail.mooo.com Tue Sep 12 14:36:00 2017 From: rick at magicmail.mooo.com (Rick Jarvis) Date: Tue, 12 Sep 2017 15:36:00 +0100 Subject: [Freeswitch-users] Fwd: Verto References: <637257B6-7F09-41B1-B5F5-621D320619F4@magicmail.mooo.com> Message-ID: <54BDD1AD-AD50-4028-BFF6-1D24D2AD9D32@magicmail.mooo.com> Resending with link to image (sorry I know images aren’t ideal for mailing lists, but the format goes if I copy&paste)... From: Rick Jarvis > Subject: Re: [Freeswitch-users] Verto Date: 12 September 2017 at 13:53:10 BST To: FreeSWITCH Users Help > Definitely making progress... Can anyone tell me what’s going on here, this is when I try and make a call (it’s an internal call, wondered if it’s hitting the wrong context or something, but the only logging I’m getting is from mod_verto, so I don’t think it’s getting that far): Link to Chrome console output: https://pasteboard.co/GK2exjV.png Verto console output: "jsonrpc": "2.0", "id": 5, "error": { "code": -32601, "message": "Invalid Method, Missing Method or Permission Denied" } -------------- next part -------------- An HTML attachment was scrubbed... URL: From mjlopez at smartic.es Tue Sep 12 14:44:05 2017 From: mjlopez at smartic.es (=?iso-8859-1?Q?Miguel_Jes=FAs_L=F3pez_Valverde?=) Date: Tue, 12 Sep 2017 16:44:05 +0200 Subject: [Freeswitch-users] WARNING message with sipjs clients and Freeswitch. Message-ID: <00fc01d32bd5$95e6b6b0$c1b42410$@smartic.es> Hello newly: I send this email in case anyone knows a solution. Using FreeSwitch together with a configuration of sipjs through wws as client, I get a correct operation except for the usual message [WARNING] switch_core_media.c:3447 NO candidate ACL defined, Defaulting to wan.auto This only happens to me using sipjs as sip client, using the same accounts through other clients, like Zoiper, this message does not appear. I have tried different directives in the file autoload_configs/acl.conf.xml but none has caused the disappearance of this message. Do you know that I have to change in the FS configuration so that this message disappears? Thank you very much. Miguel J. López Valverde Dpto. Técnico. _____ Pº de la Castellana, 135 7ª pl 28046 Madrid España Tel: 900 900 368 | Móvil: (+34) 667 772 911 E-mail: mjlopez at smartic.es | Web: http://www.smartic.es _____ Sus datos de carácter personal (nombre, apellidos, dirección postal y de correo electrónico, etc.) son tratados para la gestión de su relación con la Entidad, así como para el envío de información sobre nuestra actividad y la de terceros relacionadas con la actividad de Consulting Smartic Solutions, S.L., CIF: B85130037, Pº de la Castellana, 135 7ª pl 28046 Madrid. Usted puede ejercer sus derechos de acceso, rectificación, cancelación y oposición dirigiéndose por escrito, con copia de un documento que acredite su identidad, a la dirección info (arroba) smartic.es. Este mensaje puede contener información confidencial. Si usted no es su destinatario, no debe leerlo, copiarlo, distribuirlo, ni hacer uso de la información que contiene. En este caso, por favor, llámenos o comuníquenoslo por escrito y borre este mensaje de su sistema --- El software de antivirus Avast ha analizado este correo electrónico en busca de virus. https://www.avast.com/antivirus -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.png Type: image/png Size: 4475 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image002.jpg Type: image/jpeg Size: 1217 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image003.jpg Type: image/jpeg Size: 1201 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image004.jpg Type: image/jpeg Size: 1184 bytes Desc: not available URL: From julf at julf.com Tue Sep 12 14:58:39 2017 From: julf at julf.com (Johan Helsingius) Date: Tue, 12 Sep 2017 16:58:39 +0200 Subject: [Freeswitch-users] FreeSWITCH on Raspberry Pi In-Reply-To: References: Message-ID: <57511fd6-6f92-a9be-ac3a-e33cd6af973c@julf.com> > yes it works fine on raspbian jessie and we use it in production Great! Built from source, or is there a repository? Julf From lists at telefaks.de Tue Sep 12 15:00:08 2017 From: lists at telefaks.de (Peter Steinbach) Date: Tue, 12 Sep 2017 17:00:08 +0200 Subject: [Freeswitch-users] FreeSWITCH on Raspberry Pi In-Reply-To: References: Message-ID: <59B7F678.1050403@telefaks.de> I also have some Raspberries running with Freeswitch. Works nicely. Except: Raspberry is killing SD cards. I had to change mine every 6-12 months. Finally I switched to Alix boards which OS runs the CF card in RO mode. Best regards Peter On 09/12/17 16:29, Bipin Patel wrote: > hi, > > yes it works fine on raspbian jessie and we use it in production > > > Regards, > Bipin > > > > ------------------------------------------------------------------------ > > -------- Original Message -------- > Subject: [Freeswitch-users] FreeSWITCH on Raspberry Pi > From: Johan Helsingius > To: freeswitch-users at lists.freeswitch.org > Date: 9/12/2017, 5:37:21 PM > >> Hi, >> >> Seems the last mention of the Raspberry Pi on this list is from 2014. >> Are there any up-to-date / working installation instructions for the >> RPi (raspbian)? >> >> Julf >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de From bipin at xbipin.com Tue Sep 12 15:22:19 2017 From: bipin at xbipin.com (Bipin Patel) Date: Tue, 12 Sep 2017 19:22:19 +0400 Subject: [Freeswitch-users] FreeSWITCH on Raspberry Pi In-Reply-To: <59B7F678.1050403@telefaks.de> References: <59B7F678.1050403@telefaks.de> Message-ID: <1f5eeb57-81f4-b841-bdeb-eae4f707c7da@xbipin.com> hi, i have been using since long on a a rpi and never killed my sd card unless ur logging or writing a lot to it i have many alix and apu2 boards but we use that for pfsense rather than freswitch Regards, Bipin ------------------------------------------------------------------------ -------- Original Message -------- Subject: Re: [Freeswitch-users] FreeSWITCH on Raspberry Pi From: Peter Steinbach To: FreeSWITCH Users Help Date: 9/12/2017, 7:00:08 PM > I also have some Raspberries running with Freeswitch. Works nicely. > Except: Raspberry is killing SD cards. I had to change mine every 6-12 > months. > > Finally I switched to Alix boards which OS runs the CF card in RO mode. > > Best regards > Peter > > On 09/12/17 16:29, Bipin Patel wrote: >> hi, >> >> yes it works fine on raspbian jessie and we use it in production >> >> >> Regards, >> Bipin >> >> >> >> ------------------------------------------------------------------------ >> >> -------- Original Message -------- >> Subject: [Freeswitch-users] FreeSWITCH on Raspberry Pi >> From: Johan Helsingius >> To: freeswitch-users at lists.freeswitch.org >> Date: 9/12/2017, 5:37:21 PM >> >>> Hi, >>> >>> Seems the last mention of the Raspberry Pi on this list is from 2014. >>> Are there any up-to-date / working installation instructions for the >>> RPi (raspbian)? >>> >>> Julf >>> >>> _________________________________________________________________________ >>> >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > From sebastian_ml at gmx.net Tue Sep 12 15:57:24 2017 From: sebastian_ml at gmx.net (Sebastian Kemper) Date: Tue, 12 Sep 2017 15:57:24 +0000 Subject: [Freeswitch-users] FreeSWITCH on Raspberry Pi In-Reply-To: <59B7F678.1050403@telefaks.de> References: <59B7F678.1050403@telefaks.de> Message-ID: On OpenWRT/LEDE the default config points the log and db dir to a folder in tmpfs. That way FS can write as often as it wants without degrading the flash drive. Regards, Sebastian Am 12. September 2017 17:00:08 MESZ schrieb Peter Steinbach : >I also have some Raspberries running with Freeswitch. Works nicely. >Except: Raspberry is killing SD cards. I had to change mine every 6-12 >months. > >Finally I switched to Alix boards which OS runs the CF card in RO mode. > >Best regards >Peter > >On 09/12/17 16:29, Bipin Patel wrote: >> hi, >> >> yes it works fine on raspbian jessie and we use it in production >> >> >> Regards, >> Bipin >> >> >> >> >------------------------------------------------------------------------ >> >> -------- Original Message -------- >> Subject: [Freeswitch-users] FreeSWITCH on Raspberry Pi >> From: Johan Helsingius >> To: freeswitch-users at lists.freeswitch.org >> Date: 9/12/2017, 5:37:21 PM >> >>> Hi, >>> >>> Seems the last mention of the Raspberry Pi on this list is from >2014. >>> Are there any up-to-date / working installation instructions for the >>> RPi (raspbian)? >>> >>> Julf >>> >>> >_________________________________________________________________________ >>> >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >_________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > >-- >With kind regards >Peter Steinbach > >Telefaks Services GmbH >mailto:lists (att) telefaks.de >Internet: www.telefaks.de > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://confluence.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org From luca.pradovera at gmail.com Tue Sep 12 17:10:34 2017 From: luca.pradovera at gmail.com (Luca Pradovera) Date: Tue, 12 Sep 2017 19:10:34 +0200 Subject: [Freeswitch-users] Media bugs and video Message-ID: Hello, I know this is probably not possible or desirable, but could anyone suggest if there is a way (or even just some internal provisions I can leverage) to make one user part of multiple video conferences? I suppose some kind of video bug could work, since it is used by mod_cv and others. Thank you! Luca -------------- next part -------------- An HTML attachment was scrubbed... URL: From jungleboogie0 at gmail.com Tue Sep 12 17:39:57 2017 From: jungleboogie0 at gmail.com (jungle Boogie) Date: Tue, 12 Sep 2017 10:39:57 -0700 Subject: [Freeswitch-users] FreeSWITCH on Raspberry Pi In-Reply-To: References: Message-ID: On 12 September 2017 at 06:37, Johan Helsingius wrote: > Hi, > > Seems the last mention of the Raspberry Pi on this list is from 2014. > Are there any up-to-date / working installation instructions for the > RPi (raspbian)? > I build freeswitch from git master about once a week (although a recent move put that on hold ATM). The initial build and installing of dependencies might take a good three hours, but after that, building from src takes about 100 minutes. I'm not doing anything with verto or video, though. > Julf > > From gmaruzz at gmail.com Tue Sep 12 19:02:46 2017 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Wed, 13 Sep 2017 04:02:46 +0900 Subject: [Freeswitch-users] latest chrome 61 android does not works with mux conf Message-ID: https://freeswitch.org/jira/browse/FS-10664 chrome 61 android does not works with mux conf you can check it out on cantina/vc I found that I can make my client (not vc) to work if I set video_mirror_input=true in dialplan, and DO NOT SET minimize-video-encoding in conference flag -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From ssinyagin at gmail.com Tue Sep 12 19:50:51 2017 From: ssinyagin at gmail.com (Stanislav Sinyagin) Date: Tue, 12 Sep 2017 21:50:51 +0200 Subject: [Freeswitch-users] FreeSWITCH on Raspberry Pi In-Reply-To: <59B7F678.1050403@telefaks.de> References: <59B7F678.1050403@telefaks.de> Message-ID: Alix is too old. The company has a much better board already: http://pcengines.ch/apu2.htm It can accommodate a real SSD with a much longer life cycle. Also I made a debian installer for it: https://github.com/ssinyagin/pcengines-apu-debian-cd in the ARM world, here's a box that houses a 2.5" disk: http://www.friendlyarm.com/index.php?route=product/product&product_id=192 Also this board has onboard 8GB eMMC, which is expected to be more reliable than an SD card: http://www.friendlyarm.com/index.php?route=product/product&path=69&product_id=196 On Tue, Sep 12, 2017 at 5:00 PM, Peter Steinbach wrote: > I also have some Raspberries running with Freeswitch. Works nicely. > Except: Raspberry is killing SD cards. I had to change mine every 6-12 > months. > > Finally I switched to Alix boards which OS runs the CF card in RO mode. > > Best regards > Peter > > On 09/12/17 16:29, Bipin Patel wrote: >> hi, >> >> yes it works fine on raspbian jessie and we use it in production >> >> >> Regards, >> Bipin >> >> >> >> ------------------------------------------------------------------------ >> >> -------- Original Message -------- >> Subject: [Freeswitch-users] FreeSWITCH on Raspberry Pi >> From: Johan Helsingius >> To: freeswitch-users at lists.freeswitch.org >> Date: 9/12/2017, 5:37:21 PM >> >>> Hi, >>> >>> Seems the last mention of the Raspberry Pi on this list is from 2014. >>> Are there any up-to-date / working installation instructions for the >>> RPi (raspbian)? >>> >>> Julf >>> >>> _________________________________________________________________________ >>> >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > -- > With kind regards > Peter Steinbach > > Telefaks Services GmbH > mailto:lists (att) telefaks.de > Internet: www.telefaks.de > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jungleboogie0 at gmail.com Tue Sep 12 20:12:11 2017 From: jungleboogie0 at gmail.com (jungle Boogie) Date: Tue, 12 Sep 2017 13:12:11 -0700 Subject: [Freeswitch-users] FreeSWITCH on Raspberry Pi In-Reply-To: References: <59B7F678.1050403@telefaks.de> Message-ID: On 12 September 2017 at 12:50, Stanislav Sinyagin wrote: > Alix is too old. The company has a much better board already: > http://pcengines.ch/apu2.htm > It can accommodate a real SSD with a much longer life cycle. > Also I made a debian installer for it: > https://github.com/ssinyagin/pcengines-apu-debian-cd > > in the ARM world, here's a box that houses a 2.5" disk: > http://www.friendlyarm.com/index.php?route=product/product&product_id=192 > This looks neat. Do you think it would be faster than an SD card on a raspberry pi3? From xlin at soleocommunications.com Tue Sep 12 21:29:45 2017 From: xlin at soleocommunications.com (Lin, Xiaoping) Date: Tue, 12 Sep 2017 21:29:45 +0000 Subject: [Freeswitch-users] Httapi record Message-ID: Hello, I try to make this simple program to work. It did not record anything. Please help. Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: From ssinyagin at gmail.com Tue Sep 12 22:28:19 2017 From: ssinyagin at gmail.com (Stanislav Sinyagin) Date: Wed, 13 Sep 2017 00:28:19 +0200 Subject: [Freeswitch-users] FreeSWITCH on Raspberry Pi In-Reply-To: References: <59B7F678.1050403@telefaks.de> Message-ID: On 12 Sep 2017 22:13, "jungle Boogie" wrote: On 12 September 2017 at 12:50, Stanislav Sinyagin wrote: > Alix is too old. The company has a much better board already: > http://pcengines.ch/apu2.htm > It can accommodate a real SSD with a much longer life cycle. > Also I made a debian installer for it: > https://github.com/ssinyagin/pcengines-apu-debian-cd > > in the ARM world, here's a box that houses a 2.5" disk: > http://www.friendlyarm.com/index.php?route=product/product&product_id=192 > This looks neat. Do you think it would be faster than an SD card on a raspberry pi3? I guess so, although it's SATA over USB 2.0 -------------- next part -------------- An HTML attachment was scrubbed... URL: From rick at magicmail.mooo.com Tue Sep 12 22:36:48 2017 From: rick at magicmail.mooo.com (Rick Jarvis) Date: Tue, 12 Sep 2017 23:36:48 +0100 Subject: [Freeswitch-users] Verto In-Reply-To: <54BDD1AD-AD50-4028-BFF6-1D24D2AD9D32@magicmail.mooo.com> References: <637257B6-7F09-41B1-B5F5-621D320619F4@magicmail.mooo.com> <54BDD1AD-AD50-4028-BFF6-1D24D2AD9D32@magicmail.mooo.com> Message-ID: <1CF14F27-6B9C-49E7-9F04-1AB9B86E8CFC@magicmail.mooo.com> Anyone got any ideas? Desperate to get this working! :) That console log: https://pasteboard.co/GK2exjV.png > On 12 Sep 2017, at 15:36, Rick Jarvis wrote: > > Resending with link to image (sorry I know images aren’t ideal for mailing lists, but the format goes if I copy&paste)... > > From: Rick Jarvis > > Subject: Re: [Freeswitch-users] Verto > Date: 12 September 2017 at 13:53:10 BST > To: FreeSWITCH Users Help > > > Definitely making progress... > > Can anyone tell me what’s going on here, this is when I try and make a call (it’s an internal call, wondered if it’s hitting the wrong context or something, but the only logging I’m getting is from mod_verto, so I don’t think it’s getting that far): > > Link to Chrome console output: > https://pasteboard.co/GK2exjV.png > > Verto console output: > "jsonrpc": "2.0", > "id": 5, > "error": { > "code": -32601, > "message": "Invalid Method, Missing Method or Permission Denied" > } > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From s.safarov at gmail.com Wed Sep 13 03:10:04 2017 From: s.safarov at gmail.com (Sergey Safarov) Date: Wed, 13 Sep 2017 03:10:04 +0000 Subject: [Freeswitch-users] log libsofia Message-ID: Is possible write to log file info produced by by libsofia tport? I can see this info using "sofia loglevel tport 9", but how to write this info to file? -------------- next part -------------- An HTML attachment was scrubbed... URL: From s.safarov at gmail.com Wed Sep 13 03:24:54 2017 From: s.safarov at gmail.com (Sergey Safarov) Date: Wed, 13 Sep 2017 03:24:54 +0000 Subject: [Freeswitch-users] how to get event of failed call Message-ID: FS originates call to carrier. FS receives "200 OK" from carrier and send ACK but for some reasons ACK not received by carrier. Then carrier retransmits "200 OK" messages and all retransmits also will failed. Carrier send BYE message after timeout as call not established correctly. Is possible to get event of failed call on FS side. As example - we got several "200 OK" messages and "BYE" message is received after 10 sec of last "20 OK"? -------------- next part -------------- An HTML attachment was scrubbed... URL: From findmeinwland at gmail.com Wed Sep 13 07:34:39 2017 From: findmeinwland at gmail.com (Artur Mega) Date: Wed, 13 Sep 2017 12:34:39 +0500 Subject: [Freeswitch-users] how to get event of failed call In-Reply-To: References: Message-ID: I think here just need to watch for variables (like hangup cause) on CHANNEL_DESTROY event. Did you tried tis? 2017-09-13 8:24 GMT+05:00 Sergey Safarov : > FS originates call to carrier. FS receives "200 OK" from carrier and send > ACK but for some reasons ACK not received by carrier. > Then carrier retransmits "200 OK" messages and all retransmits also will > failed. > Carrier send BYE message after timeout as call not established correctly. > > Is possible to get event of failed call on FS side. As example - we got > several "200 OK" messages and "BYE" message is received after 10 sec of > last "20 OK"? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- ​С уважением, ​ Артур ​Regards, Arthur​ -------------- next part -------------- An HTML attachment was scrubbed... URL: From asilva at wirelessmundi.com Wed Sep 13 07:45:02 2017 From: asilva at wirelessmundi.com (=?UTF-8?Q?Ant=c3=b3nio_Silva?=) Date: Wed, 13 Sep 2017 09:45:02 +0200 Subject: [Freeswitch-users] Verto In-Reply-To: <1CF14F27-6B9C-49E7-9F04-1AB9B86E8CFC@magicmail.mooo.com> References: <637257B6-7F09-41B1-B5F5-621D320619F4@magicmail.mooo.com> <54BDD1AD-AD50-4028-BFF6-1D24D2AD9D32@magicmail.mooo.com> <1CF14F27-6B9C-49E7-9F04-1AB9B86E8CFC@magicmail.mooo.com> Message-ID: <30116e26-f07c-dda8-0593-c16d84f235be@wirelessmundi.com> Hi, you don't get logs in dialplan because the call is not hitting dialplan, it looks stuck in the  client. The method that is sent is not valid or is empty. Are you using the verto demo client with vanilla config? Also did you enable verto in jsonrpc parameters for that user, params: On 09/13/2017 12:36 AM, Rick Jarvis wrote: > Anyone got any ideas? Desperate to get this working! :) That console > log: https://pasteboard.co/GK2exjV.png > > >> On 12 Sep 2017, at 15:36, Rick Jarvis > > wrote: >> >>  Resending with link to image (sorry I know images aren’t ideal for >> mailing lists, but the format goes if I copy&paste)... >> >> *From: *Rick Jarvis > > >> *Subject: **Re: [Freeswitch-users] Verto* >> *Date: *12 September 2017 at 13:53:10 BST >> *To: *FreeSWITCH Users Help > > >> >> Definitely making progress... >> >> Can anyone tell me what’s going on here, this is when I try and make >> a call (it’s an internal call, wondered if it’s hitting the wrong >> context or something, but the only logging I’m getting is from >> mod_verto, so I don’t think it’s getting that far): >> >> Link to Chrome console output: >> https://pasteboard.co/GK2exjV.png >> >> Verto console output: >> "jsonrpc":"2.0", >> "id":5, >> "error":{ >> "code":-32601, >> "message":"Invalid Method, Missing Method or Permission Denied" >> } >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Saludos / Regards / Cumprimentos António Silva -------------- next part -------------- An HTML attachment was scrubbed... URL: From bipin at xbipin.com Wed Sep 13 08:23:56 2017 From: bipin at xbipin.com (Bipin Patel) Date: Wed, 13 Sep 2017 12:23:56 +0400 Subject: [Freeswitch-users] FreeSWITCH on Raspberry Pi In-Reply-To: References: <59B7F678.1050403@telefaks.de> Message-ID: <7b6050ae-758b-b7c7-0856-dd45e16d8b33@xbipin.com> hi, well for me to compile FS on a rpi3 takes around 20-30mins only although the first time library install takes quiet some time. i would anyday go with a apu2 with msata ssd or an external ssd, both ways it pretty good not to mention Pascal from pcengines would kindly provide a discount if u asked for it :) Regards, Bipin ------------------------------------------------------------------------ -------- Original Message -------- Subject: Re: [Freeswitch-users] FreeSWITCH on Raspberry Pi From: jungle Boogie To: FreeSWITCH Users Help Date: 9/13/2017, 12:12:11 AM > On 12 September 2017 at 12:50, Stanislav Sinyagin wrote: >> Alix is too old. The company has a much better board already: >> http://pcengines.ch/apu2.htm >> It can accommodate a real SSD with a much longer life cycle. >> Also I made a debian installer for it: >> https://github.com/ssinyagin/pcengines-apu-debian-cd >> >> in the ARM world, here's a box that houses a 2.5" disk: >> http://www.friendlyarm.com/index.php?route=product/product&product_id=192 >> > > This looks neat. Do you think it would be faster than an SD card on a > raspberry pi3? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From tculjaga at gmail.com Wed Sep 13 08:35:18 2017 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Wed, 13 Sep 2017 10:35:18 +0200 Subject: [Freeswitch-users] FreeSWITCH on Raspberry Pi In-Reply-To: References: <59B7F678.1050403@telefaks.de> Message-ID: if you need to do a lot of writes, sdcard is not a good option :=) this is why most embedded systems use RO fs and they do write something only when needed. I tried this one: http://khadas.com/vim/ and i can say its really good. As for RPi i was able to compile FS from source without no issues. Of course you need to reconfigure your OS (i used centos ) properly. I moved FS database on a ram disk and i can say it works really nice. FS logging is disabled.. i enable it only if needed when i debug somethig T. On 13 September 2017 at 00:28, Stanislav Sinyagin wrote: > > > On 12 Sep 2017 22:13, "jungle Boogie" wrote: > > On 12 September 2017 at 12:50, Stanislav Sinyagin > wrote: > > Alix is too old. The company has a much better board already: > > http://pcengines.ch/apu2.htm > > It can accommodate a real SSD with a much longer life cycle. > > Also I made a debian installer for it: > > https://github.com/ssinyagin/pcengines-apu-debian-cd > > > > in the ARM world, here's a box that houses a 2.5" disk: > > http://www.friendlyarm.com/index.php?route=product/product& > product_id=192 > > > > This looks neat. Do you think it would be faster than an SD card on a > raspberry pi3? > > > I guess so, although it's SATA over USB 2.0 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From rick at magicmail.mooo.com Wed Sep 13 13:02:57 2017 From: rick at magicmail.mooo.com (Rick Jarvis) Date: Wed, 13 Sep 2017 14:02:57 +0100 Subject: [Freeswitch-users] Verto In-Reply-To: <30116e26-f07c-dda8-0593-c16d84f235be@wirelessmundi.com> References: <637257B6-7F09-41B1-B5F5-621D320619F4@magicmail.mooo.com> <54BDD1AD-AD50-4028-BFF6-1D24D2AD9D32@magicmail.mooo.com> <1CF14F27-6B9C-49E7-9F04-1AB9B86E8CFC@magicmail.mooo.com> <30116e26-f07c-dda8-0593-c16d84f235be@wirelessmundi.com> Message-ID: <80814D79-29AD-4820-99A5-81478D7BD0C9@magicmail.mooo.com> I was missing the verto parameter in the user file, thanks! Following Italo’s howto: https://evoluxbr.github.io/verto-docs/ with a vanilla-based config. I have calls connecting now (yay!), but no audio. Am I correct in thinking that the useMic and useSpeak parameters should take the long ID strings that are shown in the console when verto serialises all the audio devices? > On 13 Sep 2017, at 08:45, António Silva wrote: > > Hi, > > you don't get logs in dialplan because the call is not hitting dialplan, it looks stuck in the client. The method that is sent is not valid or is empty. Are you using the verto demo client with vanilla config? > > > Also did you enable verto in jsonrpc parameters for that user, params: > > > > > > > On 09/13/2017 12:36 AM, Rick Jarvis wrote: >> Anyone got any ideas? Desperate to get this working! :) That console log: https://pasteboard.co/GK2exjV.png >> >> >>> On 12 Sep 2017, at 15:36, Rick Jarvis > wrote: >>> >>> Resending with link to image (sorry I know images aren’t ideal for mailing lists, but the format goes if I copy&paste)... >>> >>> From: Rick Jarvis > >>> Subject: Re: [Freeswitch-users] Verto >>> Date: 12 September 2017 at 13:53:10 BST >>> To: FreeSWITCH Users Help > >>> >>> Definitely making progress... >>> >>> Can anyone tell me what’s going on here, this is when I try and make a call (it’s an internal call, wondered if it’s hitting the wrong context or something, but the only logging I’m getting is from mod_verto, so I don’t think it’s getting that far): >>> >>> Link to Chrome console output: >>> https://pasteboard.co/GK2exjV.png >>> >>> Verto console output: >>> "jsonrpc": "2.0", >>> "id": 5, >>> "error": { >>> "code": -32601, >>> "message": "Invalid Method, Missing Method or Permission Denied" >>> } >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > -- > Saludos / Regards / Cumprimentos > António Silva > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From xlin at soleocommunications.com Wed Sep 13 13:34:15 2017 From: xlin at soleocommunications.com (Lin, Xiaoping) Date: Wed, 13 Sep 2017 13:34:15 +0000 Subject: [Freeswitch-users] Anyone has experience of using httapi record? Message-ID: HI! Anyone has experience with using freeswitch record? I am using it in my web app. It seems not working and I did not get any error message in the log. It did not generate the recording file and I do not know if anything wrong. Thanks! * Xiaoping I am using freeswitch-application-httapi-1.6.8-1.el7.centos.x86_64 The document the webapp sends over to freeswitch looks like: On the console: 2017-09-12 17:22:01.000062 [DEBUG] switch_channel.c:3770 (sofia/external/5022089400 at 10.2.11.207) Callstate Change EARLY -> ACTIVE EXECUTE sofia/external/5022089400 at 10.2.11.207 set_audio_level(read 2) 2017-09-12 17:22:01.000062 [DEBUG] switch_core_media_bug.c:828 Attaching BUG to sofia/external/5022089400 at 10.2.11.207 EXECUTE sofia/external/5022089400 at 10.2.11.207 httapi({method=POST,url=http://10.2.11.207:8690/VoiceApp/servlet/VoiceAppServlet?Context=ASRTestApp&Debug=true}) 2017-09-12 17:22:01.140851 [CRIT] mod_httapi.c:1153 Debugging Return Data: 2017-09-12 17:22:01.140851 [DEBUG] mod_httapi.c:1222 Process Tag: [playback] 2017-09-12 17:22:01.140851 [DEBUG] sofia.c:6858 Channel sofia/external/5022089400 at 10.2.11.207 entering state [completed][200] 2017-09-12 17:22:01.140851 [DEBUG] switch_ivr_play_say.c:1467 Codec Activated L16 at 8000hz 1 channels 20ms 2017-09-12 17:22:01.168443 [DEBUG] switch_rtp.c:6707 Correct audio ip/port confirmed. 2017-09-12 17:22:01.168443 [DEBUG] switch_core_io.c:448 Setting BUG Codec PCMU:0 2017-09-12 17:22:01.220055 [DEBUG] sofia.c:6858 Channel sofia/external/5022089400 at 10.2.11.207 entering state [ready][200] 2017-09-12 17:22:04.700049 [DEBUG] switch_ivr_play_say.c:1910 done playing file http://10.2.11.207:8690//CA/servlet/CA/ASRTestApp/Default/en_US/AgainNumNotInService.wav 2017-09-12 17:22:04.700049 [DEBUG] mod_httapi.c:1222 Process Tag: [playback] 2017-09-12 17:22:04.700049 [DEBUG] switch_ivr_play_say.c:1467 Codec Activated L16 at 8000hz 1 channels 20ms 2017-09-12 17:22:07.660037 [DEBUG] switch_ivr_play_say.c:1910 done playing file http://10.2.11.207:8690//CA/servlet/CA/ASRTestApp/Default/en_US/WhatBusinessLookingOne.wav 2017-09-12 17:22:07.660037 [DEBUG] mod_httapi.c:1222 Process Tag: [execute] EXECUTE sofia/external/5022089400 at 10.2.11.207 flush_dtmf() 2017-09-12 17:22:07.660037 [DEBUG] mod_httapi.c:1222 Process Tag: [record] 2017-09-12 17:22:07.660037 [DEBUG] switch_ivr_play_say.c:673 Raw Codec Activated 2017-09-12 17:22:07.660037 [DEBUG] switch_core_codec.c:221 sofia/external/5022089400 at 10.2.11.207 Push codec L16:100 2017-09-12 17:22:13.480059 [DEBUG] switch_core_codec.c:246 sofia/external/5022089400 at 10.2.11.207 Restore previous codec PCMU:0. 2017-09-12 17:22:13.480059 [CRIT] mod_httapi.c:1153 Debugging Return Data: 2017-09-12 17:22:13.480059 [DEBUG] mod_httapi.c:1222 Process Tag: [hangup] 2017-09-12 17:22:13.480059 [NOTICE] mod_httapi.c:880 Hangup sofia/external/5022089400 at 10.2.11.207 [CS_EXECUTE] [NORMAL_CLEARING] -------------- next part -------------- An HTML attachment was scrubbed... URL: From rick at magicmail.mooo.com Wed Sep 13 13:34:38 2017 From: rick at magicmail.mooo.com (Rick Jarvis) Date: Wed, 13 Sep 2017 14:34:38 +0100 Subject: [Freeswitch-users] Verto In-Reply-To: <80814D79-29AD-4820-99A5-81478D7BD0C9@magicmail.mooo.com> References: <637257B6-7F09-41B1-B5F5-621D320619F4@magicmail.mooo.com> <54BDD1AD-AD50-4028-BFF6-1D24D2AD9D32@magicmail.mooo.com> <1CF14F27-6B9C-49E7-9F04-1AB9B86E8CFC@magicmail.mooo.com> <30116e26-f07c-dda8-0593-c16d84f235be@wirelessmundi.com> <80814D79-29AD-4820-99A5-81478D7BD0C9@magicmail.mooo.com> Message-ID: <7925711E-B6DA-4692-884A-3B3C0A54BE5F@magicmail.mooo.com> And errors in Chrome console, seem to relate: Uncaught TypeError: Cannot read property 'srcObject' of undefined at onRemoteStream (jquery.FSRTC.js:269) at Object.onRemoteStream (jquery.FSRTC.js:439) at RTCPeerConnection.peer.onaddstream (jquery.FSRTC.js:819) onRemoteStream @ jquery.FSRTC.js:269 onRemoteStream @ jquery.FSRTC.js:439 RTCPeerConnection.peer.onaddstream @ jquery.FSRTC.js:819 jquery.verto.js:2103 Uncaught TypeError: Cannot read property 'sinkId' of undefined at $.verto.dialog.setAudioPlaybackDevice (jquery.verto.js:2103) at jquery.verto.js:2173 TypeError: Cannot read property 'sinkId' of undefined at $.verto.dialog.setAudioPlaybackDevice (jquery.verto.js:2103) at jquery.verto.js:2173 $.verto.dialog.setAudioPlaybackDevice @ jquery.verto.js:2103 (anonymous) @ jquery.verto.js:2173 setTimeout (async) $.verto.dialog.setState @ jquery.verto.js:2172 $.verto.dialog.processReply @ jquery.verto.js:2225 (anonymous) @ jquery.verto.js:2068 $.JsonRpcClient._wsOnMessage @ jquery.jsonrpcclient.js:460 wsOnMessage @ jquery.jsonrpcclient.js:82 > On 13 Sep 2017, at 14:02, Rick Jarvis wrote: > > I was missing the verto parameter in the user file, thanks! > > Following Italo’s howto: https://evoluxbr.github.io/verto-docs/ with a vanilla-based config. I have calls connecting now (yay!), but no audio. Am I correct in thinking that the useMic and useSpeak parameters should take the long ID strings that are shown in the console when verto serialises all the audio devices? > >> On 13 Sep 2017, at 08:45, António Silva > wrote: >> >> Hi, >> >> you don't get logs in dialplan because the call is not hitting dialplan, it looks stuck in the client. The method that is sent is not valid or is empty. Are you using the verto demo client with vanilla config? >> >> >> Also did you enable verto in jsonrpc parameters for that user, params: >> >> >> >> >> >> >> On 09/13/2017 12:36 AM, Rick Jarvis wrote: >>> Anyone got any ideas? Desperate to get this working! :) That console log: https://pasteboard.co/GK2exjV.png >>> >>> >>>> On 12 Sep 2017, at 15:36, Rick Jarvis > wrote: >>>> >>>> Resending with link to image (sorry I know images aren’t ideal for mailing lists, but the format goes if I copy&paste)... >>>> >>>> From: Rick Jarvis > >>>> Subject: Re: [Freeswitch-users] Verto >>>> Date: 12 September 2017 at 13:53:10 BST >>>> To: FreeSWITCH Users Help > >>>> >>>> Definitely making progress... >>>> >>>> Can anyone tell me what’s going on here, this is when I try and make a call (it’s an internal call, wondered if it’s hitting the wrong context or something, but the only logging I’m getting is from mod_verto, so I don’t think it’s getting that far): >>>> >>>> Link to Chrome console output: >>>> https://pasteboard.co/GK2exjV.png >>>> >>>> Verto console output: >>>> "jsonrpc": "2.0", >>>> "id": 5, >>>> "error": { >>>> "code": -32601, >>>> "message": "Invalid Method, Missing Method or Permission Denied" >>>> } >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> -- >> Saludos / Regards / Cumprimentos >> António Silva >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From rick at magicmail.mooo.com Wed Sep 13 14:31:45 2017 From: rick at magicmail.mooo.com (Rick Jarvis) Date: Wed, 13 Sep 2017 15:31:45 +0100 Subject: [Freeswitch-users] Verto In-Reply-To: <7925711E-B6DA-4692-884A-3B3C0A54BE5F@magicmail.mooo.com> References: <637257B6-7F09-41B1-B5F5-621D320619F4@magicmail.mooo.com> <54BDD1AD-AD50-4028-BFF6-1D24D2AD9D32@magicmail.mooo.com> <1CF14F27-6B9C-49E7-9F04-1AB9B86E8CFC@magicmail.mooo.com> <30116e26-f07c-dda8-0593-c16d84f235be@wirelessmundi.com> <80814D79-29AD-4820-99A5-81478D7BD0C9@magicmail.mooo.com> <7925711E-B6DA-4692-884A-3B3C0A54BE5F@magicmail.mooo.com> Message-ID: <0D3361D1-274F-4566-ABA9-FD84283545DF@magicmail.mooo.com> Interestingly I have Verto->ipphone audio on one brand of ipphone (Fanvil), but not on my normal test Yealink handset. Was thinking it must be a codec missing on the Yealink, but adding all the codecs doesn’t help. Still no audio coming back from the ipphones to Verto, from either. But at least I have one audio path working, that’s a start, right? > On 13 Sep 2017, at 14:34, Rick Jarvis wrote: > > And errors in Chrome console, seem to relate: > > Uncaught TypeError: Cannot read property 'srcObject' of undefined > at onRemoteStream (jquery.FSRTC.js:269) > at Object.onRemoteStream (jquery.FSRTC.js:439) > at RTCPeerConnection.peer.onaddstream (jquery.FSRTC.js:819) > onRemoteStream @ jquery.FSRTC.js:269 > onRemoteStream @ jquery.FSRTC.js:439 > RTCPeerConnection.peer.onaddstream @ jquery.FSRTC.js:819 > jquery.verto.js:2103 Uncaught TypeError: Cannot read property 'sinkId' of undefined > at $.verto.dialog.setAudioPlaybackDevice (jquery.verto.js:2103) > at jquery.verto.js:2173 > > TypeError: Cannot read property 'sinkId' of undefined > at $.verto.dialog.setAudioPlaybackDevice (jquery.verto.js:2103) > at jquery.verto.js:2173 > $.verto.dialog.setAudioPlaybackDevice @ jquery.verto.js:2103 > (anonymous) @ jquery.verto.js:2173 > setTimeout (async) > $.verto.dialog.setState @ jquery.verto.js:2172 > $.verto.dialog.processReply @ jquery.verto.js:2225 > (anonymous) @ jquery.verto.js:2068 > $.JsonRpcClient._wsOnMessage @ jquery.jsonrpcclient.js:460 > wsOnMessage @ jquery.jsonrpcclient.js:82 > > >> On 13 Sep 2017, at 14:02, Rick Jarvis > wrote: >> >> I was missing the verto parameter in the user file, thanks! >> >> Following Italo’s howto: https://evoluxbr.github.io/verto-docs/ with a vanilla-based config. I have calls connecting now (yay!), but no audio. Am I correct in thinking that the useMic and useSpeak parameters should take the long ID strings that are shown in the console when verto serialises all the audio devices? >> >>> On 13 Sep 2017, at 08:45, António Silva > wrote: >>> >>> Hi, >>> >>> you don't get logs in dialplan because the call is not hitting dialplan, it looks stuck in the client. The method that is sent is not valid or is empty. Are you using the verto demo client with vanilla config? >>> >>> >>> Also did you enable verto in jsonrpc parameters for that user, params: >>> >>> >>> >>> >>> >>> >>> On 09/13/2017 12:36 AM, Rick Jarvis wrote: >>>> Anyone got any ideas? Desperate to get this working! :) That console log: https://pasteboard.co/GK2exjV.png >>>> >>>> >>>>> On 12 Sep 2017, at 15:36, Rick Jarvis > wrote: >>>>> >>>>> Resending with link to image (sorry I know images aren’t ideal for mailing lists, but the format goes if I copy&paste)... >>>>> >>>>> From: Rick Jarvis > >>>>> Subject: Re: [Freeswitch-users] Verto >>>>> Date: 12 September 2017 at 13:53:10 BST >>>>> To: FreeSWITCH Users Help > >>>>> >>>>> Definitely making progress... >>>>> >>>>> Can anyone tell me what’s going on here, this is when I try and make a call (it’s an internal call, wondered if it’s hitting the wrong context or something, but the only logging I’m getting is from mod_verto, so I don’t think it’s getting that far): >>>>> >>>>> Link to Chrome console output: >>>>> https://pasteboard.co/GK2exjV.png >>>>> >>>>> Verto console output: >>>>> "jsonrpc": "2.0", >>>>> "id": 5, >>>>> "error": { >>>>> "code": -32601, >>>>> "message": "Invalid Method, Missing Method or Permission Denied" >>>>> } >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> -- >>> Saludos / Regards / Cumprimentos >>> António Silva >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From josefu at gmail.com Wed Sep 13 15:14:15 2017 From: josefu at gmail.com (=?UTF-8?Q?Jose_Fco=2E_Irles_Dur=C3=A1?=) Date: Wed, 13 Sep 2017 17:14:15 +0200 Subject: [Freeswitch-users] Implement PushKit or CallKit to call iOS app closed In-Reply-To: References: Message-ID: Hi jose, I can't send you the code because it's from my customer, but I have this scripts: * [1] Script that handles the initial call: 1. send push with the uuid of the call to ios (with mod_curl to a own web service) 2. wait to a custom event that sends the other script[2] with the function "pop" from freeswitch.EventConsumer api. 3. when event arrives, that events include the uuid of the other leg and the script bridge the two legs. * [2] Script that manages the call sended by the app in the phone 0. The app sends a call with a custom header with the uuid received in the push message. 1. The script handle the call and build a custom event with the uuid received and add to the event its channel uuid. 2. fire the event Really, it is a bit more complex (our logic sends a ringing event and answer event also), but basically this is the idea. Best regards 2017-09-08 10:47 GMT+02:00 Jose David Jurado Alonso : > Hi Jose, > > Thank you very much for commenting your experience with this case. I don't > find much information about it. > > Can you explain me a bit more in detail how you parked the call? It will do > the LUA script I suppose. > > If you could show or send me the LUA script I would be of great help ... > > Thanks, > > > 2017-09-07 17:02 GMT+02:00 Jose Fco. Irles Durá : >> >> We have implemented a push system to iOS/Android devices with >> FreeSWITCH following the next steps: >> >> 1. Incomming call to user bob in FreeSWITCH >> 2. FreeSWITCH park the call >> 3. Our manager software (xml_curl) sends a push to bob's iOS device >> 4. Notification arrives to iOS device and the softphone call to a uri >> sended in the push notification (this uri is the server that >> previously has parked the call) >> 5. FreeSWITCH bridge the two legs (before this, manage the "ringing" >> event) >> >> The FreeSWITCH part is developed in a simple script in lua. >> >> Best regards >> >> >> >> 2017-09-07 10:55 GMT+02:00 Jose David Jurado Alonso >> : >> > Hi, >> > >> > I can't find any information over how to implement PushKit or Callkit to >> > call an iOS mobile phone that has the SIP client application closed or >> > in >> > the background. >> > >> > When the application is closed it is obviously that SIP account isn't >> > registered and the FS server no send the call. >> > >> > Has anyone worked with this or know how to implement it? >> > >> > I tried with "> > value="NDLB-connectile-dysfunction"/>" option and this temporally solved >> > when the application is alive in background. >> > >> > Thanks, >> > >> > >> > >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://confluence.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> >> >> >> -- >> Jose Fco. Irles Durá >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Jose Fco. Irles Durá From chad at apartmentlines.com Wed Sep 13 16:58:20 2017 From: chad at apartmentlines.com (Chad Phillips) Date: Wed, 13 Sep 2017 09:58:20 -0700 Subject: [Freeswitch-users] Tracking down calls with spoofed Caller ID Message-ID: This question isn’t directly related to FreeSWITCH, and, I’m curious what others in this community have learned about dealing with incoming ‘spam’ phone calls… For many months now I’ve been getting travel-related spam phone calls to my personal cell. I figured the caller ID was spoofed, so would just hang up, but a few days ago I decided to call one of the numbers back to see what would happen. A woman answered, and I asked her to remove me from their call list, to which she angrily responded “no, you are the one who has been calling me, and I’ve contacted the police!” It was then that I realized the spammers were using caller ID numbers of other people they were spamming…as if this industry could get any slimier. I would love to bust these jerks, but I’m not sure it’s even possible to trace the origins of these calls. It seems futile to block incoming calls based on caller ID, because they change all the time. How do the telephony bad-asses in this community handle such shenanigans? Chad -------------- next part -------------- An HTML attachment was scrubbed... URL: From rbetancor at gmail.com Wed Sep 13 17:11:22 2017 From: rbetancor at gmail.com (=?UTF-8?Q?Ra=C3=BAl_Alexis_Betancor_Santana?=) Date: Wed, 13 Sep 2017 18:11:22 +0100 Subject: [Freeswitch-users] Tracking down calls with spoofed Caller ID In-Reply-To: References: Message-ID: The policy that we apply is easy: - We do not accept any external (off-net) call that comes with a CallerID belonging to US - We allow customers to set their own blacklist. So they could block the CallerIDs they want - If we receive any complains about one of our CallerID been using for spam, we do a check, if the spam comes from the CallerID owner, we just terminated their contract, because of violations of our service usage policies. If it cames fron external sources using that CallerID and the owner blames, the only thing we could do is just to notify the authorities and let the customer to change the CallerID. 2017-09-13 17:58 GMT+01:00 Chad Phillips : > This question isn’t directly related to FreeSWITCH, and, I’m curious what > others in this community have learned about dealing with incoming ‘spam’ > phone calls… > > For many months now I’ve been getting travel-related spam phone calls to > my personal cell. I figured the caller ID was spoofed, so would just hang > up, but a few days ago I decided to call one of the numbers back to see > what would happen. > > A woman answered, and I asked her to remove me from their call list, to > which she angrily responded “no, you are the one who has been calling me, > and I’ve contacted the police!” > > It was then that I realized the spammers were using caller ID numbers of > other people they were spamming…as if this industry could get any slimier. > > I would love to bust these jerks, but I’m not sure it’s even possible to > trace the origins of these calls. It seems futile to block incoming calls > based on caller ID, because they change all the time. > > How do the telephony bad-asses in this community handle such shenanigans? > > Chad > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.com Wed Sep 13 17:13:22 2017 From: brian at freeswitch.com (Brian West) Date: Wed, 13 Sep 2017 12:13:22 -0500 Subject: [Freeswitch-users] Tracking down calls with spoofed Caller ID In-Reply-To: References: Message-ID: I have seen similar, I've also seen them spoof numbers of my husband and sister to get to me, Its frustrating because I can't block them. I've reported them till I'm blue in the face. /b On Wed, Sep 13, 2017 at 11:58 AM, Chad Phillips wrote: > This question isn’t directly related to FreeSWITCH, and, I’m curious what > others in this community have learned about dealing with incoming ‘spam’ > phone calls… > > For many months now I’ve been getting travel-related spam phone calls to > my personal cell. I figured the caller ID was spoofed, so would just hang > up, but a few days ago I decided to call one of the numbers back to see > what would happen. > > A woman answered, and I asked her to remove me from their call list, to > which she angrily responded “no, you are the one who has been calling me, > and I’ve contacted the police!” > > It was then that I realized the spammers were using caller ID numbers of > other people they were spamming…as if this industry could get any slimier. > > I would love to bust these jerks, but I’m not sure it’s even possible to > trace the origins of these calls. It seems futile to block incoming calls > based on caller ID, because they change all the time. > > How do the telephony bad-asses in this community handle such shenanigans? > > Chad > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.com *Twitter: @FreeSWITCH , @cluecon* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) -------------- next part -------------- An HTML attachment was scrubbed... URL: From mario_fs at mgtech.com Wed Sep 13 17:17:54 2017 From: mario_fs at mgtech.com (Mario G) Date: Wed, 13 Sep 2017 10:17:54 -0700 Subject: [Freeswitch-users] Tracking down calls with spoofed Caller ID In-Reply-To: References: Message-ID: <96D7EE18-3BB3-47BD-B335-6284523E0242@mgtech.com> Same here, it used to be easy to block calls form area codes no one should be calling from. But in the last year the spammers are calling frequently with spoofed numbers from our local area code so you can’t really block them. The sad thing is if you report it as a spam number, it messes someone up with a legitimate number. Mario G > On Sep 13, 2017, at 10:13 AM, Brian West wrote: > > I have seen similar, I've also seen them spoof numbers of my husband and sister to get to me, Its frustrating because I can't block them. I've reported them till I'm blue in the face. > > /b > > > On Wed, Sep 13, 2017 at 11:58 AM, Chad Phillips > wrote: > This question isn’t directly related to FreeSWITCH, and, I’m curious what others in this community have learned about dealing with incoming ‘spam’ phone calls… > > For many months now I’ve been getting travel-related spam phone calls to my personal cell. I figured the caller ID was spoofed, so would just hang up, but a few days ago I decided to call one of the numbers back to see what would happen. > > A woman answered, and I asked her to remove me from their call list, to which she angrily responded “no, you are the one who has been calling me, and I’ve contacted the police!” > > It was then that I realized the spammers were using caller ID numbers of other people they were spamming…as if this industry could get any slimier. > > I would love to bust these jerks, but I’m not sure it’s even possible to trace the origins of these calls. It seems futile to block incoming calls based on caller ID, because they change all the time. > > How do the telephony bad-asses in this community handle such shenanigans? > > Chad > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > Brian West > brian at freeswitch.com > Twitter: @FreeSWITCH , @cluecon > > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > Got Bugs? Report them here ! | Reddit: /r/freeswitch > T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From rbetancor at gmail.com Wed Sep 13 18:14:30 2017 From: rbetancor at gmail.com (=?UTF-8?Q?Ra=C3=BAl_Alexis_Betancor_Santana?=) Date: Wed, 13 Sep 2017 19:14:30 +0100 Subject: [Freeswitch-users] Tracking down calls with spoofed Caller ID In-Reply-To: <96D7EE18-3BB3-47BD-B335-6284523E0242@mgtech.com> References: <96D7EE18-3BB3-47BD-B335-6284523E0242@mgtech.com> Message-ID: But your carrier, the one giving the service TO YOUR line, will know from where the call comes ... and could send that trace to the authorities or do some upwaters guessing with their peering carriers. The tools are there to catch them. We helped to close an spamming bussines last year, we got to the point of identifing the carrier who was selling the service to them ... and they received a penalty fee from their local telco commission, for allowing customer to play with the callerid freely. 2017-09-13 18:17 GMT+01:00 Mario G : > Same here, it used to be easy to block calls form area codes no one should > be calling from. But in the last year the spammers are calling > frequently with spoofed numbers from our local area code so you can’t > really block them. The sad thing is if you report it as a spam number, it > messes someone up with a legitimate number. > Mario G > > > > On Sep 13, 2017, at 10:13 AM, Brian West wrote: > > I have seen similar, I've also seen them spoof numbers of my husband and > sister to get to me, Its frustrating because I can't block them. I've > reported them till I'm blue in the face. > > /b > > > On Wed, Sep 13, 2017 at 11:58 AM, Chad Phillips > wrote: > >> This question isn’t directly related to FreeSWITCH, and, I’m curious what >> others in this community have learned about dealing with incoming ‘spam’ >> phone calls… >> >> For many months now I’ve been getting travel-related spam phone calls to >> my personal cell. I figured the caller ID was spoofed, so would just hang >> up, but a few days ago I decided to call one of the numbers back to see >> what would happen. >> >> A woman answered, and I asked her to remove me from their call list, to >> which she angrily responded “no, you are the one who has been calling me, >> and I’ve contacted the police!” >> >> It was then that I realized the spammers were using caller ID numbers of >> other people they were spamming…as if this industry could get any slimier. >> >> I would love to bust these jerks, but I’m not sure it’s even possible to >> trace the origins of these calls. It seems futile to block incoming calls >> based on caller ID, because they change all the time. >> >> How do the telephony bad-asses in this community handle such shenanigans? >> >> Chad >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > > *Brian West* > brian at freeswitch.com > > *Twitter: @FreeSWITCH , @cluecon* > > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > > Got Bugs? Report them here ! | Reddit: > /r/freeswitch > > *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002> > | *M:*+1918424WEST (9378) > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From steveayre at gmail.com Wed Sep 13 18:36:26 2017 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 13 Sep 2017 19:36:26 +0100 Subject: [Freeswitch-users] log libsofia In-Reply-To: References: Message-ID: Try 'sofia tracelevel debug' - it sets the level used to log those messages. The default is console, anything else will go to the log file. On 13 September 2017 at 04:10, Sergey Safarov wrote: > Is possible write to log file info produced by by libsofia tport? > I can see this info using "sofia loglevel tport 9", but how to write this > info to file? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From rmundkowsky at ets.org Wed Sep 13 18:24:13 2017 From: rmundkowsky at ets.org (Mundkowsky, Robert) Date: Wed, 13 Sep 2017 18:24:13 +0000 Subject: [Freeswitch-users] Tracking down calls with spoofed Caller ID In-Reply-To: References: <96D7EE18-3BB3-47BD-B335-6284523E0242@mgtech.com> Message-ID: Last time I contacted a major carrier about blocking callers that have no caller id, they play dumb saying “oh we have not way to know how to do that.” Actually ended up installing a app on my cell phone to block such calls which kind of worked. I doubt’t they will block any type of SPAM or etc. Since they are not charging us per call, I have no clue why they do not have services to block such calls, maybe they make a lot of money off SPAMmers I guess. Robert From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Raúl Alexis Betancor Santana Sent: Wednesday, September 13, 2017 2:15 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Tracking down calls with spoofed Caller ID But your carrier, the one giving the service TO YOUR line, will know from where the call comes ... and could send that trace to the authorities or do some upwaters guessing with their peering carriers. The tools are there to catch them. We helped to close an spamming bussines last year, we got to the point of identifing the carrier who was selling the service to them ... and they received a penalty fee from their local telco commission, for allowing customer to play with the callerid freely. 2017-09-13 18:17 GMT+01:00 Mario G >: Same here, it used to be easy to block calls form area codes no one should be calling from. But in the last year the spammers are calling frequently with spoofed numbers from our local area code so you can’t really block them. The sad thing is if you report it as a spam number, it messes someone up with a legitimate number. Mario G On Sep 13, 2017, at 10:13 AM, Brian West > wrote: I have seen similar, I've also seen them spoof numbers of my husband and sister to get to me, Its frustrating because I can't block them. I've reported them till I'm blue in the face. /b On Wed, Sep 13, 2017 at 11:58 AM, Chad Phillips > wrote: This question isn’t directly related to FreeSWITCH, and, I’m curious what others in this community have learned about dealing with incoming ‘spam’ phone calls… For many months now I’ve been getting travel-related spam phone calls to my personal cell. I figured the caller ID was spoofed, so would just hang up, but a few days ago I decided to call one of the numbers back to see what would happen. A woman answered, and I asked her to remove me from their call list, to which she angrily responded “no, you are the one who has been calling me, and I’ve contacted the police!” It was then that I realized the spammers were using caller ID numbers of other people they were spamming…as if this industry could get any slimier. I would love to bust these jerks, but I’m not sure it’s even possible to trace the origins of these calls. It seems futile to block incoming calls based on caller ID, because they change all the time. How do the telephony bad-asses in this community handle such shenanigans? Chad _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Brian West brian at freeswitch.com Twitter: @FreeSWITCH , @cluecon http://www.freeswitchbook.com http://www.freeswitchcookbook.com Got Bugs? Report them here! | Reddit: /r/freeswitch T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ________________________________ This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. Thank you for your compliance. ________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: From rbetancor at gmail.com Wed Sep 13 18:57:34 2017 From: rbetancor at gmail.com (=?UTF-8?Q?Ra=C3=BAl_Alexis_Betancor_Santana?=) Date: Wed, 13 Sep 2017 19:57:34 +0100 Subject: [Freeswitch-users] Tracking down calls with spoofed Caller ID In-Reply-To: References: <96D7EE18-3BB3-47BD-B335-6284523E0242@mgtech.com> Message-ID: At least here, on Spain ... all carriers offer you the service to block incoming annonymous calls and also block incoming calls based on CallerID ... tradicional telcos limit you to 10 numbers ... but SIP telcos do no put any limit on that and either any smartphone could block any call you tell them ... there is apps either that you could install that check the CallerID agains know web portals where people report spammers of disturbing numbers ... so far easier to you. 2017-09-13 19:24 GMT+01:00 Mundkowsky, Robert : > Last time I contacted a major carrier about blocking callers that have no > caller id, they play dumb saying “oh we have not way to know how to do > that.” Actually ended up installing a app on my cell phone to block such > calls which kind of worked. > > > > I doubt’t they will block any type of SPAM or etc. Since they are not > charging us per call, I have no clue why they do not have services to block > such calls, maybe they make a lot of money off SPAMmers I guess. > > > > > > Robert > > > > *From:* FreeSWITCH-users [mailto:freeswitch-users- > bounces at lists.freeswitch.org] *On Behalf Of *Raúl Alexis Betancor Santana > *Sent:* Wednesday, September 13, 2017 2:15 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Tracking down calls with spoofed Caller > ID > > > > But your carrier, the one giving the service TO YOUR line, will know from > where the call comes ... and could send that trace to the authorities or do > some upwaters guessing with their peering carriers. > > The tools are there to catch them. > > We helped to close an spamming bussines last year, we got to the point of > identifing the carrier who was selling the service to them ... and they > received a penalty fee from their local telco commission, for allowing > customer to play with the callerid freely. > > > > 2017-09-13 18:17 GMT+01:00 Mario G : > > Same here, it used to be easy to block calls form area codes no one should > be calling from. But in the last year the spammers are calling > frequently with spoofed numbers from our local area code so you can’t > really block them. The sad thing is if you report it as a spam number, it > messes someone up with a legitimate number. > > Mario G > > > > > > > > On Sep 13, 2017, at 10:13 AM, Brian West wrote: > > > > I have seen similar, I've also seen them spoof numbers of my husband and > sister to get to me, Its frustrating because I can't block them. I've > reported them till I'm blue in the face. > > > > /b > > > > > > On Wed, Sep 13, 2017 at 11:58 AM, Chad Phillips > wrote: > > This question isn’t directly related to FreeSWITCH, and, I’m curious what > others in this community have learned about dealing with incoming ‘spam’ > phone calls… > > > > For many months now I’ve been getting travel-related spam phone calls to > my personal cell. I figured the caller ID was spoofed, so would just hang > up, but a few days ago I decided to call one of the numbers back to see > what would happen. > > > > A woman answered, and I asked her to remove me from their call list, to > which she angrily responded “no, you are the one who has been calling me, > and I’ve contacted the police!” > > > > It was then that I realized the spammers were using caller ID numbers of > other people they were spamming…as if this industry could get any slimier. > > > > I would love to bust these jerks, but I’m not sure it’s even possible to > trace the origins of these calls. It seems futile to block incoming calls > based on caller ID, because they change all the time. > > > > How do the telephony bad-asses in this community handle such shenanigans? > > > > Chad > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > Official FreeSWITCH Sites > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > -- > > *Brian West* > brian at freeswitch.com > > *Twitter: @FreeSWITCH , @cluecon* > > http://www.freeswitchbook.com > > > http://www.freeswitchcookbook.com > > > > Got Bugs? Report them here > ! > | Reddit: /r/freeswitch > > > *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002> > | *M:*+1918424WEST (9378) > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > Official FreeSWITCH Sites > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > Official FreeSWITCH Sites > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > ------------------------------ > > This e-mail and any files transmitted with it may contain privileged or > confidential information. It is solely for use by the individual for whom > it is intended, even if addressed incorrectly. If you received this e-mail > in error, please notify the sender; do not disclose, copy, distribute, or > take any action in reliance on the contents of this information; and delete > it from your system. Any other use of this e-mail is prohibited. > > Thank you for your compliance. > ------------------------------ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From s.safarov at gmail.com Wed Sep 13 19:15:14 2017 From: s.safarov at gmail.com (Sergey Safarov) Date: Wed, 13 Sep 2017 19:15:14 +0000 Subject: [Freeswitch-users] log libsofia In-Reply-To: References: Message-ID: Thanks Steven I now checked logs - looks like now logs contains required info. Sergey ср, 13 сент. 2017 г. в 21:37, Steven Ayre : > Try 'sofia tracelevel debug' - it sets the level used to log those > messages. The default is console, anything else will go to the log file. > > > On 13 September 2017 at 04:10, Sergey Safarov wrote: > >> Is possible write to log file info produced by by libsofia tport? >> I can see this info using "sofia loglevel tport 9", but how to write this >> info to file? >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From clive at lansink.co.nz Wed Sep 13 19:17:14 2017 From: clive at lansink.co.nz (Clive Lansink) Date: Thu, 14 Sep 2017 07:17:14 +1200 Subject: [Freeswitch-users] PortAudio and Windows devices Message-ID: An embedded and charset-unspecified text was scrubbed... Name: not available URL: From bipin at xbipin.com Wed Sep 13 19:49:24 2017 From: bipin at xbipin.com (Bipin Patel) Date: Wed, 13 Sep 2017 23:49:24 +0400 Subject: [Freeswitch-users] Tracking down calls with spoofed Caller ID In-Reply-To: References: Message-ID: <15e7cc9f5a0.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com> The way you can figure out if the person is using VoIP with a faked cli is normally in most countries when actual ppl call numbers, the destination party see cli starting with 0 followed by area code or mobile code but when ppl use VoIP, most carriers prefix the cli with a + by default so no idea if this is the case there but if I see a local cli starting with + followed by country code etc then I know it's a VoIP spammer call. On September 13, 2017 9:00:48 PM Chad Phillips wrote: > This question isn’t directly related to FreeSWITCH, and, I’m curious what > others in this community have learned about dealing with incoming ‘spam’ > phone calls… > > For many months now I’ve been getting travel-related spam phone calls to my > personal cell. I figured the caller ID was spoofed, so would just hang up, > but a few days ago I decided to call one of the numbers back to see what > would happen. > > A woman answered, and I asked her to remove me from their call list, to > which she angrily responded “no, you are the one who has been calling me, > and I’ve contacted the police!” > > It was then that I realized the spammers were using caller ID numbers of > other people they were spamming…as if this industry could get any slimier. > > I would love to bust these jerks, but I’m not sure it’s even possible to > trace the origins of these calls. It seems futile to block incoming calls > based on caller ID, because they change all the time. > > How do the telephony bad-asses in this community handle such shenanigans? > > Chad > > > > ---------- > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From nandy1925 at gmail.com Thu Sep 14 02:27:33 2017 From: nandy1925 at gmail.com (Nandy Dagondon) Date: Thu, 14 Sep 2017 02:27:33 +0000 Subject: [Freeswitch-users] OSLEC on FreeTDM on Sangoma A102 Message-ID: Hello folks, I encountered echo problem with FreeTDM Wanpipe R2 with Sangoma A102 in an insert-and-drop setup. Anyone tried OSLEC (DAHDI) ? Is it working? Tks, /Nandy -------------- next part -------------- An HTML attachment was scrubbed... URL: From krice at tollfreegateway.com Thu Sep 14 02:45:38 2017 From: krice at tollfreegateway.com (Ken Rice) Date: Wed, 13 Sep 2017 21:45:38 -0500 Subject: [Freeswitch-users] OSLEC on FreeTDM on Sangoma A102 In-Reply-To: References: Message-ID: <5DE8158B-120D-4332-959A-D4BAA8646183@tollfreegateway.com> indoubt anyone has integrated oslec with freetd as its gpl thus not license compatible Sent from my iPhone > On Sep 13, 2017, at 21:27, Nandy Dagondon wrote: > > > Hello folks, > > I encountered echo problem with FreeTDM Wanpipe R2 with Sangoma A102 in an insert-and-drop setup. Anyone tried OSLEC (DAHDI) ? Is it working? > > Tks, > /Nandy > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From richard at treeboxsolutions.com Thu Sep 14 05:22:36 2017 From: richard at treeboxsolutions.com (Richard Chan) Date: Thu, 14 Sep 2017 13:22:36 +0800 Subject: [Freeswitch-users] Chrome/VC without STUN sends 0.0.0.0 and gets INCOMPATIBLE_DESTINATION Message-ID: Hi, I'd like to get your opinion on whether this WebRTC failure to connect points to a Chrome/VC or FS bug. FreeSWTICH: 1.6.19 Verto Communicator: from v1.8 branch Chrome: 59-61 Reproducer: 0. Chrome and FS on a local LAN 1. In VC, disable STUN as environment is local LAN 2. Chrome sends 2017-09-14 11:48:33.658249 [DEBUG] mod_rtc.c:389 () State Change CS_NEW -> CS_INIT 2017-09-14 11:48:33.658249 [DEBUG] switch_core_session.c:615 N/A set UUID=b138714b-740a-de23-101c-c76d5676a6d4 2017-09-14 11:48:33.658249 [NOTICE] switch_channel.c:1104 New Channel verto.rtc/3000 [b138714b-740a-de23-101c-c76d5676a6d4] 2017-09-14 11:48:33.658249 [DEBUG] mod_verto.c:3661 Remote SDP verto.rtc/3000: v=0 o=- 3881351537341052870 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE audio video a=msid-semantic: WMS Tub3STBaDw6tNMrINhAyvJBWK96IeIanerjj m=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126 c=IN IP4 0.0.0.0 a=rtcp:9 IN IP4 0.0.0.0 a=ice-ufrag:VU1P a=ice-pwd:davKC7PLYeQ04ZkN4TNvGo9s a=ice-options:trickle a=fingerprint:sha-256 79:A5:79:AE:62:B9:7F:00:5F:0C:04:C1:18:81:C4:09:ED:47:3B:1B:0A:3C:2F:F2:7B:6F:D5:E7:3C:DA:E0:2C a=setup:actpass a=mid:audio a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=sendrecv a=rtcp-mux 3. FS replies [DEBUG] switch_core_media.c:3523 Searching for rtp candidate. [DEBUG] switch_core_media.c:3523 Searching for rtcp candidate. [DEBUG] switch_core_media.c:3567 verto.rtc/3000 no suitable candidates found. [NOTICE] switch_channel.c:3812 Hangup verto.rtc/3000 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] [DEBUG] switch_core_session.c:2815 verto.rtc/3000 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 4. With STUN on, it connects successfully. Chrome sends the external NAT address (STUN discovered). This seems unnecessary for a local LAN. 5. For Firefox+VC without STUN, the verto.invite contains all the local LAN addresses and it connect successfully. Any suggestions? Should Chrome+VC without be expected to work w/o STUN on a local LAN while sending c=IN IP4 0.0.0.0? -- Richard Chan -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Thu Sep 14 05:39:14 2017 From: mike at jerris.com (Michael Jerris) Date: Thu, 14 Sep 2017 05:39:14 +0000 Subject: [Freeswitch-users] OSLEC on FreeTDM on Sangoma A102 In-Reply-To: <5DE8158B-120D-4332-959A-D4BAA8646183@tollfreegateway.com> References: <5DE8158B-120D-4332-959A-D4BAA8646183@tollfreegateway.com> Message-ID: Please contact sangoma for assistance with issues with their cards. On Wed, Sep 13, 2017 at 10:46 PM Ken Rice wrote: > indoubt anyone has integrated oslec with freetd as its gpl thus not > license compatible > > Sent from my iPhone > > > On Sep 13, 2017, at 21:27, Nandy Dagondon wrote: > > > > > > Hello folks, > > > > I encountered echo problem with FreeTDM Wanpipe R2 with Sangoma A102 in > an insert-and-drop setup. Anyone tried OSLEC (DAHDI) ? Is it working? > > > > Tks, > > /Nandy > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From italo at freeswitch.org Thu Sep 14 10:51:35 2017 From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=) Date: Thu, 14 Sep 2017 10:51:35 +0000 Subject: [Freeswitch-users] Chrome/VC without STUN sends 0.0.0.0 and gets INCOMPATIBLE_DESTINATION In-Reply-To: References: Message-ID: Which chrome version? Try restarting it or on canary to see if there's differences. Em qui, 14 de set de 2017 às 02:24, Richard Chan < richard at treeboxsolutions.com> escreveu: > Hi, > > I'd like to get your opinion on whether this WebRTC failure to connect > points to a Chrome/VC or FS bug. > > FreeSWTICH: 1.6.19 > Verto Communicator: from v1.8 branch > Chrome: 59-61 > > Reproducer: > 0. Chrome and FS on a local LAN > 1. In VC, disable STUN as environment is local LAN > 2. Chrome sends > 2017-09-14 11:48:33.658249 [DEBUG] mod_rtc.c:389 () State Change CS_NEW -> > CS_INIT > 2017-09-14 11:48:33.658249 [DEBUG] switch_core_session.c:615 N/A set > UUID=b138714b-740a-de23-101c-c76d5676a6d4 > 2017-09-14 11:48:33.658249 [NOTICE] switch_channel.c:1104 New Channel > verto.rtc/3000 [b138714b-740a-de23-101c-c76d5676a6d4] > 2017-09-14 11:48:33.658249 [DEBUG] mod_verto.c:3661 Remote SDP > verto.rtc/3000: > v=0 > o=- 3881351537341052870 2 IN IP4 127.0.0.1 > s=- > t=0 0 > a=group:BUNDLE audio video > a=msid-semantic: WMS Tub3STBaDw6tNMrINhAyvJBWK96IeIanerjj > m=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126 > c=IN IP4 0.0.0.0 > a=rtcp:9 IN IP4 0.0.0.0 > a=ice-ufrag:VU1P > a=ice-pwd:davKC7PLYeQ04ZkN4TNvGo9s > a=ice-options:trickle > a=fingerprint:sha-256 > 79:A5:79:AE:62:B9:7F:00:5F:0C:04:C1:18:81:C4:09:ED:47:3B:1B:0A:3C:2F:F2:7B:6F:D5:E7:3C:DA:E0:2C > a=setup:actpass > a=mid:audio > a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level > a=sendrecv > a=rtcp-mux > > 3. FS replies > > [DEBUG] switch_core_media.c:3523 Searching for rtp candidate. > [DEBUG] switch_core_media.c:3523 Searching for rtcp candidate. > [DEBUG] switch_core_media.c:3567 verto.rtc/3000 no suitable candidates > found. > [NOTICE] switch_channel.c:3812 Hangup verto.rtc/3000 [CS_EXECUTE] > [INCOMPATIBLE_DESTINATION] > [DEBUG] switch_core_session.c:2815 verto.rtc/3000 skip receive message > [APPLICATION_EXEC_COMPLETE] (channel is hungup already) > > 4. With STUN on, it connects successfully. Chrome sends the external NAT > address (STUN discovered). This seems unnecessary for a local LAN. > > 5. For Firefox+VC without STUN, the verto.invite contains all the local > LAN addresses and it connect successfully. > > Any suggestions? Should Chrome+VC without be expected to work w/o STUN on > a local LAN while sending c=IN IP4 0.0.0.0? > > > > -- > Richard Chan > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From richard at treeboxsolutions.com Thu Sep 14 12:12:30 2017 From: richard at treeboxsolutions.com (Richard Chan) Date: Thu, 14 Sep 2017 20:12:30 +0800 Subject: [Freeswitch-users] [SOLVED] Re: Chrome/VC without STUN sends 0.0.0.0 and gets INCOMPATIBLE_DESTINATION Message-ID: Sorry for the noise — it was the Privacy Badger extension in Chrome hiding all the host candidates. https://github.com/EFForg/privacybadger/issues/1099 It seems to be quite an issue with the way it sets chrome.privacy.network.webRTCIPHandlingPolicy Richard -------------- next part -------------- An HTML attachment was scrubbed... URL: From ynasida at gmail.com Thu Sep 14 13:00:56 2017 From: ynasida at gmail.com (Yuriy Nasida) Date: Thu, 14 Sep 2017 16:00:56 +0300 Subject: [Freeswitch-users] vm_cc weird behavior Message-ID: Hi, I noted that FS sends a lot of emails to my email box instead of only one in case I use vm_cc. It's very simple configuration. I just set vm_cc before mod_voicemail is called (according wiki page) and leave voicemail message. For main emails box I recieve only one message as expected. But I have ~60 exactly same messages at email box from vm_cc. In console I also noted that FS really sends emails many times without obvios reason. mod_voicemail.c:2832 Deliver VM to yuriytest2 at mydomain.com mod_voicemail.c:1925 Update MWI: Processing for yuriytest2 at mydomain.com in inbox mod_voicemail.c:1950 Update MWI: Messages Waiting yes mod_voicemail.c:1951 Update MWI: Update Reason NEW mod_voicemail.c:1952 Update MWI: Message Account yuriytest2 at mydomain.com mod_voicemail.c:1953 Update MWI: Voice Message 69/0 switch_utils.c:1180 Emailed file [/tmp/mail.1505386894e5a8] to [ myemailbox at gmail.com] mod_voicemail.c:3042 Sending message to myemailbox at gmail.com Next FS does this again and tries to Deliver VM to yuriytest2 at mydomain.com. Please advice. -------------- next part -------------- An HTML attachment was scrubbed... URL: From freeswitch940 at gmail.com Thu Sep 14 12:37:42 2017 From: freeswitch940 at gmail.com (Freeswitch user) Date: Thu, 14 Sep 2017 18:07:42 +0530 Subject: [Freeswitch-users] JAVA ISSUE Message-ID: Hello Everyone, Sorry to disturb you.I'm getting an error while i'm trying to call java file from dialplan. Error:- [ERR] switch_cpp.cpp:683 object is not initalized i have compiled mod_java successfully. there was no error during make and make install. but it's showing error in below java code. ------------------------------------------- import org.freeswitch.*; import org.freeswitch.swig.*; public class PhoneTest implements FreeswitchScript, DTMFCallback, HangupHook { public PhoneTest() { } public String onDTMF(Object object, int i, String arg) { if (object instanceof String) freeswitch.console_log("notice", "DTMF: " + (String)object + " ARG: " + arg + "\n"); else freeswitch.console_log("notice", "WOW GOT AN EVENT: " + object.toString()); return "true"; } public void onHangup() { freeswitch.console_log("notice", "HANGUP!\n"); } public void run(String sessionUuid, String args) { freeswitch.console_log("notice", "UUID: " + sessionUuid + " ARGS: " + args + "\n"); JavaSession session = null; try { { session = new JavaSession(sessionUuid); session.answer(); // getting Error in This Line freeswitch.console_log("INFO","\nHIII\n"); }catch(Exception ee){freeswitch.console_log("error",ee.getMessage());} finally { if (session != null) session.delete(); } } } -------------- next part -------------- An HTML attachment was scrubbed... URL: From deepikay at iiitd.ac.in Thu Sep 14 09:38:03 2017 From: deepikay at iiitd.ac.in (Deepika Yadav) Date: Thu, 14 Sep 2017 15:08:03 +0530 Subject: [Freeswitch-users] DTMF events not occurring while using VOIP-GSM gateway hardware locally Message-ID: Hi I need to originate calls to Indian mobile numbers from my python ESL script. I used to use service from a company for sending calls from Freeswitch to GSM network for which I set up external SIP profile. But recently I bought my own GOIP hardware, though I am able to make calls but DTMF events are not getting captured on Freeswitch, API like playandGetDigits does not work and there are no logs on Freeswitch console. -- Regards, Deepika https://deepikay.wixsite.com/deepika -------------- next part -------------- An HTML attachment was scrubbed... URL: From deepikay at iiitd.ac.in Thu Sep 14 10:01:28 2017 From: deepikay at iiitd.ac.in (Deepika Yadav) Date: Thu, 14 Sep 2017 15:31:28 +0530 Subject: [Freeswitch-users] Hangup_hook is not getting called in mod_python script Message-ID: Hi, I have implemented a functionality where when a person with his mobile phone number calls a specific number (that of SIM inserted in our GOIP hardware), the incoming is hanged and that person is called back. After the IVR menu starts playing and if in between caller hangs up, hangup_event is not detected and the registered hangup_hook does not get called. Code is as follows; *XML file that starts python script when the person answers the cal*l: *Python Script handler:* def handler(session, args): session.setHangupHook(hangup_hook) session.setInputCallback(input_callback) print('session object value is ', str(session)) callerid = session.getVariable('caller_id_number') call_uuid = session.getVariable('uuid') phone_number = callerid[len(callerid)-10:] initial_greeting(session, phone_number, call_uuid) def hangup_hook(session, what, args=''): """ Must be explicitly set up with session.setHangupHook(hangup_hook). `session` is a session object. `what` is "hangup" or "transfer". `args` is populated if you pass extra args to session.setInputCallback(). """ print('PERSON HANGED UPPPPP...........') freeswitch.consoleLog("INFO", "hangup hook for '%s'\n" % what) -- Regards, Deepika https://deepikay.wixsite.com/deepika -------------- next part -------------- An HTML attachment was scrubbed... URL: From amit.nakum2009 at gmail.com Thu Sep 14 10:49:09 2017 From: amit.nakum2009 at gmail.com (Amit Nakum) Date: Thu, 14 Sep 2017 16:19:09 +0530 Subject: [Freeswitch-users] Fax processing not successful - result (35) Unexpected DCN while waiting for DCS or DIS. Message-ID: Dear users, I am new to freeswitch and i am try to learn fax to email. but i am getting sometime this error and i am fail to received fax. I try to set below parameter for fax service. i try to search solution for this problem and couldn’t found solution. Can anyone guide. Thank Amit Nakum -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Thu Sep 14 15:46:35 2017 From: mike at jerris.com (Michael Jerris) Date: Thu, 14 Sep 2017 11:46:35 -0400 Subject: [Freeswitch-users] Hangup_hook is not getting called in mod_python script In-Reply-To: References: Message-ID: <269C9355-00B2-49D9-BDE6-A85D362826BD@jerris.com> > On Sep 14, 2017, at 6:01 AM, Deepika Yadav wrote: > > Hi, > > I have implemented a functionality where when a person with his mobile phone number calls a specific number (that of SIM inserted in our GOIP hardware), the incoming is hanged and that person is called back. > > After the IVR menu starts playing and if in between caller hangs up, hangup_event is not detected and the registered hangup_hook does not get called. > > Code is as follows; > > XML file that starts python script when the person answers the call: > > > > > > > > > This part is really strange. If you want to execute that python script when the call answers after originating to the endpoint, just make the python the only thing in dial plan, and use ignore_early_media on the originate. It won’t hit the dial plan until answer. As for the rest of your question, it seems like you are saying you don’t get the hanguphook when the python script isn’t running? > Python Script handler: > > def handler(session, args): > > session.setHangupHook(hangup_hook) > session.setInputCallback(input_callback) > > print('session object value is ', str(session)) > > callerid = session.getVariable('caller_id_number') > call_uuid = session.getVariable('uuid') > > phone_number = callerid[len(callerid)-10:] > > initial_greeting(session, phone_number, call_uuid) > > > def hangup_hook(session, what, args=''): > """ > Must be explicitly set up with session.setHangupHook(hangup_hook). > > `session` is a session object. > `what` is "hangup" or "transfer". > `args` is populated if you pass extra args to session.setInputCallback(). > > """ > print('PERSON HANGED UPPPPP...........') > freeswitch.consoleLog("INFO", "hangup hook for '%s'\n" % what) > > -- > Regards, > Deepika > https://deepikay.wixsite.com/deepika > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Thu Sep 14 15:47:24 2017 From: mike at jerris.com (Michael Jerris) Date: Thu, 14 Sep 2017 11:47:24 -0400 Subject: [Freeswitch-users] Fax processing not successful - result (35) Unexpected DCN while waiting for DCS or DIS. In-Reply-To: References: Message-ID: <789D7240-6385-4354-B2B7-CEDB51EF5AC8@jerris.com> is there a reason you are doing fax_enable_t38_request=true? most likely you should bot be setting that. > On Sep 14, 2017, at 6:49 AM, Amit Nakum wrote: > > Dear users, > > I am new to freeswitch and i am try to learn fax to email. but i am getting sometime this error and i am fail to received fax. > > I try to set below parameter for fax service. > > > > i try to search solution for this problem and couldn’t found solution. > > Can anyone guide. > > Thank > Amit Nakum From freeswitch940 at gmail.com Thu Sep 14 16:38:18 2017 From: freeswitch940 at gmail.com (Freeswitch user) Date: Thu, 14 Sep 2017 16:38:18 +0000 Subject: [Freeswitch-users] DTMF events not occurring while using VOIP-GSM gateway hardware locally In-Reply-To: References: Message-ID: Hello Deepika, As per my experience playandGetDigits is channel variable so we can use it with session only. I don't think it will work with ESL. You can use mod_perl for same.. On Thu, 14 Sep 2017 at 9:10 PM, Deepika Yadav wrote: > Hi > > I need to originate calls to Indian mobile numbers from my python ESL > script. I used to use service from a company for sending calls from > Freeswitch to GSM network for which I set up external SIP profile. > > But recently I bought my own GOIP hardware, though I am able to make calls > but DTMF events are not getting captured on Freeswitch, API like > playandGetDigits does not work and there are no logs on Freeswitch console. > > -- > Regards, > Deepika > https://deepikay.wixsite.com/deepika > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From deepikay at iiitd.ac.in Thu Sep 14 17:42:35 2017 From: deepikay at iiitd.ac.in (Deepika Yadav) Date: Thu, 14 Sep 2017 23:12:35 +0530 Subject: [Freeswitch-users] DTMF events not occurring while using VOIP-GSM gateway hardware locally In-Reply-To: References: Message-ID: It is not ESl but a python script only which is called from a XML extension. The code is as follows: def initial_greeting(session, phone_number, channel_uuid): sangoshthi_greeting = "/usr/local/freeswitch/sounds/sangoshthi_ivr_audios/Sangoshthi_greeting.wav" invalid_digit = "/usr/local/freeswitch/sounds/sangoshthi_ivr_audios/Invalid_entry.wav" topic = str(session.playAndGetDigits(1, 1, 5, 5000,"", sangoshthi_greeting, invalid_digit, "[12345*]")) ivr_handler(topic, session, phone_number, channel_uuid) def handler(session, args): session.setHangupHook(hangup_hook) session.setInputCallback(input_callback) print('session object value is ', str(session)) callerid = session.getVariable('caller_id_number') call_uuid = session.getVariable('uuid') phone_number = callerid[len(callerid)-10:] initial_greeting(session, phone_number, call_uuid) On Thu, Sep 14, 2017 at 10:08 PM, Freeswitch user wrote: > Hello Deepika, > > As per my experience playandGetDigits is channel variable so we can use > it with session only. I don't think it will work with ESL. You can use > mod_perl for same.. > > On Thu, 14 Sep 2017 at 9:10 PM, Deepika Yadav > wrote: > >> Hi >> >> I need to originate calls to Indian mobile numbers from my python ESL >> script. I used to use service from a company for sending calls from >> Freeswitch to GSM network for which I set up external SIP profile. >> >> But recently I bought my own GOIP hardware, though I am able to make >> calls but DTMF events are not getting captured on Freeswitch, API like >> playandGetDigits does not work and there are no logs on Freeswitch console. >> >> -- >> Regards, >> Deepika >> https://deepikay.wixsite.com/deepika >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Regards, Deepika https://deepikay.wixsite.com/deepika -------------- next part -------------- An HTML attachment was scrubbed... URL: From deepikay at iiitd.ac.in Thu Sep 14 18:50:12 2017 From: deepikay at iiitd.ac.in (Deepika Yadav) Date: Fri, 15 Sep 2017 00:20:12 +0530 Subject: [Freeswitch-users] Hangup_hook is not getting called in mod_python script In-Reply-To: <269C9355-00B2-49D9-BDE6-A85D362826BD@jerris.com> References: <269C9355-00B2-49D9-BDE6-A85D362826BD@jerris.com> Message-ID: Since it is a callback mechanism. Before passing to the 4449 extension, the user simply dials a fixed mobile number, the extension '1234' passes it to python script incoming_call_handler.py as follows: *1234 extension in public domain* After the* incoming_call_handler.py* hangs it up and initiate a thread to callback def handler(session, args): callerid = session.getVariable('caller_id_number') number = (callerid[len(callerid)-10:]) freeswitch.consoleLog('INFO','Called by %s\n' % number ) session.hangup() new_api_obj = API() new_api_obj.executeString("pyrun python.callback " + callerid) *Callback.py:* It originates the call by transferring to extension 4449 def runtime(arg1): phone_number = arg1[len(arg1)-10:] new_api_obj = API() command_string = "originate sofia/gateway/MySIP/91" + phone_number + " 4449" result = new_api_obj.executeString("reloadxml") return_val = new_api_obj.executeString(command_string) *4449* 4449 calls Sangoshthi_IVR_handler to play IVR menu on call answer Yes, the python script is running successfully the only thing is when caller drops the call the hang u callback registered using hangup_hook does not called. Regards, Deepika On Thu, Sep 14, 2017 at 9:16 PM, Michael Jerris wrote: > > On Sep 14, 2017, at 6:01 AM, Deepika Yadav wrote: > > Hi, > > I have implemented a functionality where when a person with his mobile > phone number calls a specific number (that of SIM inserted in our GOIP > hardware), the incoming is hanged and that person is called back. > > After the IVR menu starts playing and if in between caller hangs up, > hangup_event is not detected and the registered hangup_hook does not get > called. > > Code is as follows; > > *XML file that starts python script when the person answers the cal*l: > > > > > > > > > > > This part is really strange. If you want to execute that python script > when the call answers after originating to the endpoint, just make the > python the only thing in dial plan, and use ignore_early_media on the > originate. It won’t hit the dial plan until answer. As for the rest of > your question, it seems like you are saying you don’t get the hanguphook > when the python script isn’t running? > > *Python Script handler:* > > def handler(session, args): > > session.setHangupHook(hangup_hook) > session.setInputCallback(input_callback) > > print('session object value is ', str(session)) > > callerid = session.getVariable('caller_id_number') > call_uuid = session.getVariable('uuid') > > phone_number = callerid[len(callerid)-10:] > > initial_greeting(session, phone_number, call_uuid) > > > def hangup_hook(session, what, args=''): > """ > Must be explicitly set up with session.setHangupHook(hangup_hook). > > `session` is a session object. > `what` is "hangup" or "transfer". > `args` is populated if you pass extra args to > session.setInputCallback(). > > """ > print('PERSON HANGED UPPPPP...........') > freeswitch.consoleLog("INFO", "hangup hook for '%s'\n" % what) > > -- > Regards, > Deepika > https://deepikay.wixsite.com/deepika > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Regards, Deepika https://deepikay.wixsite.com/deepika -------------- next part -------------- An HTML attachment was scrubbed... URL: From astashov.andrey at gmail.com Fri Sep 15 04:07:34 2017 From: astashov.andrey at gmail.com (=?UTF-8?B?0JDQvdC00YDQtdC5INCQ0YHRgtCw0YjQvtCy?=) Date: Fri, 15 Sep 2017 10:07:34 +0600 Subject: [Freeswitch-users] DTMF events not occurring while using VOIP-GSM gateway hardware locally In-Reply-To: References: Message-ID: Hi I also had a similar problem. The problem was only on some GSM gateways. And on the incoming call, playanddigits recognizes DTMF normally, and the outgoing one does not. It did not work out that way. In the console, when dtmf_verbose is enabled, pressing the buttons is visible. In tcpdump it is seen that it is in RFC that a click comes. Moved to bind_digit_action. Works without problems on all types of locks and in any directions. Here is the discussion and setting of the profile: https://freeswitchforum.com/viewtopic.php?f=6&t=864 2017-09-14 15:38 GMT+06:00 Deepika Yadav : > Hi > > I need to originate calls to Indian mobile numbers from my python ESL > script. I used to use service from a company for sending calls from > Freeswitch to GSM network for which I set up external SIP profile. > > But recently I bought my own GOIP hardware, though I am able to make calls > but DTMF events are not getting captured on Freeswitch, API like > playandGetDigits does not work and there are no logs on Freeswitch console. > > -- > Regards, > Deepika > https://deepikay.wixsite.com/deepika > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From ssinyagin at gmail.com Fri Sep 15 10:01:24 2017 From: ssinyagin at gmail.com (Stanislav Sinyagin) Date: Fri, 15 Sep 2017 12:01:24 +0200 Subject: [Freeswitch-users] debian-unstable on Stretch Message-ID: I'm trying to install the unstable FreeSWITCH on debian Stretch, and I get the following error: # apt-get update Hit:1 http://security.debian.org stretch/updates InRelease Get:3 http://files.freeswitch.org/repo/deb/debian-unstable stretch InRelease [4,595 B] Ign:2 http://cdn-fastly.deb.debian.org/debian stretch InRelease Hit:4 http://cdn-fastly.deb.debian.org/debian stretch Release Ign:3 http://files.freeswitch.org/repo/deb/debian-unstable stretch InRelease Fetched 4,595 B in 1s (3,266 B/s) Reading package lists... Done W: GPG error: http://files.freeswitch.org/repo/deb/debian-unstable stretch InRelease: The following signatures were invalid: 20B06EE621AB150D40F6079FD76EDC7725E010CF W: The repository 'http://files.freeswitch.org/repo/deb/debian-unstable stretch InRelease' is not signed. N: Data from such a repository can't be authenticated and is therefore potentially dangerous to use. N: See apt-secure(8) manpage for repository creation and user configuration details. I imported all public keys that I could find, but it didn't help. Which key was used for signing this repo? From tculjaga at gmail.com Fri Sep 15 11:07:34 2017 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Fri, 15 Sep 2017 13:07:34 +0200 Subject: [Freeswitch-users] DTMF events not occurring while using VOIP-GSM gateway hardware locally In-Reply-To: References: Message-ID: Hi, try tcpdump and than export audio to a file so you can play it ( or play it directly in wireshark ) maybe the GSMGW is sending InBand DTMF who knows :=) also, depending of the GSMGW, used, try to identify what DTMF method is accepting/sending On 15 September 2017 at 06:07, Андрей Асташов wrote: > Hi > > I also had a similar problem. > The problem was only on some GSM gateways. And on the incoming call, > playanddigits recognizes DTMF normally, and the outgoing one does not. > It did not work out that way. > In the console, when dtmf_verbose is enabled, pressing the buttons is > visible. In tcpdump it is seen that it is in RFC that a click comes. > Moved to bind_digit_action. Works without problems on all types of locks > and in any directions. > Here is the discussion and setting of the profile: > https://freeswitchforum.com/viewtopic.php?f=6&t=864 > > 2017-09-14 15:38 GMT+06:00 Deepika Yadav : > >> Hi >> >> I need to originate calls to Indian mobile numbers from my python ESL >> script. I used to use service from a company for sending calls from >> Freeswitch to GSM network for which I set up external SIP profile. >> >> But recently I bought my own GOIP hardware, though I am able to make >> calls but DTMF events are not getting captured on Freeswitch, API like >> playandGetDigits does not work and there are no logs on Freeswitch console. >> >> -- >> Regards, >> Deepika >> https://deepikay.wixsite.com/deepika >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bipin at xbipin.com Fri Sep 15 11:44:33 2017 From: bipin at xbipin.com (Bipin Patel) Date: Fri, 15 Sep 2017 15:44:33 +0400 Subject: [Freeswitch-users] TLS and SRTP commercial certs go in which file? Message-ID: <09ea1523-534d-b769-38b7-1a729a45241f@xbipin.com> hi, when i setup verto on my server i used commercial certificates with wss.pem containing the following and all that works brilliant: -----BEGIN CERTIFICATE----- -----END CERTIFICATE----- -----BEGIN RSA PRIVATE KEY----- -----END RSA PRIVATE KEY----- -----BEGIN CERTIFICATE----- -----END CERTIFICATE----- -----BEGIN CERTIFICATE----- -----END CERTIFICATE----- now i want to use the same certificate for TLS and SRTP and i was reading the docs and it mentioned to create a agent.pem file with the actual server cert and key but where do i copy the intermediatory and root cert of the CA, which folders do both go in and with what filename? any help is appreciated -- Regards, Bipin ------------------------------------------------------------------------ -------------- next part -------------- An HTML attachment was scrubbed... URL: From krice at freeswitch.org Fri Sep 15 12:23:41 2017 From: krice at freeswitch.org (Ken Rice) Date: Fri, 15 Sep 2017 07:23:41 -0500 Subject: [Freeswitch-users] debian-unstable on Stretch In-Reply-To: References: Message-ID: there are no available packages for stretch at this time. Sent from my iPhone > On Sep 15, 2017, at 05:01, Stanislav Sinyagin wrote: > > I'm trying to install the unstable FreeSWITCH on debian Stretch, and I > get the following error: > > # apt-get update > Hit:1 http://security.debian.org stretch/updates InRelease > Get:3 http://files.freeswitch.org/repo/deb/debian-unstable stretch > InRelease [4,595 B] > Ign:2 http://cdn-fastly.deb.debian.org/debian stretch InRelease > Hit:4 http://cdn-fastly.deb.debian.org/debian stretch Release > Ign:3 http://files.freeswitch.org/repo/deb/debian-unstable stretch InRelease > Fetched 4,595 B in 1s (3,266 B/s) > Reading package lists... Done > W: GPG error: http://files.freeswitch.org/repo/deb/debian-unstable > stretch InRelease: The following signatures were invalid: > 20B06EE621AB150D40F6079FD76EDC7725E010CF > W: The repository > 'http://files.freeswitch.org/repo/deb/debian-unstable stretch > InRelease' is not signed. > N: Data from such a repository can't be authenticated and is therefore > potentially dangerous to use. > N: See apt-secure(8) manpage for repository creation and user > configuration details. > > I imported all public keys that I could find, but it didn't help. > Which key was used for signing this repo? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From samir.doshi at inextrix.com Fri Sep 15 12:27:36 2017 From: samir.doshi at inextrix.com (Samir Doshi) Date: Fri, 15 Sep 2017 17:57:36 +0530 Subject: [Freeswitch-users] debian-unstable on Stretch In-Reply-To: References: Message-ID: https://freeswitch.org/confluence/display/FREESWITCH/Debian+9+Stretch Sent with Mailtrack <#> Best Regards -- Samir Doshi *iNextrix Technologie**s Pvt. Ltd*. http://www.inextrix.com *Disclaimer:* The information contained in this communication is confidential and may be legally privileged. It is intended solely for the use of the individual or entity to whom it is addressed and others authorized to receive it. If you are not the intended recipient you are hereby notified that any disclosure, copying, distribution or taking action in reliance of the contents of this information is strictly prohibited and may be unlawful. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. On Fri, Sep 15, 2017 at 3:31 PM, Stanislav Sinyagin wrote: > I'm trying to install the unstable FreeSWITCH on debian Stretch, and I > get the following error: > > # apt-get update > Hit:1 http://security.debian.org stretch/updates InRelease > Get:3 http://files.freeswitch.org/repo/deb/debian-unstable stretch > InRelease [4,595 B] > Ign:2 http://cdn-fastly.deb.debian.org/debian stretch InRelease > Hit:4 http://cdn-fastly.deb.debian.org/debian stretch Release > Ign:3 http://files.freeswitch.org/repo/deb/debian-unstable stretch > InRelease > Fetched 4,595 B in 1s (3,266 B/s) > Reading package lists... Done > W: GPG error: http://files.freeswitch.org/repo/deb/debian-unstable > stretch InRelease: The following signatures were invalid: > 20B06EE621AB150D40F6079FD76EDC7725E010CF > W: The repository > 'http://files.freeswitch.org/repo/deb/debian-unstable stretch > InRelease' is not signed. > N: Data from such a repository can't be authenticated and is therefore > potentially dangerous to use. > N: See apt-secure(8) manpage for repository creation and user > configuration details. > > I imported all public keys that I could find, but it didn't help. > Which key was used for signing this repo? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From deepikay at iiitd.ac.in Fri Sep 15 13:30:20 2017 From: deepikay at iiitd.ac.in (Deepika Yadav) Date: Fri, 15 Sep 2017 19:00:20 +0530 Subject: [Freeswitch-users] DTMF events not occurring while using VOIP-GSM gateway hardware locally In-Reply-To: References: Message-ID: Hey, the issue got resolved by changing the value of "DTMF Method" field under the "DTMF parameter" from SIGNAL to RFC2833. Regards, Deepika On Fri, Sep 15, 2017 at 4:37 PM, Tihomir Culjaga wrote: > Hi, > > try tcpdump and than export audio to a file so you can play it ( or play > it directly in wireshark ) > maybe the GSMGW is sending InBand DTMF who knows :=) > > also, depending of the GSMGW, used, try to identify what DTMF method is > accepting/sending > > > > On 15 September 2017 at 06:07, Андрей Асташов > wrote: > >> Hi >> >> I also had a similar problem. >> The problem was only on some GSM gateways. And on the incoming call, >> playanddigits recognizes DTMF normally, and the outgoing one does not. >> It did not work out that way. >> In the console, when dtmf_verbose is enabled, pressing the buttons is >> visible. In tcpdump it is seen that it is in RFC that a click comes. >> Moved to bind_digit_action. Works without problems on all types of locks >> and in any directions. >> Here is the discussion and setting of the profile: >> https://freeswitchforum.com/viewtopic.php?f=6&t=864 >> >> 2017-09-14 15:38 GMT+06:00 Deepika Yadav : >> >>> Hi >>> >>> I need to originate calls to Indian mobile numbers from my python ESL >>> script. I used to use service from a company for sending calls from >>> Freeswitch to GSM network for which I set up external SIP profile. >>> >>> But recently I bought my own GOIP hardware, though I am able to make >>> calls but DTMF events are not getting captured on Freeswitch, API like >>> playandGetDigits does not work and there are no logs on Freeswitch console. >>> >>> -- >>> Regards, >>> Deepika >>> https://deepikay.wixsite.com/deepika >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Regards, Deepika https://deepikay.wixsite.com/deepika -------------- next part -------------- An HTML attachment was scrubbed... URL: From olegstolyar at gmail.com Fri Sep 15 15:05:50 2017 From: olegstolyar at gmail.com (Oleg Stolyar) Date: Fri, 15 Sep 2017 08:05:50 -0700 Subject: [Freeswitch-users] Calling uuid_record multiple times Message-ID: Hi guys, is it safe to call uuid_record multiple times for the *same uuid and path* ? Will it just ignore the subsequent calls and continue recording or will it clear the file and restart the recording? -------------- next part -------------- An HTML attachment was scrubbed... URL: From anthony.minessale at gmail.com Fri Sep 15 16:07:08 2017 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 15 Sep 2017 11:07:08 -0500 Subject: [Freeswitch-users] JAVA ISSUE In-Reply-To: References: Message-ID: session = new JavaSession(sessionUuid); is not returning a valid session so then session.answer() will not work because the session is not initialized. On Thu, Sep 14, 2017 at 7:37 AM, Freeswitch user wrote: > Hello Everyone, > > Sorry to disturb you.I'm getting an error while i'm trying to call java > file from dialplan. > > Error:- [ERR] switch_cpp.cpp:683 object is not initalized > > i have compiled mod_java successfully. there was no error during make and > make install. but it's showing error in below java code. > > ------------------------------------------- > > import org.freeswitch.*; > import org.freeswitch.swig.*; > > public class PhoneTest implements FreeswitchScript, DTMFCallback, > HangupHook > { > public PhoneTest() > { > } > > public String onDTMF(Object object, int i, String arg) > { > if (object instanceof String) > freeswitch.console_log("notice", "DTMF: " + (String)object + > " ARG: " + arg + "\n"); > else > freeswitch.console_log("notice", "WOW GOT AN EVENT: " + > object.toString()); > return "true"; > } > > public void onHangup() > { > freeswitch.console_log("notice", "HANGUP!\n"); > } > > public void run(String sessionUuid, String args) > { > freeswitch.console_log("notice", "UUID: " + sessionUuid + " ARGS: > " + args + "\n"); > JavaSession session = null; > try > { > > { > session = new JavaSession(sessionUuid); > session.answer(); // getting Error in This Line > freeswitch.console_log("INFO","\nHIII\n"); > }catch(Exception ee){freeswitch.console_log(" > error",ee.getMessage());} > finally > { > if (session != null) > session.delete(); > } > } > } > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ * ClueCon Weekly Development Call ☎ sip:888 at conference.freeswitch.org ☎ +19193869900 https://www.youtube.com/watch?v=oAxXgyx5jUw https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: From amit.nakum2009 at gmail.com Fri Sep 15 07:22:13 2017 From: amit.nakum2009 at gmail.com (Amit Nakum) Date: Fri, 15 Sep 2017 12:52:13 +0530 Subject: [Freeswitch-users] Fax processing not successful - result (35) Unexpected DCN while waiting for DCS or DIS. In-Reply-To: <789D7240-6385-4354-B2B7-CEDB51EF5AC8@jerris.com> References: <789D7240-6385-4354-B2B7-CEDB51EF5AC8@jerris.com> Message-ID: Dear User, i blindly follow instruction from net. so i don’t know why its necessary to set fax_enable_t38_request=true. If i remove this parameter then it will resolved my issue? On Thu, Sep 14, 2017 at 9:17 PM, Michael Jerris wrote: > is there a reason you are doing fax_enable_t38_request=true? most likely > you should bot be setting that. > > > On Sep 14, 2017, at 6:49 AM, Amit Nakum > wrote: > > > > Dear users, > > > > I am new to freeswitch and i am try to learn fax to email. but i am > getting sometime this error and i am fail to received fax. > > > > I try to set below parameter for fax service. > > > > > > > > i try to search solution for this problem and couldn’t found solution. > > > > Can anyone guide. > > > > Thank > > Amit Nakum > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From freeswitch940 at gmail.com Thu Sep 14 18:15:58 2017 From: freeswitch940 at gmail.com (Freeswitch user) Date: Thu, 14 Sep 2017 23:45:58 +0530 Subject: [Freeswitch-users] DTMF events not occurring while using VOIP-GSM gateway hardware locally In-Reply-To: References: Message-ID: For detect In-band dtmf . You can try below application in your xml dialplan. On Thu, Sep 14, 2017 at 11:12 PM, Deepika Yadav wrote: > It is not ESl but a python script only which is called from a XML > extension. The code is as follows: > > > def initial_greeting(session, phone_number, channel_uuid): > > sangoshthi_greeting = "/usr/local/freeswitch/sounds/ > sangoshthi_ivr_audios/Sangoshthi_greeting.wav" > invalid_digit = "/usr/local/freeswitch/sounds/ > sangoshthi_ivr_audios/Invalid_entry.wav" > > topic = str(session.playAndGetDigits(1, 1, 5, 5000,"", > sangoshthi_greeting, invalid_digit, "[12345*]")) > > ivr_handler(topic, session, phone_number, channel_uuid) > > > def handler(session, args): > > session.setHangupHook(hangup_hook) > session.setInputCallback(input_callback) > > print('session object value is ', str(session)) > > callerid = session.getVariable('caller_id_number') > call_uuid = session.getVariable('uuid') > > phone_number = callerid[len(callerid)-10:] > > initial_greeting(session, phone_number, call_uuid) > > > > > On Thu, Sep 14, 2017 at 10:08 PM, Freeswitch user > wrote: > >> Hello Deepika, >> >> As per my experience playandGetDigits is channel variable so we can use >> it with session only. I don't think it will work with ESL. You can use >> mod_perl for same.. >> >> On Thu, 14 Sep 2017 at 9:10 PM, Deepika Yadav >> wrote: >> >>> Hi >>> >>> I need to originate calls to Indian mobile numbers from my python ESL >>> script. I used to use service from a company for sending calls from >>> Freeswitch to GSM network for which I set up external SIP profile. >>> >>> But recently I bought my own GOIP hardware, though I am able to make >>> calls but DTMF events are not getting captured on Freeswitch, API like >>> playandGetDigits does not work and there are no logs on Freeswitch console. >>> >>> -- >>> Regards, >>> Deepika >>> https://deepikay.wixsite.com/deepika >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Regards, > Deepika > https://deepikay.wixsite.com/deepika > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From freeswitch940 at gmail.com Fri Sep 15 16:29:18 2017 From: freeswitch940 at gmail.com (Freeswitch user) Date: Fri, 15 Sep 2017 16:29:18 +0000 Subject: [Freeswitch-users] JAVA ISSUE In-Reply-To: References: Message-ID: How can I do that I didn't get any example over freeswitch wiki... it would be really for me if you can suggest me or give me a demo script. On Fri, 15 Sep 2017 at 9:38 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > session = new JavaSession(sessionUuid); is not returning a valid session > so then session.answer() will not work because the session is not > initialized. > > > On Thu, Sep 14, 2017 at 7:37 AM, Freeswitch user > wrote: > >> Hello Everyone, >> >> Sorry to disturb you.I'm getting an error while i'm trying to call java >> file from dialplan. >> >> Error:- [ERR] switch_cpp.cpp:683 object is not initalized >> >> i have compiled mod_java successfully. there was no error during make and >> make install. but it's showing error in below java code. >> >> ------------------------------------------- >> >> import org.freeswitch.*; >> import org.freeswitch.swig.*; >> >> public class PhoneTest implements FreeswitchScript, DTMFCallback, >> HangupHook >> { >> public PhoneTest() >> { >> } >> >> public String onDTMF(Object object, int i, String arg) >> { >> if (object instanceof String) >> freeswitch.console_log("notice", "DTMF: " + (String)object + >> " ARG: " + arg + "\n"); >> else >> freeswitch.console_log("notice", "WOW GOT AN EVENT: " + >> object.toString()); >> return "true"; >> } >> >> public void onHangup() >> { >> freeswitch.console_log("notice", "HANGUP!\n"); >> } >> >> public void run(String sessionUuid, String args) >> { >> freeswitch.console_log("notice", "UUID: " + sessionUuid + " ARGS: >> " + args + "\n"); >> JavaSession session = null; >> try >> { >> >> { >> session = new JavaSession(sessionUuid); >> session.answer(); // getting Error in This Line >> freeswitch.console_log("INFO","\nHIII\n"); >> }catch(Exception >> ee){freeswitch.console_log("error",ee.getMessage());} >> finally >> { >> if (session != null) >> session.delete(); >> } >> } >> } >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ > > ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ > http://twitter.com/FreeSWITCH > ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ > * > > ClueCon Weekly Development Call > ☎ sip:888 at conference.freeswitch.org ☎ +19193869900 > > https://www.youtube.com/watch?v=oAxXgyx5jUw > https://www.youtube.com/watch?v=9XXgW34t40s > https://www.youtube.com/watch?v=NLaDpGQuZDA > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From joelists at tm.net.uk Fri Sep 15 20:08:47 2017 From: joelists at tm.net.uk (Joseph Waite) Date: Fri, 15 Sep 2017 21:08:47 +0100 Subject: [Freeswitch-users] cdr_csv.conf.xml Message-ID: Hi Guys I have checked online but can’t seem to find anything. I am wondering if it is possible to set a different template for a leg and bleg cdr’s? If so how would I go about this? Regards From jurijs.ivolga at gmail.com Fri Sep 15 20:33:17 2017 From: jurijs.ivolga at gmail.com (Jurijs Ivolga) Date: Fri, 15 Sep 2017 20:33:17 +0000 Subject: [Freeswitch-users] cdr_csv.conf.xml In-Reply-To: References: Message-ID: Hi, Is it what you looking for: https://wiki.freeswitch.org/wiki/Mod_cdr_csv On Fri, 15 Sep 2017 at 23:09, Joseph Waite wrote: > Hi Guys > > I have checked online but can’t seem to find anything. > > I am wondering if it is possible to set a different template for a leg and > bleg cdr’s? > If so how would I go about this? > > Regards > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Jurijs -------------- next part -------------- An HTML attachment was scrubbed... URL: From joelists at tm.net.uk Fri Sep 15 20:42:13 2017 From: joelists at tm.net.uk (Joseph Waite) Date: Fri, 15 Sep 2017 21:42:13 +0100 Subject: [Freeswitch-users] cdr_csv.conf.xml In-Reply-To: References: Message-ID: <86EC3187-1974-46A2-8899-C34AECE20BFF@tm.net.uk> Hi Unfortunalty not. This simply allows you to tell it only a or b leg cars to be created. I want cdr’s for both legs, however I want a different format for a leg cdr to b leg cdr. Regards > On 15 Sep 2017, at 21:33, Jurijs Ivolga wrote: > > Hi, > > Is it what you looking for: > > > > > https://wiki.freeswitch.org/wiki/Mod_cdr_csv > > > On Fri, 15 Sep 2017 at 23:09, Joseph Waite > wrote: > Hi Guys > > I have checked online but can’t seem to find anything. > > I am wondering if it is possible to set a different template for a leg and bleg cdr’s? > If so how would I go about this? > > Regards > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- > Jurijs > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Fri Sep 15 20:50:29 2017 From: mike at jerris.com (Michael Jerris) Date: Fri, 15 Sep 2017 16:50:29 -0400 Subject: [Freeswitch-users] cdr_csv.conf.xml In-Reply-To: References: Message-ID: <5C403F57-70E6-47D8-B2C6-439384A670F0@jerris.com> https://freeswitch.org/confluence/display/FREESWITCH/mod_cdr_csv#mod_cdr_csv-accountcode > On Sep 15, 2017, at 4:08 PM, Joseph Waite wrote: > > Hi Guys > > I have checked online but can’t seem to find anything. > > I am wondering if it is possible to set a different template for a leg and bleg cdr’s? > If so how would I go about this? -------------- next part -------------- An HTML attachment was scrubbed... URL: From joelists at tm.net.uk Fri Sep 15 21:01:30 2017 From: joelists at tm.net.uk (Joseph Waite) Date: Fri, 15 Sep 2017 22:01:30 +0100 Subject: [Freeswitch-users] cdr_csv.conf.xml In-Reply-To: <5C403F57-70E6-47D8-B2C6-439384A670F0@jerris.com> References: <5C403F57-70E6-47D8-B2C6-439384A670F0@jerris.com> Message-ID: I need a template for each leg. Would naming the templates aleg & bleg do what I need like the example of naming the template as an account code? > On 15 Sep 2017, at 21:50, Michael Jerris wrote: > > https://freeswitch.org/confluence/display/FREESWITCH/mod_cdr_csv#mod_cdr_csv-accountcode > >> On Sep 15, 2017, at 4:08 PM, Joseph Waite > wrote: >> >> Hi Guys >> >> I have checked online but can’t seem to find anything. >> >> I am wondering if it is possible to set a different template for a leg and bleg cdr’s? >> If so how would I go about this? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From jungleboogie0 at gmail.com Fri Sep 15 21:08:46 2017 From: jungleboogie0 at gmail.com (jungle Boogie) Date: Fri, 15 Sep 2017 14:08:46 -0700 Subject: [Freeswitch-users] cdr_csv.conf.xml In-Reply-To: References: <5C403F57-70E6-47D8-B2C6-439384A670F0@jerris.com> Message-ID: On 15 September 2017 at 14:01, Joseph Waite wrote: > I need a template for each leg. Would naming the templates aleg & bleg do > what I need like the example of naming the template as an account code? > So as an example: aleg will have: time, date, callerID bleg will have: date, time, holdtime This is what you want? From joelists at tm.net.uk Fri Sep 15 21:14:30 2017 From: joelists at tm.net.uk (Joseph Waite) Date: Fri, 15 Sep 2017 22:14:30 +0100 Subject: [Freeswitch-users] cdr_csv.conf.xml In-Reply-To: References: <5C403F57-70E6-47D8-B2C6-439384A670F0@jerris.com> Message-ID: <34B0344E-9080-468E-B972-68E0AC41879F@tm.net.uk> > On 15 Sep 2017, at 22:08, jungle Boogie wrote: > > On 15 September 2017 at 14:01, Joseph Waite wrote: >> I need a template for each leg. Would naming the templates aleg & bleg do >> what I need like the example of naming the template as an account code? >> > > So as an example: > aleg will have: time, date, callerID > bleg will have: date, time, holdtime > > This is what you want? > Yes basically, different values but thats the idea! From mike at jerris.com Fri Sep 15 21:23:13 2017 From: mike at jerris.com (Michael Jerris) Date: Fri, 15 Sep 2017 17:23:13 -0400 Subject: [Freeswitch-users] cdr_csv.conf.xml In-Reply-To: References: <5C403F57-70E6-47D8-B2C6-439384A670F0@jerris.com> Message-ID: <5F0BB8DB-9E62-442C-B92C-206A0737D6FC@jerris.com> The link i posted should answer your question. Please read through it. > On Sep 15, 2017, at 5:01 PM, Joseph Waite wrote: > > I need a template for each leg. Would naming the templates aleg & bleg do what I need like the example of naming the template as an account code? > >> On 15 Sep 2017, at 21:50, Michael Jerris > wrote: >> >> https://freeswitch.org/confluence/display/FREESWITCH/mod_cdr_csv#mod_cdr_csv-accountcode >> >>> On Sep 15, 2017, at 4:08 PM, Joseph Waite > wrote: >>> >>> Hi Guys >>> >>> I have checked online but can’t seem to find anything. >>> >>> I am wondering if it is possible to set a different template for a leg and bleg cdr’s? >>> If so how would I go about this? >> -------------- next part -------------- An HTML attachment was scrubbed... URL: From joelists at tm.net.uk Fri Sep 15 21:30:11 2017 From: joelists at tm.net.uk (Joseph Waite) Date: Fri, 15 Sep 2017 22:30:11 +0100 Subject: [Freeswitch-users] cdr_csv.conf.xml In-Reply-To: <5F0BB8DB-9E62-442C-B92C-206A0737D6FC@jerris.com> References: <5C403F57-70E6-47D8-B2C6-439384A670F0@jerris.com> <5F0BB8DB-9E62-442C-B92C-206A0737D6FC@jerris.com> Message-ID: <541578F1-C350-4AD3-8B5C-04AB6FA2DCBE@tm.net.uk> I have read through the link sent, but it only mentions an account code. We don’t use account codes. > On 15 Sep 2017, at 22:23, Michael Jerris wrote: > > The link i posted should answer your question. Please read through it. > >> On Sep 15, 2017, at 5:01 PM, Joseph Waite > wrote: >> >> I need a template for each leg. Would naming the templates aleg & bleg do what I need like the example of naming the template as an account code? >> >>> On 15 Sep 2017, at 21:50, Michael Jerris > wrote: >>> >>> https://freeswitch.org/confluence/display/FREESWITCH/mod_cdr_csv#mod_cdr_csv-accountcode >>> >>>> On Sep 15, 2017, at 4:08 PM, Joseph Waite > wrote: >>>> >>>> Hi Guys >>>> >>>> I have checked online but can’t seem to find anything. >>>> >>>> I am wondering if it is possible to set a different template for a leg and bleg cdr’s? >>>> If so how would I go about this? >>> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Fri Sep 15 21:43:14 2017 From: mike at jerris.com (Michael Jerris) Date: Fri, 15 Sep 2017 17:43:14 -0400 Subject: [Freeswitch-users] cdr_csv.conf.xml In-Reply-To: <541578F1-C350-4AD3-8B5C-04AB6FA2DCBE@tm.net.uk> References: <5C403F57-70E6-47D8-B2C6-439384A670F0@jerris.com> <5F0BB8DB-9E62-442C-B92C-206A0737D6FC@jerris.com> <541578F1-C350-4AD3-8B5C-04AB6FA2DCBE@tm.net.uk> Message-ID: if you want to set different templates, thats how you do it. > On Sep 15, 2017, at 5:30 PM, Joseph Waite wrote: > > I have read through the link sent, but it only mentions an account code. We don’t use account codes. >> On 15 Sep 2017, at 22:23, Michael Jerris > wrote: >> >> The link i posted should answer your question. Please read through it. >> >>> On Sep 15, 2017, at 5:01 PM, Joseph Waite > wrote: >>> >>> I need a template for each leg. Would naming the templates aleg & bleg do what I need like the example of naming the template as an account code? >>> >>>> On 15 Sep 2017, at 21:50, Michael Jerris > wrote: >>>> >>>> https://freeswitch.org/confluence/display/FREESWITCH/mod_cdr_csv#mod_cdr_csv-accountcode >>>> >>>>> On Sep 15, 2017, at 4:08 PM, Joseph Waite > wrote: >>>>> >>>>> Hi Guys >>>>> >>>>> I have checked online but can’t seem to find anything. >>>>> >>>>> I am wondering if it is possible to set a different template for a leg and bleg cdr’s? >>>>> If so how would I go about this? >>>> -------------- next part -------------- An HTML attachment was scrubbed... URL: From bipin at xbipin.com Sat Sep 16 05:48:37 2017 From: bipin at xbipin.com (Bipin Patel) Date: Sat, 16 Sep 2017 09:48:37 +0400 Subject: [Freeswitch-users] TLS and SRTP commercial certs go in which file? In-Reply-To: <09ea1523-534d-b769-38b7-1a729a45241f@xbipin.com> References: <09ea1523-534d-b769-38b7-1a729a45241f@xbipin.com> Message-ID: <02addee6-3a74-b95b-b2ab-e90851e6f3e2@xbipin.com> hi, no one? Regards, Bipin ------------------------------------------------------------------------ -------- Original Message -------- Subject: [Freeswitch-users] TLS and SRTP commercial certs go in which file? From: Bipin Patel To: FreeSWITCH Users Help Date: 9/15/2017, 3:44:33 PM > hi, > > when i setup verto on my server i used commercial certificates with > wss.pem containing the following and all that works brilliant: > -----BEGIN CERTIFICATE----- > > -----END CERTIFICATE----- > -----BEGIN RSA PRIVATE KEY----- > > -----END RSA PRIVATE KEY----- > -----BEGIN CERTIFICATE----- > > -----END CERTIFICATE----- > -----BEGIN CERTIFICATE----- > > -----END CERTIFICATE----- > > now i want to use the same certificate for TLS and SRTP and i was > reading the docs and it mentioned to create a agent.pem file with the > actual server cert and key but where do i copy the intermediatory and > root cert of the CA, which folders do both go in and with what filename? > > any help is appreciated > > > -- > Regards, > Bipin > > > ------------------------------------------------------------------------ > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From vma at vallimamod.org Sat Sep 16 13:24:47 2017 From: vma at vallimamod.org (Vallimamod Abdullah) Date: Sat, 16 Sep 2017 15:24:47 +0200 Subject: [Freeswitch-users] cdr_csv.conf.xml In-Reply-To: References: <5C403F57-70E6-47D8-B2C6-439384A670F0@jerris.com> <5F0BB8DB-9E62-442C-B92C-206A0737D6FC@jerris.com> <541578F1-C350-4AD3-8B5C-04AB6FA2DCBE@tm.net.uk> Message-ID: Hi, From the wiki as Michael said: > NOTE: if the value of the accountcode variable matches the name of a template then that template will be used. This is valuable for having a specific template be used on a per-call basis. Best Regards, -- Vallimamod Abdullah SIP Solutions vma at sipsolutions.fr . > On 15 Sep 2017, at 23:43, Michael Jerris wrote: > > if you want to set different templates, thats how you do it. > >> On Sep 15, 2017, at 5:30 PM, Joseph Waite wrote: >> >> I have read through the link sent, but it only mentions an account code. We don’t use account codes. >>> On 15 Sep 2017, at 22:23, Michael Jerris wrote: >>> >>> The link i posted should answer your question. Please read through it. >>> >>>> On Sep 15, 2017, at 5:01 PM, Joseph Waite wrote: >>>> >>>> I need a template for each leg. Would naming the templates aleg & bleg do what I need like the example of naming the template as an account code? >>>> >>>>> On 15 Sep 2017, at 21:50, Michael Jerris wrote: >>>>> >>>>> https://freeswitch.org/confluence/display/FREESWITCH/mod_cdr_csv#mod_cdr_csv-accountcode >>>>> >>>>>> On Sep 15, 2017, at 4:08 PM, Joseph Waite wrote: >>>>>> >>>>>> Hi Guys >>>>>> >>>>>> I have checked online but can’t seem to find anything. >>>>>> >>>>>> I am wondering if it is possible to set a different template for a leg and bleg cdr’s? >>>>>> If so how would I go about this? From ynasida at gmail.com Sat Sep 16 15:46:31 2017 From: ynasida at gmail.com (Yuriy Nasida) Date: Sat, 16 Sep 2017 18:46:31 +0300 Subject: [Freeswitch-users] vm_cc weird behavior In-Reply-To: References: Message-ID: Any advice? 14 Сен 2017 г. 16:00 пользователь "Yuriy Nasida" написал: > Hi, > > I noted that FS sends a lot of emails to my email box instead of only one > in case I use vm_cc. > > It's very simple configuration. I just set vm_cc before mod_voicemail is > called (according wiki page) and leave voicemail message. > For main emails box I recieve only one message as expected. > But I have ~60 exactly same messages at email box from vm_cc. > > In console I also noted that FS really sends emails many times without > obvios reason. > > mod_voicemail.c:2832 Deliver VM to yuriytest2 at mydomain.com > mod_voicemail.c:1925 Update MWI: Processing for yuriytest2 at mydomain.com > in inbox > mod_voicemail.c:1950 Update MWI: Messages Waiting yes > mod_voicemail.c:1951 Update MWI: Update Reason NEW > mod_voicemail.c:1952 Update MWI: Message Account yuriytest2 at mydomain.com > mod_voicemail.c:1953 Update MWI: Voice Message 69/0 > switch_utils.c:1180 Emailed file [/tmp/mail.1505386894e5a8] to [ > myemailbox at gmail.com] > mod_voicemail.c:3042 Sending message to myemailbox at gmail.com > > Next FS does this again and tries to Deliver VM to yuriytest2 at mydomain.com > . > > Please advice. > -------------- next part -------------- An HTML attachment was scrubbed... URL: From ssinyagin at gmail.com Sat Sep 16 16:53:39 2017 From: ssinyagin at gmail.com (Stanislav Sinyagin) Date: Sat, 16 Sep 2017 18:53:39 +0200 Subject: [Freeswitch-users] mod_expr random not thread safe? Message-ID: I'm building a test for voice quality, and I need to randomize the silence preamble. At the same time, other sessions are recorded into a tmpfs file. I noticed that when I added this, the audio recordings started to have gaps of dropped audio 40ms long. Are the randomize() and random() thread-safe at all? It seems like the whole FreeSWITCH process freezes when one of those functions gets executed. From ssinyagin at gmail.com Sat Sep 16 17:48:55 2017 From: ssinyagin at gmail.com (Stanislav Sinyagin) Date: Sat, 16 Sep 2017 19:48:55 +0200 Subject: [Freeswitch-users] mod_expr random not thread safe? In-Reply-To: References: Message-ID: a ho, mod_expr was not loaded, and that caused the drops when I called expr() On Sat, Sep 16, 2017 at 6:53 PM, Stanislav Sinyagin wrote: > I'm building a test for voice quality, and I need to randomize the > silence preamble. At the same time, other sessions are recorded into a > tmpfs file. > > data="rand_val=${expr(randomize(&x);ceil(random(0,100,&x)))}"/> > > I noticed that when I added this, the audio recordings started to have > gaps of dropped audio 40ms long. > > Are the randomize() and random() thread-safe at all? It seems like the > whole FreeSWITCH process freezes when one of those functions gets > executed. From freeswitch940 at gmail.com Sat Sep 16 08:10:03 2017 From: freeswitch940 at gmail.com (Freeswitch user) Date: Sat, 16 Sep 2017 08:10:03 +0000 Subject: [Freeswitch-users] JAVA ISSUE In-Reply-To: References: Message-ID: Hello users, Anyone ? On Fri, 15 Sep 2017 at 9:57 PM, Freeswitch user wrote: > How can I do that I didn't get any example over freeswitch wiki... it > would be really for me if you can suggest me or give me a demo script. > > On Fri, 15 Sep 2017 at 9:38 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> session = new JavaSession(sessionUuid); is not returning a valid session >> so then session.answer() will not work because the session is not >> initialized. >> >> >> On Thu, Sep 14, 2017 at 7:37 AM, Freeswitch user > > wrote: >> >>> Hello Everyone, >>> >>> Sorry to disturb you.I'm getting an error while i'm trying to call java >>> file from dialplan. >>> >>> Error:- [ERR] switch_cpp.cpp:683 object is not initalized >>> >>> i have compiled mod_java successfully. there was no error during make >>> and make install. but it's showing error in below java code. >>> >>> ------------------------------------------- >>> >>> import org.freeswitch.*; >>> import org.freeswitch.swig.*; >>> >>> public class PhoneTest implements FreeswitchScript, DTMFCallback, >>> HangupHook >>> { >>> public PhoneTest() >>> { >>> } >>> >>> public String onDTMF(Object object, int i, String arg) >>> { >>> if (object instanceof String) >>> freeswitch.console_log("notice", "DTMF: " + (String)object + >>> " ARG: " + arg + "\n"); >>> else >>> freeswitch.console_log("notice", "WOW GOT AN EVENT: " + >>> object.toString()); >>> return "true"; >>> } >>> >>> public void onHangup() >>> { >>> freeswitch.console_log("notice", "HANGUP!\n"); >>> } >>> >>> public void run(String sessionUuid, String args) >>> { >>> freeswitch.console_log("notice", "UUID: " + sessionUuid + " >>> ARGS: " + args + "\n"); >>> JavaSession session = null; >>> try >>> { >>> >>> { >>> session = new JavaSession(sessionUuid); >>> session.answer(); // getting Error in This Line >>> freeswitch.console_log("INFO","\nHIII\n"); >>> }catch(Exception >>> ee){freeswitch.console_log("error",ee.getMessage());} >>> finally >>> { >>> if (session != null) >>> session.delete(); >>> } >>> } >>> } >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ >> >> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >> http://twitter.com/FreeSWITCH >> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >> * >> >> ClueCon Weekly Development Call >> ☎ sip:888 at conference.freeswitch.org ☎ +19193869900 >> >> https://www.youtube.com/watch?v=oAxXgyx5jUw >> https://www.youtube.com/watch?v=9XXgW34t40s >> https://www.youtube.com/watch?v=NLaDpGQuZDA >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From shaun.stokes at itec-support.co.uk Sat Sep 16 09:03:34 2017 From: shaun.stokes at itec-support.co.uk (Shaun Stokes) Date: Sat, 16 Sep 2017 09:03:34 +0000 Subject: [Freeswitch-users] TLS and SRTP commercial certs go in which file? In-Reply-To: <02addee6-3a74-b95b-b2ab-e90851e6f3e2@xbipin.com> References: <09ea1523-534d-b769-38b7-1a729a45241f@xbipin.com> <02addee6-3a74-b95b-b2ab-e90851e6f3e2@xbipin.com> Message-ID: <6FD2F8B5BB72834E9939AEDF9FB802A901E869C97F@mbx-01.sysconfig.co.uk> Hi Bipin, We've found that these are the certs which FreeSWITCH will use, the pem file will need the public and private key (same as your wss cert). You'll also need to make sure the user for FreeSWITCH has read permission to the certs. agent.pem dtls-srtp.pem tls.pem wss.pem FreeSWITCH doesn't seem to need the intermediary and root cert of the CA. Here are some of the TLS parameters you might also want on your SIP profile. Name: tls Value: true Name: tls-bind-params Value: transport=tls Name: tls-cert-dir Value: "Your Cert Directory Path" Name: tls-sip-port Value: 5061 Name: tls-verify-date Value: true Name: tls-verify-depth Value: 2 Name: tls-verify-policy Value: all|subjects_all Name: tls-version Value: tlsv1.2 Shaun From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bipin Patel Sent: 16 September 2017 06:49 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] TLS and SRTP commercial certs go in which file? hi, no one? Regards, Bipin ________________________________ -------- Original Message -------- Subject: [Freeswitch-users] TLS and SRTP commercial certs go in which file? From: Bipin Patel To: FreeSWITCH Users Help Date: 9/15/2017, 3:44:33 PM hi, when i setup verto on my server i used commercial certificates with wss.pem containing the following and all that works brilliant: -----BEGIN CERTIFICATE----- -----END CERTIFICATE----- -----BEGIN RSA PRIVATE KEY----- -----END RSA PRIVATE KEY----- -----BEGIN CERTIFICATE----- -----END CERTIFICATE----- -----BEGIN CERTIFICATE----- -----END CERTIFICATE----- now i want to use the same certificate for TLS and SRTP and i was reading the docs and it mentioned to create a agent.pem file with the actual server cert and key but where do i copy the intermediatory and root cert of the CA, which folders do both go in and with what filename? any help is appreciated -- Regards, Bipin ________________________________ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ [http://www.itec-support.co.uk/wp-content/uploads/2016/07/email_logo.jpg] Shaun Stokes - Infrastructure Analyst T : 01453 700713 E : shaun.stokes at itec-support.co.uk W : www.itec-support.co.uk Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road, Stroud, Gloucestershire GL5 3QF Company No. 06908001 CONFIDENTIALITY NOTICE This communication and the information it contains are intended for the person or organisation to which it is addressed. Its contents are confidential and may be protected in law. Unauthorised use, copying or disclosure of any of it may be unlawful. If you are not the intended recipient, please contact us immediately. The contents of any attachments in this e-mail may contain software viruses, which could damage your own computer system. While ITEC Support has taken every reasonable precaution to minimise this risk, we cannot accept liability for any damage which you sustain as a result of software viruses. You should carry out your own virus checking procedure before opening any attachment. ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: From shaun.stokes at itec-support.co.uk Sat Sep 16 09:29:59 2017 From: shaun.stokes at itec-support.co.uk (Shaun Stokes) Date: Sat, 16 Sep 2017 09:29:59 +0000 Subject: [Freeswitch-users] TLS and SRTP commercial certs go in which file? In-Reply-To: <6FD2F8B5BB72834E9939AEDF9FB802A901E869C97A@mbx-01.sysconfig.co.uk> References: <09ea1523-534d-b769-38b7-1a729a45241f@xbipin.com> <02addee6-3a74-b95b-b2ab-e90851e6f3e2@xbipin.com> <6FD2F8B5BB72834E9939AEDF9FB802A901E869C97A@mbx-01.sysconfig.co.uk> Message-ID: <6FD2F8B5BB72834E9939AEDF9FB802A901E869CA54@mbx-01.sysconfig.co.uk> Hi Bipin, We've found that these are the certs which FreeSWITCH will use, the pem file will need the public and private key (same as your wss cert). You'll also need to make sure the user for FreeSWITCH has read permission to the certs. agent.pem dtls-srtp.pem tls.pem wss.pem FreeSWITCH doesn't seem to need the intermediary and root cert of the CA. Here are some of the TLS parameters you might also want on your SIP profile. Name: tls Value: true Name: tls-bind-params Value: transport=tls Name: tls-cert-dir Value: "Your Cert Directory Path" Name: tls-sip-port Value: 5061 Name: tls-verify-date Value: true Name: tls-verify-depth Value: 2 Name: tls-verify-policy Value: all|subjects_all Name: tls-version Value: tlsv1.2 Shaun From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bipin Patel Sent: 16 September 2017 06:49 To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] TLS and SRTP commercial certs go in which file? hi, no one? Regards, Bipin ________________________________ -------- Original Message -------- Subject: [Freeswitch-users] TLS and SRTP commercial certs go in which file? From: Bipin Patel To: FreeSWITCH Users Help Date: 9/15/2017, 3:44:33 PM hi, when i setup verto on my server i used commercial certificates with wss.pem containing the following and all that works brilliant: -----BEGIN CERTIFICATE----- -----END CERTIFICATE----- -----BEGIN RSA PRIVATE KEY----- -----END RSA PRIVATE KEY----- -----BEGIN CERTIFICATE----- -----END CERTIFICATE----- -----BEGIN CERTIFICATE----- -----END CERTIFICATE----- now i want to use the same certificate for TLS and SRTP and i was reading the docs and it mentioned to create a agent.pem file with the actual server cert and key but where do i copy the intermediatory and root cert of the CA, which folders do both go in and with what filename? any help is appreciated -- Regards, Bipin ________________________________ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ [http://www.itec-support.co.uk/wp-content/uploads/2016/07/email_logo.jpg] Shaun Stokes - Infrastructure Analyst T : 01453 700713 E : shaun.stokes at itec-support.co.uk W : www.itec-support.co.uk Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road, Stroud, Gloucestershire GL5 3QF Company No. 06908001 CONFIDENTIALITY NOTICE This communication and the information it contains are intended for the person or organisation to which it is addressed. Its contents are confidential and may be protected in law. Unauthorised use, copying or disclosure of any of it may be unlawful. If you are not the intended recipient, please contact us immediately. The contents of any attachments in this e-mail may contain software viruses, which could damage your own computer system. While ITEC Support has taken every reasonable precaution to minimise this risk, we cannot accept liability for any damage which you sustain as a result of software viruses. You should carry out your own virus checking procedure before opening any attachment. ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: From bipin at xbipin.com Sun Sep 17 04:43:17 2017 From: bipin at xbipin.com (Bipin Patel) Date: Sun, 17 Sep 2017 08:43:17 +0400 Subject: [Freeswitch-users] TLS and SRTP commercial certs go in which file? In-Reply-To: <6FD2F8B5BB72834E9939AEDF9FB802A901E869CA54@mbx-01.sysconfig.co.uk> References: <09ea1523-534d-b769-38b7-1a729a45241f@xbipin.com> <02addee6-3a74-b95b-b2ab-e90851e6f3e2@xbipin.com> <6FD2F8B5BB72834E9939AEDF9FB802A901E869C97A@mbx-01.sysconfig.co.uk> <6FD2F8B5BB72834E9939AEDF9FB802A901E869CA54@mbx-01.sysconfig.co.uk> Message-ID: <15e8e25d288.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com> Hi, Thanks for that info so if I understood it right the agent.pem file will have my cert and key inside it only and no intermediary or root ca cert is required at all, but what goes in the tls.pem file? On September 17, 2017 5:45:28 AM Shaun Stokes wrote: > Hi Bipin, > > We've found that these are the certs which FreeSWITCH will use, the pem > file will need the public and private key (same as your wss cert). You'll > also need to make sure the user for FreeSWITCH has read permission to the > certs. > agent.pem > dtls-srtp.pem > tls.pem > wss.pem > > FreeSWITCH doesn't seem to need the intermediary and root cert of the CA. > > Here are some of the TLS parameters you might also want on your SIP profile. > > Name: tls > Value: true > > Name: tls-bind-params > Value: transport=tls > > Name: tls-cert-dir > Value: "Your Cert Directory Path" > > Name: tls-sip-port > Value: 5061 > > Name: tls-verify-date > Value: true > > Name: tls-verify-depth > Value: 2 > > Name: tls-verify-policy > Value: all|subjects_all > > Name: tls-version > Value: tlsv1.2 > > > Shaun > > From: FreeSWITCH-users > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bipin Patel > Sent: 16 September 2017 06:49 > To: FreeSWITCH Users Help > > > Subject: Re: [Freeswitch-users] TLS and SRTP commercial certs go in which file? > > hi, > > no one? > Regards, > Bipin > ________________________________ > -------- Original Message -------- > Subject: [Freeswitch-users] TLS and SRTP commercial certs go in which file? > From: Bipin Patel > To: FreeSWITCH Users Help > > Date: 9/15/2017, 3:44:33 PM > hi, > > when i setup verto on my server i used commercial certificates with wss.pem > containing the following and all that works brilliant: > -----BEGIN CERTIFICATE----- > > -----END CERTIFICATE----- > -----BEGIN RSA PRIVATE KEY----- > > -----END RSA PRIVATE KEY----- > -----BEGIN CERTIFICATE----- > > -----END CERTIFICATE----- > -----BEGIN CERTIFICATE----- > > -----END CERTIFICATE----- > > now i want to use the same certificate for TLS and SRTP and i was reading > the docs and it mentioned to create a agent.pem file with the actual server > cert and key but where do i copy the intermediatory and root cert of the > CA, which folders do both go in and with what filename? > > any help is appreciated > -- > Regards, > Bipin > ________________________________ > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > ______________________________________________________________________ > This message has been checked for all known viruses by MessageLabs Virus > Scanning Service. > ______________________________________________________________________ > [http://www.itec-support.co.uk/wp-content/uploads/2016/07/email_logo.jpg] > Shaun Stokes - Infrastructure Analyst > > T : 01453 700713 > E : shaun.stokes at itec-support.co.uk > W : www.itec-support.co.uk > > Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road, > Stroud, Gloucestershire GL5 3QF > Company No. 06908001 > > CONFIDENTIALITY NOTICE > This communication and the information it contains are intended for the > person or organisation to which it is addressed. Its contents are > confidential and may be protected in law. Unauthorised use, copying or > disclosure of any of it may be unlawful. If you are not the intended > recipient, please contact us immediately. > The contents of any attachments in this e-mail may contain software > viruses, which could damage your own computer system. While ITEC Support > has taken every reasonable precaution to minimise this risk, we cannot > accept liability for any damage which you sustain as a result of software > viruses. You should carry out your own virus checking procedure before > opening any attachment. > > ______________________________________________________________________ > This message has been checked for all known viruses by MessageLabs Virus > Scanning Service. > ______________________________________________________________________ > > > ---------- > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From deepikay at iiitd.ac.in Sun Sep 17 09:27:27 2017 From: deepikay at iiitd.ac.in (Deepika Yadav) Date: Sun, 17 Sep 2017 14:57:27 +0530 Subject: [Freeswitch-users] How to record User Voice and listen for the DTMF simultaneously Message-ID: Hi, I want to give users an option to record their voice by presssing some digit in the IVR menu through my python script. I use the code as follows: session.recordFile(filename, 120, 500, 3) However, this code will block for 2 minutes and then proceed. In case a user has little to say and after 1 minute wants to stop the recording, how I can do that. Is there any function similar to "session.playAndGetDigits" to also hear for DTMFs at the same time? -- Regards, Deepika https://deepikay.wixsite.com/deepika -------------- next part -------------- An HTML attachment was scrubbed... URL: From deepikay at iiitd.ac.in Sun Sep 17 11:51:22 2017 From: deepikay at iiitd.ac.in (Deepika Yadav) Date: Sun, 17 Sep 2017 17:21:22 +0530 Subject: [Freeswitch-users] Freeswitch Crashes on Call Hangup of IVR Session using mod_python script Message-ID: Hi, I am facing an issue, in my python script whenever user hangs up Freeswitch crashes with following logs: 7f290a7ee000-7f290a7ef000 r--p 0000d000 08:02 129175424 /usr/lib/x86_64-linux-gnu/liblber-2.4.so.2.8.3 7f290a7ef000-7f290a7f0000 rw-p 0000e000 08:02 129175424 /usr/lib/x86_64-linux-gnu/liblber-2.4.so.2.8.3 7f290a7f0000-7f290a834000 r-xp 00000000 08:02 129173227 /usr/lib/x86_64-linux-gnu/libgssapi_krb5.so.2.2 7f290a834000-7f290aa34000 ---p 00044000 08:02 129173227 /usr/lib/x86_64-linux-gnu/libgssapi_krb5.so.2.2 7f290aa34000-7f290aa35000 r--p 00044000 08:02 129173227 /usr/lib/x86_64-linux-gnu/libgssapi_krb5.so.2.2 7f290aa35000-7f290aa37000 rw-p 00045000 08:02 129173227 /usr/lib/x86_64-linux-gnu/libgssapi_krb5.so.2.2 7f290aa37000-7f290aa50000 r-xp 00000000 08:02 129172855 /usr/lib/x86_64-linux-gnu/librtmp.so.0 I am using version - 1.6.15~64bit My Python Script looks something like this; def hangup_hook(session, what, args=' '): freeswitch.consoleLog("INFO", "hangup hook for '%s'\n" % what) def handler(session, args): session.setHangupHook(hangup_hook) session.setInputCallback(input_callback) print('session object value is ', str(session)) callerid = session.getVariable('caller_id_number') call_uuid = session.getVariable('uuid') phone_number = callerid[len(callerid)-10:] initial_greeting(session, phone_number, call_uuid) -- Regards, Deepika https://deepikay.wixsite.com/deepika -------------- next part -------------- An HTML attachment was scrubbed... URL: From deepikay at iiitd.ac.in Mon Sep 18 06:08:06 2017 From: deepikay at iiitd.ac.in (Deepika Yadav) Date: Mon, 18 Sep 2017 11:38:06 +0530 Subject: [Freeswitch-users] Hanging Call- no CALL DESTROY events even when users ended the call Message-ID: Hi, I am using mod_python script for IVR functionality, and facing an issue. Even after users end the call, no CALL DESTROY events are captured by the Freeswitch and after a while Freeswitch itself crashes with following logs: *7fd202913000-7fd202914000 rw-p 0000a000 08:02 129173231 /usr/lib/x86_64-linux-gnu/libkrb5support.so.0.1* *7fd202914000-7fd202917000 r-xp 00000000 08:02 43057313 /lib/x86_64-linux-gnu/libcom_err.so.2.1* *7fd202917000-7fd202b16000 ---p 00003000 08:02 43057313 /lib/x86_64-linux-gnu/libcom_err.so.2.1* *7fd202b16000-7fd202b17000 r--p 00002000 08:02 43057313 /lib/x86_64-linux-gnu/libcom_err.so.2.1* *7fd202b17000-7fd202b18000 rw-p 00003000 08:02 43057313 /lib/x86_64-linux-gnu/libcom_err.so.2.1* *7fd202b18000-7fd202b44000 r-xp 00000000 08:02 129177695 /usr/lib/x86_64-linux-gnu/libk5crypto.so.3.1* *7fd202b44000-7fd202d43000 ---p 0002c000 08:02 129177695 /usr/lib/x86_64-linux-gnu/libk5crypto.so.3.1* *7fd202d43000-7fd202d45000 r--p 0002b000 08:02 129177695 /usr/lib/x86_64-linux-gnu/libk5crypto.so.3.1* *7fd202d45000-7fd202d46000 rw-p 0002d000 08:02 129177695 /usr/lib/x86_64-linux-gnu/libk5crypto.so.3.1* *7fd202d46000-7fd202d47000 rw-p 00000000 00:00 0* *7fd202d47000-7fd202e03000 r-xp 00000000 08:02 129173229 /usr/lib/x86_64-linux-gnu/libkrb5.so.3.3* *7fd202e03000-7fd203003000 ---p 000bc000 08:02 129173229 /usr/lib/x86_64-linux-gnu/libkrb5.so.3.3* *7fd203003000-7fd203010000 r--p 000bc000 08:02 129173229 /usr/lib/x86_64-linux-gnu/libkrb5.so.3.3* *7fd203010000-7fd203012000 rw-p 000c9000 08:02 129173229 /usr/lib/x86_64-linux-gnu/libkrb5.so.3.3* *7fd203012000-7fd20308e000 r-xp 00000000 08:02 43057497 /lib/x86_64-linux-gnu/libgcrypt.so.11.8.2* *7fd20308e000-7fd20328e000 ---p 0007c000 08:02 43057497 /lib/x86_64-linux-gnu/libgcrypt.so.11.8.2* *7fd20328e000-7fd20328f000 r--p 0007c000 08:02 43057497 /lib/x86_64-linux-gnu/libgcrypt.so.11.8.2* *7fd20328f000-7fd203292000 rw-p 0007d000 08:02 43057497 /lib/x86_64-linux-gnu/libgcrypt.so.11.8.2* *7fd203292000-7fd203349000 r-xp 00000000 08:02 129173818 /usr/lib/x86_64-linux-gnu/libgnutls.so.26.22.6* *7fd203349000-7fd203549000 ---p 000b7000 08:02 129173818 /usr/lib/x86_64-linux-gnu/libgnutls.so.26.22.6* *7fd203549000-7fd20354f000 r--p 000b7000 08:02 129173818 /usr/lib/x86_64-linux-gnu/libgnutls.so.26.22.6* *7fd20354f000-7fd203550000 rw-p 000bd000 08:02 129173818 /usr/lib/x86_64-linux-gnu/libgnutls.so.26.22.6* *7fd203550000-7fd203551000 rw-p 00000000 00:00 0* *7fd203551000-7fd20355a000 r-xp 00000000 08:02 129174659 /usr/lib/x86_64-linux-gnu/libltdl.so.7.3.0* *7fd20355a000-7fd203759000 ---p 00009000 08:02 129174659 /usr/lib/x86_64-linux-gnu/libltdl.so.7.3.0* *7fd203759000-7fd20375a000 r--p 00008000 08:02 129174659 /usr/lib/x86_64-linux-gnu/libltdl.so.7.3.0* *7fd20375a000-7fd20375b000 rw-p 00009000 08:02 129174659 /usr/lib/x86_64-linux-gnu/libltdl.so.7.3.0* *7fd20375b000-7fd203780000 r-xp 00000000 08:02 43057428 /lib/x86_64-linux-gnu/libtinfo.so.5.9* *7fd203780000-7fd20397f000 ---p 00025000 08:02 43057428 /lib/x86_64-linux-gnu/libtinfo.so.5.9* *7fd20397f000-7fd203983000 r--p 00024000 08:02 43057428 /lib/x86_64-linux-gnu/libtinfo.so.5.9* *7fd203983000-7fd203984000 rw-p 00028000 08:02 43057428 /lib/x86_64-linux-gnu/libtinfo.so.5.9* *7fd203984000-7fd2039d1000 r-xp 00000000 08:02 129172944 /usr/lib/x86_64-linux-gnu/libldap_r-2.4.so.2.8.3* *7fd2039d1000-7fd203bd0000 ---p 0004d000 08:02 129172944 /usr/lib/x86_64-linux-gnu/libldap_r-2.4.so.2.8.3* *7fd203bd0000-7fd203bd2000 r--p 0004c000 08:02 129172944 /usr/lib/x86_64-linux-gnu/libldap_r-2.4.so.2.8.3* *7fd203bd2000-7fd203bd3000 rw-p 0004e000 08:02 129172944 /usr/lib/x86_64-linux-gnu/libldap_r-2.4.so.2.8.3* *7fd203bd3000-7fd203bd5000 rw-p 00000000 00:00 0* *7fd203bd5000-7fd203be2000 r-xp 00000000 08:02 129175424 /usr/lib/x86_64-linux-gnu/liblber-2.4.so.2.8.3* *7fd203be2000-7fd203de2000 ---p 0000d000 08:02 129175424 /usr/lib/x86_64-linux-gnu/liblber-2.4.so.2.8.3* *7fd203de2000-7fd203de3000 r--p 0000d000 08:02 129175424 /usr/lib/x86_64-linux-gnu/liblber-2.4.so.2.8.3* *7fd203de3000-7fd203de4000 rw-p 0000e000 08:02 129175424 /usr/lib/x86_64-linux-gnu/liblber-2.4.so.2.8.3* *7fd203de4000-7fd203e28000 r-xp 00000000 08:02 129173227 /usr/lib/x86_64-linux-gnu/libgssapi_krb5.so.2.2* *7fd203e28000-7fd204028000 ---p 00044000 08:02 129173227 /usr/lib/x86_64-linux-gnu/libgssapi_krb5.so.2.2* *7fd204028000-7fd204029000 r--p 00044000 08:02 129173227 /usr/lib/x86_64-linux-gnu/libgssapi_krb5.so.2.2* *7fd204029000-7fd20402b000 rw-p 00045000 08:02 129173227 /usr/lib/x86_64-linux-gnu/libgssapi_krb5.so.2.2* *7fd20402b000-7fd204044000 r-xp 00000000 08:02 129172855 /usr/lib/x86_64-linux-gnu/librtmp.so.0* *7fd204044000-7fd204243000 ---p 00019000 08:02 129172855 /usr/lib/x86_64-linux-gnu/librtmp.so.0* *7fd204243000-7fd204244000 r--p 00018000 08:02 129172855 /usr/lib/x86_64-linux-gnu/librtmp.so.0* *7fd204244000-7fd204245000 rw-p 00019000 08:02 129172855 /usr/lib/x86_64-linux-gnu/librtmp.so.0* *7fd204245000-7fd204276000 r-xp 00000000 08:02 129172819 /usr/lib/x86_64-linux-gnu/libidn.so.11.6.11* *7fd204276000-7fd204476000 ---p 00031000 08:02 129172819 /usr/lib/x86_64-linux-gnu/libidn.so.11.6.11* *7fd204476000-7fd204477000 r--p 00031000 08:02 129172819 /usr/lib/x86_64-linux-gnu/libidn.so.11.6.11* -- Regards, Deepika https://deepikay.wixsite.com/deepika -------------- next part -------------- An HTML attachment was scrubbed... URL: From krice at freeswitch.org Mon Sep 18 06:13:25 2017 From: krice at freeswitch.org (Ken Rice) Date: Mon, 18 Sep 2017 01:13:25 -0500 Subject: [Freeswitch-users] Hanging Call- no CALL DESTROY events even when users ended the call In-Reply-To: References: Message-ID: Segfaults are almost always bugs… however, the bug maynot be in freeswitch… but to figure this out you need to file a project jira see https://freeswitch.org/confluence/display/FREESWITCH/Reporting+Bugs+to+JIRA From: FreeSWITCH-users on behalf of Deepika Yadav Reply-To: FreeSWITCH Users Help Date: Monday, September 18, 2017 at 1:09 AM To: FreeSWITCH Users Help Subject: [Freeswitch-users] Hanging Call- no CALL DESTROY events even when users ended the call Hi, I am using mod_python script for IVR functionality, and facing an issue. Even after users end the call, no CALL DESTROY events are captured by the Freeswitch and after a while Freeswitch itself crashes with following logs: 7fd202913000-7fd202914000 rw-p 0000a000 08:02 129173231 /usr/lib/x86_64-linux-gnu/libkrb5support.so.0.1 7fd202914000-7fd202917000 r-xp 00000000 08:02 43057313 /lib/x86_64-linux-gnu/libcom_err.so.2.1 7fd202917000-7fd202b16000 ---p 00003000 08:02 43057313 /lib/x86_64-linux-gnu/libcom_err.so.2.1 7fd202b16000-7fd202b17000 r--p 00002000 08:02 43057313 /lib/x86_64-linux-gnu/libcom_err.so.2.1 7fd202b17000-7fd202b18000 rw-p 00003000 08:02 43057313 /lib/x86_64-linux-gnu/libcom_err.so.2.1 7fd202b18000-7fd202b44000 r-xp 00000000 08:02 129177695 /usr/lib/x86_64-linux-gnu/libk5crypto.so.3.1 7fd202b44000-7fd202d43000 ---p 0002c000 08:02 129177695 /usr/lib/x86_64-linux-gnu/libk5crypto.so.3.1 7fd202d43000-7fd202d45000 r--p 0002b000 08:02 129177695 /usr/lib/x86_64-linux-gnu/libk5crypto.so.3.1 7fd202d45000-7fd202d46000 rw-p 0002d000 08:02 129177695 /usr/lib/x86_64-linux-gnu/libk5crypto.so.3.1 7fd202d46000-7fd202d47000 rw-p 00000000 00:00 0 7fd202d47000-7fd202e03000 r-xp 00000000 08:02 129173229 /usr/lib/x86_64-linux-gnu/libkrb5.so.3.3 7fd202e03000-7fd203003000 ---p 000bc000 08:02 129173229 /usr/lib/x86_64-linux-gnu/libkrb5.so.3.3 7fd203003000-7fd203010000 r--p 000bc000 08:02 129173229 /usr/lib/x86_64-linux-gnu/libkrb5.so.3.3 7fd203010000-7fd203012000 rw-p 000c9000 08:02 129173229 /usr/lib/x86_64-linux-gnu/libkrb5.so.3.3 7fd203012000-7fd20308e000 r-xp 00000000 08:02 43057497 /lib/x86_64-linux-gnu/libgcrypt.so.11.8.2 7fd20308e000-7fd20328e000 ---p 0007c000 08:02 43057497 /lib/x86_64-linux-gnu/libgcrypt.so.11.8.2 7fd20328e000-7fd20328f000 r--p 0007c000 08:02 43057497 /lib/x86_64-linux-gnu/libgcrypt.so.11.8.2 7fd20328f000-7fd203292000 rw-p 0007d000 08:02 43057497 /lib/x86_64-linux-gnu/libgcrypt.so.11.8.2 7fd203292000-7fd203349000 r-xp 00000000 08:02 129173818 /usr/lib/x86_64-linux-gnu/libgnutls.so.26.22.6 7fd203349000-7fd203549000 ---p 000b7000 08:02 129173818 /usr/lib/x86_64-linux-gnu/libgnutls.so.26.22.6 7fd203549000-7fd20354f000 r--p 000b7000 08:02 129173818 /usr/lib/x86_64-linux-gnu/libgnutls.so.26.22.6 7fd20354f000-7fd203550000 rw-p 000bd000 08:02 129173818 /usr/lib/x86_64-linux-gnu/libgnutls.so.26.22.6 7fd203550000-7fd203551000 rw-p 00000000 00:00 0 7fd203551000-7fd20355a000 r-xp 00000000 08:02 129174659 /usr/lib/x86_64-linux-gnu/libltdl.so.7.3.0 7fd20355a000-7fd203759000 ---p 00009000 08:02 129174659 /usr/lib/x86_64-linux-gnu/libltdl.so.7.3.0 7fd203759000-7fd20375a000 r--p 00008000 08:02 129174659 /usr/lib/x86_64-linux-gnu/libltdl.so.7.3.0 7fd20375a000-7fd20375b000 rw-p 00009000 08:02 129174659 /usr/lib/x86_64-linux-gnu/libltdl.so.7.3.0 7fd20375b000-7fd203780000 r-xp 00000000 08:02 43057428 /lib/x86_64-linux-gnu/libtinfo.so.5.9 7fd203780000-7fd20397f000 ---p 00025000 08:02 43057428 /lib/x86_64-linux-gnu/libtinfo.so.5.9 7fd20397f000-7fd203983000 r--p 00024000 08:02 43057428 /lib/x86_64-linux-gnu/libtinfo.so.5.9 7fd203983000-7fd203984000 rw-p 00028000 08:02 43057428 /lib/x86_64-linux-gnu/libtinfo.so.5.9 7fd203984000-7fd2039d1000 r-xp 00000000 08:02 129172944 /usr/lib/x86_64-linux-gnu/libldap_r-2.4.so.2.8.3 7fd2039d1000-7fd203bd0000 ---p 0004d000 08:02 129172944 /usr/lib/x86_64-linux-gnu/libldap_r-2.4.so.2.8.3 7fd203bd0000-7fd203bd2000 r--p 0004c000 08:02 129172944 /usr/lib/x86_64-linux-gnu/libldap_r-2.4.so.2.8.3 7fd203bd2000-7fd203bd3000 rw-p 0004e000 08:02 129172944 /usr/lib/x86_64-linux-gnu/libldap_r-2.4.so.2.8.3 7fd203bd3000-7fd203bd5000 rw-p 00000000 00:00 0 7fd203bd5000-7fd203be2000 r-xp 00000000 08:02 129175424 /usr/lib/x86_64-linux-gnu/liblber-2.4.so.2.8.3 7fd203be2000-7fd203de2000 ---p 0000d000 08:02 129175424 /usr/lib/x86_64-linux-gnu/liblber-2.4.so.2.8.3 7fd203de2000-7fd203de3000 r--p 0000d000 08:02 129175424 /usr/lib/x86_64-linux-gnu/liblber-2.4.so.2.8.3 7fd203de3000-7fd203de4000 rw-p 0000e000 08:02 129175424 /usr/lib/x86_64-linux-gnu/liblber-2.4.so.2.8.3 7fd203de4000-7fd203e28000 r-xp 00000000 08:02 129173227 /usr/lib/x86_64-linux-gnu/libgssapi_krb5.so.2.2 7fd203e28000-7fd204028000 ---p 00044000 08:02 129173227 /usr/lib/x86_64-linux-gnu/libgssapi_krb5.so.2.2 7fd204028000-7fd204029000 r--p 00044000 08:02 129173227 /usr/lib/x86_64-linux-gnu/libgssapi_krb5.so.2.2 7fd204029000-7fd20402b000 rw-p 00045000 08:02 129173227 /usr/lib/x86_64-linux-gnu/libgssapi_krb5.so.2.2 7fd20402b000-7fd204044000 r-xp 00000000 08:02 129172855 /usr/lib/x86_64-linux-gnu/librtmp.so.0 7fd204044000-7fd204243000 ---p 00019000 08:02 129172855 /usr/lib/x86_64-linux-gnu/librtmp.so.0 7fd204243000-7fd204244000 r--p 00018000 08:02 129172855 /usr/lib/x86_64-linux-gnu/librtmp.so.0 7fd204244000-7fd204245000 rw-p 00019000 08:02 129172855 /usr/lib/x86_64-linux-gnu/librtmp.so.0 7fd204245000-7fd204276000 r-xp 00000000 08:02 129172819 /usr/lib/x86_64-linux-gnu/libidn.so.11.6.11 7fd204276000-7fd204476000 ---p 00031000 08:02 129172819 /usr/lib/x86_64-linux-gnu/libidn.so.11.6.11 7fd204476000-7fd204477000 r--p 00031000 08:02 129172819 /usr/lib/x86_64-linux-gnu/libidn.so.11.6.11 -- Regards, Deepika https://deepikay.wixsite.com/deepika _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Mon Sep 18 06:32:32 2017 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Mon, 18 Sep 2017 08:32:32 +0200 Subject: [Freeswitch-users] How to record User Voice and listen for the DTMF simultaneously In-Reply-To: References: Message-ID: Playandgetdigits() sent from mobile cell: +39 347 266 56 18 Giovanni Maruzzelli OpenTelecom.IT On Sep 17, 2017 11:28, "Deepika Yadav" wrote: > Hi, > > I want to give users an option to record their voice by presssing some > digit in the IVR menu through my python script. I use the code as follows: > > session.recordFile(filename, 120, 500, 3) > > However, this code will block for 2 minutes and then proceed. In case a > user has little to say and after 1 minute wants to stop the recording, how > I can do that. Is there any function similar to "session.playAndGetDigits" > to also hear for DTMFs at the same time? > > > -- > Regards, > Deepika > https://deepikay.wixsite.com/deepika > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From ynasida at gmail.com Mon Sep 18 07:03:47 2017 From: ynasida at gmail.com (Yuriy Nasida) Date: Mon, 18 Sep 2017 10:03:47 +0300 Subject: [Freeswitch-users] vm_cc weird behavior In-Reply-To: References: Message-ID: Update. It isn't just a lot of emails. FS got segfault afte 69th attampt... Sep 18 02:59:07 fs1 kernel: [15421422.922960] freeswitch[43697]: segfault at 7f30e35249b8 ip 00007f313e6c3ef3 sp 00007f30e35249a0 error 6 in libcurl.so.4.3.0[7f313e69d000+70000] On 16 September 2017 at 18:46, Yuriy Nasida wrote: > Any advice? > > 14 Сен 2017 г. 16:00 пользователь "Yuriy Nasida" > написал: > > Hi, >> >> I noted that FS sends a lot of emails to my email box instead of only one >> in case I use vm_cc. >> >> It's very simple configuration. I just set vm_cc before mod_voicemail is >> called (according wiki page) and leave voicemail message. >> For main emails box I recieve only one message as expected. >> But I have ~60 exactly same messages at email box from vm_cc. >> >> In console I also noted that FS really sends emails many times without >> obvios reason. >> >> mod_voicemail.c:2832 Deliver VM to yuriytest2 at mydomain.com >> mod_voicemail.c:1925 Update MWI: Processing for yuriytest2 at mydomain.com >> in inbox >> mod_voicemail.c:1950 Update MWI: Messages Waiting yes >> mod_voicemail.c:1951 Update MWI: Update Reason NEW >> mod_voicemail.c:1952 Update MWI: Message Account yuriytest2 at mydomain.com >> mod_voicemail.c:1953 Update MWI: Voice Message 69/0 >> switch_utils.c:1180 Emailed file [/tmp/mail.1505386894e5a8] to [ >> myemailbox at gmail.com] >> mod_voicemail.c:3042 Sending message to myemailbox at gmail.com >> >> Next FS does this again and tries to Deliver VM to >> yuriytest2 at mydomain.com. >> >> Please advice. >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From deepikay at iiitd.ac.in Mon Sep 18 07:10:55 2017 From: deepikay at iiitd.ac.in (Deepika Yadav) Date: Mon, 18 Sep 2017 12:40:55 +0530 Subject: [Freeswitch-users] How to record User Voice and listen for the DTMF simultaneously In-Reply-To: References: Message-ID: Playandgetdigits() - plays a file and gets digits at the same time. However I need to initiate recording of the user voice and get the digits for quitting On Mon, Sep 18, 2017 at 12:02 PM, Giovanni Maruzzelli wrote: > Playandgetdigits() > > sent from mobile > cell: +39 347 266 56 18 > Giovanni Maruzzelli > OpenTelecom.IT > > On Sep 17, 2017 11:28, "Deepika Yadav" wrote: > >> Hi, >> >> I want to give users an option to record their voice by presssing some >> digit in the IVR menu through my python script. I use the code as follows: >> >> session.recordFile(filename, 120, 500, 3) >> >> However, this code will block for 2 minutes and then proceed. In case a >> user has little to say and after 1 minute wants to stop the recording, how >> I can do that. Is there any function similar to "session.playAndGetDigits" >> to also hear for DTMFs at the same time? >> >> >> -- >> Regards, >> Deepika >> https://deepikay.wixsite.com/deepika >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Regards, Deepika https://deepikay.wixsite.com/deepika -------------- next part -------------- An HTML attachment was scrubbed... URL: From shaun.stokes at itec-support.co.uk Mon Sep 18 08:03:47 2017 From: shaun.stokes at itec-support.co.uk (Shaun Stokes) Date: Mon, 18 Sep 2017 08:03:47 +0000 Subject: [Freeswitch-users] TLS and SRTP commercial certs go in which file? In-Reply-To: <15e8e25d288.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com> References: <09ea1523-534d-b769-38b7-1a729a45241f@xbipin.com> <02addee6-3a74-b95b-b2ab-e90851e6f3e2@xbipin.com> <6FD2F8B5BB72834E9939AEDF9FB802A901E869C97A@mbx-01.sysconfig.co.uk> <6FD2F8B5BB72834E9939AEDF9FB802A901E869CA54@mbx-01.sysconfig.co.uk> <15e8e25d288.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com> Message-ID: <6FD2F8B5BB72834E9939AEDF9FB802A901E869D6ED@mbx-01.sysconfig.co.uk> Hi Bipin, Putting the same into all 4 pem's certainly works, and that's correct no intermediary or root ca required providing you have the public and private key in your pem files. The wiki article for SIP_TLS could do with being updated, but I don't know exactly what each of the 4 certs are used for. This is my best guess at what each of the certificates are used for. agent.pem - FreeSWITCH public server cert tls.pem - Used for TLS communication dtls-srtp.pem - Used for DTLS\SRTP communication wss.pem - Used for WebRTC communication Shaun From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bipin Patel Sent: 17 September 2017 05:43 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] TLS and SRTP commercial certs go in which file? Hi, Thanks for that info so if I understood it right the agent.pem file will have my cert and key inside it only and no intermediary or root ca cert is required at all, but what goes in the tls.pem file? On September 17, 2017 5:45:28 AM Shaun Stokes > wrote: Hi Bipin, We've found that these are the certs which FreeSWITCH will use, the pem file will need the public and private key (same as your wss cert). You'll also need to make sure the user for FreeSWITCH has read permission to the certs. agent.pem dtls-srtp.pem tls.pem wss.pem FreeSWITCH doesn't seem to need the intermediary and root cert of the CA. Here are some of the TLS parameters you might also want on your SIP profile. Name: tls Value: true Name: tls-bind-params Value: transport=tls Name: tls-cert-dir Value: "Your Cert Directory Path" Name: tls-sip-port Value: 5061 Name: tls-verify-date Value: true Name: tls-verify-depth Value: 2 Name: tls-verify-policy Value: all|subjects_all Name: tls-version Value: tlsv1.2 Shaun From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bipin Patel Sent: 16 September 2017 06:49 To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] TLS and SRTP commercial certs go in which file? hi, no one? Regards, Bipin ________________________________ -------- Original Message -------- Subject: [Freeswitch-users] TLS and SRTP commercial certs go in which file? From: Bipin Patel To: FreeSWITCH Users Help Date: 9/15/2017, 3:44:33 PM hi, when i setup verto on my server i used commercial certificates with wss.pem containing the following and all that works brilliant: -----BEGIN CERTIFICATE----- -----END CERTIFICATE----- -----BEGIN RSA PRIVATE KEY----- -----END RSA PRIVATE KEY----- -----BEGIN CERTIFICATE----- -----END CERTIFICATE----- -----BEGIN CERTIFICATE----- -----END CERTIFICATE----- now i want to use the same certificate for TLS and SRTP and i was reading the docs and it mentioned to create a agent.pem file with the actual server cert and key but where do i copy the intermediatory and root cert of the CA, which folders do both go in and with what filename? any help is appreciated -- Regards, Bipin ________________________________ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ [http://www.itec-support.co.uk/wp-content/uploads/2016/07/email_logo.jpg] Shaun Stokes - Infrastructure Analyst T : 01453 700713 E : shaun.stokes at itec-support.co.uk W : www.itec-support.co.uk Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road, Stroud, Gloucestershire GL5 3QF Company No. 06908001 CONFIDENTIALITY NOTICE This communication and the information it contains are intended for the person or organisation to which it is addressed. Its contents are confidential and may be protected in law. Unauthorised use, copying or disclosure of any of it may be unlawful. If you are not the intended recipient, please contact us immediately. The contents of any attachments in this e-mail may contain software viruses, which could damage your own computer system. While ITEC Support has taken every reasonable precaution to minimise this risk, we cannot accept liability for any damage which you sustain as a result of software viruses. You should carry out your own virus checking procedure before opening any attachment. ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Mon Sep 18 08:08:09 2017 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Mon, 18 Sep 2017 10:08:09 +0200 Subject: [Freeswitch-users] How to record User Voice and listen for the DTMF simultaneously In-Reply-To: References: Message-ID: You may use the set-meta* apps. Check local extension in demo configuration sent from mobile cell: +39 347 266 56 18 Giovanni Maruzzelli OpenTelecom.IT On Sep 18, 2017 09:12, "Deepika Yadav" wrote: > Playandgetdigits() - plays a file and gets digits at the same time. > However I need to initiate recording of the user voice and get the digits > for quitting > > > > On Mon, Sep 18, 2017 at 12:02 PM, Giovanni Maruzzelli > wrote: > >> Playandgetdigits() >> >> sent from mobile >> cell: +39 347 266 56 18 >> Giovanni Maruzzelli >> OpenTelecom.IT >> >> On Sep 17, 2017 11:28, "Deepika Yadav" wrote: >> >>> Hi, >>> >>> I want to give users an option to record their voice by presssing some >>> digit in the IVR menu through my python script. I use the code as follows: >>> >>> session.recordFile(filename, 120, 500, 3) >>> >>> However, this code will block for 2 minutes and then proceed. In case a >>> user has little to say and after 1 minute wants to stop the recording, how >>> I can do that. Is there any function similar to "session.playAndGetDigits" >>> to also hear for DTMFs at the same time? >>> >>> >>> -- >>> Regards, >>> Deepika >>> https://deepikay.wixsite.com/deepika >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Regards, > Deepika > https://deepikay.wixsite.com/deepika > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bipin at xbipin.com Mon Sep 18 10:32:54 2017 From: bipin at xbipin.com (Bipin Patel) Date: Mon, 18 Sep 2017 14:32:54 +0400 Subject: [Freeswitch-users] TLS and SRTP commercial certs go in which file? In-Reply-To: <6FD2F8B5BB72834E9939AEDF9FB802A901E869D6ED@mbx-01.sysconfig.co.uk> References: <09ea1523-534d-b769-38b7-1a729a45241f@xbipin.com> <02addee6-3a74-b95b-b2ab-e90851e6f3e2@xbipin.com> <6FD2F8B5BB72834E9939AEDF9FB802A901E869C97A@mbx-01.sysconfig.co.uk> <6FD2F8B5BB72834E9939AEDF9FB802A901E869CA54@mbx-01.sysconfig.co.uk> <15e8e25d288.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com> <6FD2F8B5BB72834E9939AEDF9FB802A901E869D6ED@mbx-01.sysconfig.co.uk> Message-ID: hi, after playing around i figured this, some corrections i need to make for others willing to use the same: agent.pem file needs to have ur commercial cert followed by the private key but if u just use that zoiper for android will give a warning about untrusted cert so we need to also add the intermediary followed by the root CA cert in the the cafile.pem to avoid this warning on client side. wss.pem is used for verto which needs to have ur commercial cert followed by its private key followed by intermediary cert followed by root CA cert, if u dont use verto then ignore this file and FS will self create it when using only TLS and SRTP. tls.pem file i think is used as default cert but i just copied the data from agent.pem to this dtls-srtp.pem is used for verto so i think that got created by default also bear in mind zoiper for android doesnt allow to import ur own certs but just uses its own self signed cert so i add to set tls-verify-policy to none to make it work Regards, Bipin ------------------------------------------------------------------------ -------- Original Message -------- Subject: Re: [Freeswitch-users] TLS and SRTP commercial certs go in which file? From: Shaun Stokes To: FreeSWITCH Users Help Date: 9/18/2017, 12:03:47 PM > > Hi Bipin, > > Putting the same into all 4 pem’s certainly works, and that’s correct > no intermediary or root ca required providing you have the public and > private key in your pem files. > > The wiki article for SIP_TLS could do with being updated, but I don’t > know exactly what each of the 4 certs are used for. > > This is my best guess at what each of the certificates are used for. > > agent.pem – FreeSWITCH public server cert > > tls.pem – Used for TLS communication > > dtls-srtp.pem – Used for DTLS\SRTP communication > > wss.pem – Used for WebRTC communication > > Shaun > > *From:*FreeSWITCH-users > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of > *Bipin Patel > *Sent:* 17 September 2017 05:43 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] TLS and SRTP commercial certs go in > which file? > > Hi, > > Thanks for that info so if I understood it right the agent.pem file > will have my cert and key inside it only and no intermediary or root > ca cert is required at all, but what goes in the tls.pem file? > > On September 17, 2017 5:45:28 AM Shaun Stokes > > wrote: > > Hi Bipin, > > We've found that these are the certs which FreeSWITCH will use, > the pem file will need the public and private key (same as your > wss cert). You'll also need to make sure the user for FreeSWITCH > has read permission to the certs. > > agent.pem > > dtls-srtp.pem > > tls.pem > > wss.pem > > FreeSWITCH doesn't seem to need the intermediary and root cert of > the CA. > > Here are some of the TLS parameters you might also want on your > SIP profile. > > Name: tls > > Value: true > > Name: tls-bind-params > > Value: transport=tls > > Name: tls-cert-dir > > Value: "Your Cert Directory Path" > > Name: tls-sip-port > > Value: 5061 > > Name: tls-verify-date > > Value: true > > Name: tls-verify-depth > > Value: 2 > > Name: tls-verify-policy > > Value: all|subjects_all > > Name: tls-version > > Value: tlsv1.2 > > Shaun > > *From:*FreeSWITCH-users > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf > Of *Bipin Patel > *Sent:* 16 September 2017 06:49 > *To:* FreeSWITCH Users Help > > *Subject:* Re: [Freeswitch-users] TLS and SRTP commercial certs go > in which file? > > hi, > > no one? > > Regards, > Bipin > > ------------------------------------------------------------------------ > > -------- Original Message -------- > Subject: [Freeswitch-users] TLS and SRTP commercial certs go in > which file? > From: Bipin Patel > To: FreeSWITCH Users Help > > Date: 9/15/2017, 3:44:33 PM > > hi, > > when i setup verto on my server i used commercial certificates > with wss.pem containing the following and all that works > brilliant: > -----BEGIN CERTIFICATE----- > > -----END CERTIFICATE----- > -----BEGIN RSA PRIVATE KEY----- > > -----END RSA PRIVATE KEY----- > -----BEGIN CERTIFICATE----- > > -----END CERTIFICATE----- > -----BEGIN CERTIFICATE----- > > -----END CERTIFICATE----- > > now i want to use the same certificate for TLS and SRTP and i > was reading the docs and it mentioned to create a agent.pem > file with the actual server cert and key but where do i copy > the intermediatory and root cert of the CA, which folders do > both go in and with what filename? > > any help is appreciated > > -- > Regards, > Bipin > > ------------------------------------------------------------------------ > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > ______________________________________________________________________ > This message has been checked for all known viruses by MessageLabs > Virus Scanning Service. > ______________________________________________________________________ > > > > Shaun Stokes - Infrastructure Analyst > > > > > T : > > > > 01453 700713 > > E : > > > > shaun.stokes at itec-support.co.uk > > > W : > > > > www.itec-support.co.uk > > Registered Address :- ITEC Support, Suite 2 Prospect House, Bath > Road, Stroud, Gloucestershire GL5 3QF > Company No. 06908001 > > > CONFIDENTIALITY NOTICE > This communication and the information it contains are intended > for the person or organisation to which it is addressed. Its > contents are confidential and may be protected in law. > Unauthorised use, copying or disclosure of any of it may be > unlawful. If you are not the intended recipient, please contact us > immediately. > The contents of any attachments in this e-mail may contain > software viruses, which could damage your own computer system. > While ITEC Support has taken every reasonable precaution to > minimise this risk, we cannot accept liability for any damage > which you sustain as a result of software viruses. You should > carry out your own virus checking procedure before opening any > attachment. > > > ______________________________________________________________________ > This message has been checked for all known viruses by MessageLabs > Virus Scanning Service. > ______________________________________________________________________ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ______________________________________________________________________ > This message has been checked for all known viruses by MessageLabs > Virus Scanning Service. > ______________________________________________________________________ > > > ______________________________________________________________________ > This message has been checked for all known viruses by MessageLabs > Virus Scanning Service. > ______________________________________________________________________ > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From gregor at infomedia.si Mon Sep 18 10:43:39 2017 From: gregor at infomedia.si (Gregor Nanger) Date: Mon, 18 Sep 2017 12:43:39 +0200 Subject: [Freeswitch-users] TLS and SRTP commercial certs go in which file? In-Reply-To: References: <09ea1523-534d-b769-38b7-1a729a45241f@xbipin.com> <02addee6-3a74-b95b-b2ab-e90851e6f3e2@xbipin.com> <6FD2F8B5BB72834E9939AEDF9FB802A901E869C97A@mbx-01.sysconfig.co.uk> <6FD2F8B5BB72834E9939AEDF9FB802A901E869CA54@mbx-01.sysconfig.co.uk> <15e8e25d288.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com> <6FD2F8B5BB72834E9939AEDF9FB802A901E869D6ED@mbx-01.sysconfig.co.uk> Message-ID: Hi! I have wss.pem as you described And for tls, I just copied wss.pem. Regards, Gregor 2017-09-18 12:32 GMT+02:00 Bipin Patel : > hi, > > after playing around i figured this, some corrections i need to make for > others willing to use the same: > > agent.pem file needs to have ur commercial cert followed by the private > key but if u just use that zoiper for android will give a warning about > untrusted cert so we need to also add the intermediary followed by the root > CA cert in the the cafile.pem to avoid this warning on client side. > wss.pem is used for verto which needs to have ur commercial cert followed > by its private key followed by intermediary cert followed by root CA cert, > if u dont use verto then ignore this file and FS will self create it when > using only TLS and SRTP. > tls.pem file i think is used as default cert but i just copied the data > from agent.pem to this > dtls-srtp.pem is used for verto so i think that got created by default also > > bear in mind zoiper for android doesnt allow to import ur own certs but > just uses its own self signed cert so i add to set tls-verify-policy to > none to make it work > > > Regards, > Bipin > > > ------------------------------ > -------- Original Message -------- > Subject: Re: [Freeswitch-users] TLS and SRTP commercial certs go in which > file? > From: Shaun Stokes > > To: FreeSWITCH Users Help > > Date: 9/18/2017, 12:03:47 PM > > Hi Bipin, > > > > Putting the same into all 4 pem’s certainly works, and that’s correct no > intermediary or root ca required providing you have the public and private > key in your pem files. > > > > The wiki article for SIP_TLS could do with being updated, but I don’t know > exactly what each of the 4 certs are used for. > > > > This is my best guess at what each of the certificates are used for. > > agent.pem – FreeSWITCH public server cert > > tls.pem – Used for TLS communication > > dtls-srtp.pem – Used for DTLS\SRTP communication > > wss.pem – Used for WebRTC communication > > > > Shaun > > > > *From:* FreeSWITCH-users [mailto:freeswitch-users- > bounces at lists.freeswitch.org > ] *On Behalf Of *Bipin > Patel > *Sent:* 17 September 2017 05:43 > *To:* FreeSWITCH Users Help > > *Subject:* Re: [Freeswitch-users] TLS and SRTP commercial certs go in > which file? > > > > Hi, > > Thanks for that info so if I understood it right the agent.pem file will > have my cert and key inside it only and no intermediary or root ca cert is > required at all, but what goes in the tls.pem file? > > On September 17, 2017 5:45:28 AM Shaun Stokes < > shaun.stokes at itec-support.co.uk> wrote: > > Hi Bipin, > > > > We've found that these are the certs which FreeSWITCH will use, the pem > file will need the public and private key (same as your wss cert). You'll > also need to make sure the user for FreeSWITCH has read permission to the > certs. > > agent.pem > > dtls-srtp.pem > > tls.pem > > wss.pem > > > > FreeSWITCH doesn't seem to need the intermediary and root cert of the CA. > > > > Here are some of the TLS parameters you might also want on your SIP > profile. > > > > Name: tls > > Value: true > > > > Name: tls-bind-params > > Value: transport=tls > > > > Name: tls-cert-dir > > Value: "Your Cert Directory Path" > > > > Name: tls-sip-port > > Value: 5061 > > > > Name: tls-verify-date > > Value: true > > > > Name: tls-verify-depth > > Value: 2 > > > > Name: tls-verify-policy > > Value: all|subjects_all > > > > Name: tls-version > > Value: tlsv1.2 > > > > > > Shaun > > > > *From:* FreeSWITCH-users [mailto:freeswitch-users- > bounces at lists.freeswitch.org > ] *On Behalf Of *Bipin > Patel > *Sent:* 16 September 2017 06:49 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] TLS and SRTP commercial certs go in > which file? > > > > hi, > > no one? > > Regards, > Bipin > ------------------------------ > > -------- Original Message -------- > Subject: [Freeswitch-users] TLS and SRTP commercial certs go in which file? > From: Bipin Patel > To: FreeSWITCH Users Help > > Date: 9/15/2017, 3:44:33 PM > > hi, > > when i setup verto on my server i used commercial certificates with > wss.pem containing the following and all that works brilliant: > -----BEGIN CERTIFICATE----- > > -----END CERTIFICATE----- > -----BEGIN RSA PRIVATE KEY----- > > -----END RSA PRIVATE KEY----- > -----BEGIN CERTIFICATE----- > > -----END CERTIFICATE----- > -----BEGIN CERTIFICATE----- > > -----END CERTIFICATE----- > > now i want to use the same certificate for TLS and SRTP and i was reading > the docs and it mentioned to create a agent.pem file with the actual server > cert and key but where do i copy the intermediatory and root cert of the > CA, which folders do both go in and with what filename? > > any help is appreciated > > -- > Regards, > Bipin > ------------------------------ > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > ______________________________________________________________________ > This message has been checked for all known viruses by MessageLabs Virus > Scanning Service. > ______________________________________________________________________ > > Shaun Stokes - Infrastructure Analyst > > > T : > > 01453 700713 > > E : > > shaun.stokes at itec-support.co.uk > > W : > > www.itec-support.co.uk > > Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road, > Stroud, Gloucestershire GL5 3QF > Company No. 06908001 > > > CONFIDENTIALITY NOTICE > This communication and the information it contains are intended for the > person or organisation to which it is addressed. Its contents are > confidential and may be protected in law. Unauthorised use, copying or > disclosure of any of it may be unlawful. If you are not the intended > recipient, please contact us immediately. > The contents of any attachments in this e-mail may contain software > viruses, which could damage your own computer system. While ITEC Support > has taken every reasonable precaution to minimise this risk, we cannot > accept liability for any damage which you sustain as a result of software > viruses. You should carry out your own virus checking procedure before > opening any attachment. > > > ______________________________________________________________________ > This message has been checked for all known viruses by MessageLabs Virus > Scanning Service. > ______________________________________________________________________ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ______________________________________________________________________ > This message has been checked for all known viruses by MessageLabs Virus > Scanning Service. > ______________________________________________________________________ > > ______________________________________________________________________ > This message has been checked for all known viruses by MessageLabs Virus > Scanning Service. > ______________________________________________________________________ > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Gregor Nanger *CTO* t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia • www.infomedia.si -------------- next part -------------- An HTML attachment was scrubbed... URL: From agubbe at gmail.com Mon Sep 18 11:03:23 2017 From: agubbe at gmail.com (=?UTF-8?Q?Agust=C3=AD_Ubalde_Bellot?=) Date: Mon, 18 Sep 2017 13:03:23 +0200 Subject: [Freeswitch-users] 488 INCOMPATIBLE DESTINATION with Firefox 55.03 Message-ID: Hi all, I am testing WebRTC calls with Firefox 55.03 and I have a problem receiving calls. The case is that FreeSWITCH receives a call with G711 codec from an external Asterisk, but when it processes INVITE, it receives a 488 from the browser: ====================================================================== nta.c: 3297 agent_recv_response () nta: received 488 Not Acceptable Here for INVITE (112529915) ====================================================================== Any idea what may be going on? An additional note: calls from the navbegador to this same Asterisk are generated correctly, and in Chrome the 2 cases work correctly. Regards, Agustí Ubalde -------------- next part -------------- An HTML attachment was scrubbed... URL: From bipin at xbipin.com Mon Sep 18 12:19:05 2017 From: bipin at xbipin.com (Bipin Patel) Date: Mon, 18 Sep 2017 16:19:05 +0400 Subject: [Freeswitch-users] TLS and SRTP commercial certs go in which file? In-Reply-To: References: <09ea1523-534d-b769-38b7-1a729a45241f@xbipin.com> <02addee6-3a74-b95b-b2ab-e90851e6f3e2@xbipin.com> <6FD2F8B5BB72834E9939AEDF9FB802A901E869C97A@mbx-01.sysconfig.co.uk> <6FD2F8B5BB72834E9939AEDF9FB802A901E869CA54@mbx-01.sysconfig.co.uk> <15e8e25d288.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com> <6FD2F8B5BB72834E9939AEDF9FB802A901E869D6ED@mbx-01.sysconfig.co.uk> Message-ID: <15e94ed76c0.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com> Hi, So it would mean if you put the cert, key, intermediary cert and root ca cert all in agent.pem and delete the cafile.pem file then it should work although I didn't try that with zoiper if it gives a warning or no. Frankly if the devs gave a reply to this cleaning the doubts I could update the docs on what file is required for what function and it needs to have what all certs and key inside it when using commercial certificates. On September 18, 2017 2:46:03 PM Gregor Nanger wrote: > Hi! > > I have wss.pem as you described > > And for tls, I just copied wss.pem. > > Regards, Gregor > > 2017-09-18 12:32 GMT+02:00 Bipin Patel : > >> hi, >> >> after playing around i figured this, some corrections i need to make for >> others willing to use the same: >> >> agent.pem file needs to have ur commercial cert followed by the private >> key but if u just use that zoiper for android will give a warning about >> untrusted cert so we need to also add the intermediary followed by the root >> CA cert in the the cafile.pem to avoid this warning on client side. >> wss.pem is used for verto which needs to have ur commercial cert followed >> by its private key followed by intermediary cert followed by root CA cert, >> if u dont use verto then ignore this file and FS will self create it when >> using only TLS and SRTP. >> tls.pem file i think is used as default cert but i just copied the data >> from agent.pem to this >> dtls-srtp.pem is used for verto so i think that got created by default also >> >> bear in mind zoiper for android doesnt allow to import ur own certs but >> just uses its own self signed cert so i add to set tls-verify-policy to >> none to make it work >> >> >> Regards, >> Bipin >> >> >> ------------------------------ >> -------- Original Message -------- >> Subject: Re: [Freeswitch-users] TLS and SRTP commercial certs go in which >> file? >> From: Shaun Stokes >> >> To: FreeSWITCH Users Help >> >> Date: 9/18/2017, 12:03:47 PM >> >> Hi Bipin, >> >> >> >> Putting the same into all 4 pem’s certainly works, and that’s correct no >> intermediary or root ca required providing you have the public and private >> key in your pem files. >> >> >> >> The wiki article for SIP_TLS could do with being updated, but I don’t know >> exactly what each of the 4 certs are used for. >> >> >> >> This is my best guess at what each of the certificates are used for. >> >> agent.pem – FreeSWITCH public server cert >> >> tls.pem – Used for TLS communication >> >> dtls-srtp.pem – Used for DTLS\SRTP communication >> >> wss.pem – Used for WebRTC communication >> >> >> >> Shaun >> >> >> >> *From:* FreeSWITCH-users [mailto:freeswitch-users- >> bounces at lists.freeswitch.org >> ] *On Behalf Of *Bipin >> Patel >> *Sent:* 17 September 2017 05:43 >> *To:* FreeSWITCH Users Help >> >> *Subject:* Re: [Freeswitch-users] TLS and SRTP commercial certs go in >> which file? >> >> >> >> Hi, >> >> Thanks for that info so if I understood it right the agent.pem file will >> have my cert and key inside it only and no intermediary or root ca cert is >> required at all, but what goes in the tls.pem file? >> >> On September 17, 2017 5:45:28 AM Shaun Stokes < >> shaun.stokes at itec-support.co.uk> wrote: >> >> Hi Bipin, >> >> >> >> We've found that these are the certs which FreeSWITCH will use, the pem >> file will need the public and private key (same as your wss cert). You'll >> also need to make sure the user for FreeSWITCH has read permission to the >> certs. >> >> agent.pem >> >> dtls-srtp.pem >> >> tls.pem >> >> wss.pem >> >> >> >> FreeSWITCH doesn't seem to need the intermediary and root cert of the CA. >> >> >> >> Here are some of the TLS parameters you might also want on your SIP >> profile. >> >> >> >> Name: tls >> >> Value: true >> >> >> >> Name: tls-bind-params >> >> Value: transport=tls >> >> >> >> Name: tls-cert-dir >> >> Value: "Your Cert Directory Path" >> >> >> >> Name: tls-sip-port >> >> Value: 5061 >> >> >> >> Name: tls-verify-date >> >> Value: true >> >> >> >> Name: tls-verify-depth >> >> Value: 2 >> >> >> >> Name: tls-verify-policy >> >> Value: all|subjects_all >> >> >> >> Name: tls-version >> >> Value: tlsv1.2 >> >> >> >> >> >> Shaun >> >> >> >> *From:* FreeSWITCH-users [mailto:freeswitch-users- >> bounces at lists.freeswitch.org >> ] *On Behalf Of *Bipin >> Patel >> *Sent:* 16 September 2017 06:49 >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] TLS and SRTP commercial certs go in >> which file? >> >> >> >> hi, >> >> no one? >> >> Regards, >> Bipin >> ------------------------------ >> >> -------- Original Message -------- >> Subject: [Freeswitch-users] TLS and SRTP commercial certs go in which file? >> From: Bipin Patel >> To: FreeSWITCH Users Help >> >> Date: 9/15/2017, 3:44:33 PM >> >> hi, >> >> when i setup verto on my server i used commercial certificates with >> wss.pem containing the following and all that works brilliant: >> -----BEGIN CERTIFICATE----- >> >> -----END CERTIFICATE----- >> -----BEGIN RSA PRIVATE KEY----- >> >> -----END RSA PRIVATE KEY----- >> -----BEGIN CERTIFICATE----- >> >> -----END CERTIFICATE----- >> -----BEGIN CERTIFICATE----- >> >> -----END CERTIFICATE----- >> >> now i want to use the same certificate for TLS and SRTP and i was reading >> the docs and it mentioned to create a agent.pem file with the actual server >> cert and key but where do i copy the intermediatory and root cert of the >> CA, which folders do both go in and with what filename? >> >> any help is appreciated >> >> -- >> Regards, >> Bipin >> ------------------------------ >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://confluence.freeswitch.org >> >> http://www.cluecon.com >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> >> ______________________________________________________________________ >> This message has been checked for all known viruses by MessageLabs Virus >> Scanning Service. >> ______________________________________________________________________ >> >> Shaun Stokes - Infrastructure Analyst >> >> >> T : >> >> 01453 700713 >> >> E : >> >> shaun.stokes at itec-support.co.uk >> >> W : >> >> www.itec-support.co.uk >> >> Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road, >> Stroud, Gloucestershire GL5 3QF >> Company No. 06908001 >> >> >> CONFIDENTIALITY NOTICE >> This communication and the information it contains are intended for the >> person or organisation to which it is addressed. Its contents are >> confidential and may be protected in law. Unauthorised use, copying or >> disclosure of any of it may be unlawful. If you are not the intended >> recipient, please contact us immediately. >> The contents of any attachments in this e-mail may contain software >> viruses, which could damage your own computer system. While ITEC Support >> has taken every reasonable precaution to minimise this risk, we cannot >> accept liability for any damage which you sustain as a result of software >> viruses. You should carry out your own virus checking procedure before >> opening any attachment. >> >> >> ______________________________________________________________________ >> This message has been checked for all known viruses by MessageLabs Virus >> Scanning Service. >> ______________________________________________________________________ >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> ______________________________________________________________________ >> This message has been checked for all known viruses by MessageLabs Virus >> Scanning Service. >> ______________________________________________________________________ >> >> ______________________________________________________________________ >> This message has been checked for all known viruses by MessageLabs Virus >> Scanning Service. >> ______________________________________________________________________ >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting >> Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >> >> Official FreeSWITCH >> Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com >> >> FreeSWITCH-users mailing >> listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Gregor Nanger > > *CTO* > t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 > • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia > • www.infomedia.si > > > > ---------- > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From rick at magicmail.mooo.com Mon Sep 18 15:35:36 2017 From: rick at magicmail.mooo.com (Rick Jarvis) Date: Mon, 18 Sep 2017 16:35:36 +0100 Subject: [Freeswitch-users] Verto In-Reply-To: References: <03E75515-780B-451B-A30F-62C99020D8A7@magicmail.mooo.com> Message-ID: <8B3FB66F-E6AA-4B3A-A802-6CF79A24D1CE@magicmail.mooo.com> Italo, do you have a ‘complete code’ example for the really helpful verto example at evoluxbr.github.io/verto-docs ? I just can’t seem to get all the code in the right place, so that it all hangs together correctly. I’m also trying to send commands via Verto but not getting anywhere > On 7 Sep 2017, at 21:11, Ítalo Rossi wrote: > > Verto communicator is a example implementation on what can be done with Verto. > > Check our tutorial to understand how to build a minimum app: > > https://evoluxbr.github.io/verto-docs/ > Em qui, 7 de set de 2017 às 17:02, Rick Jarvis > escreveu: > Looking to get into Verto, in particular handling voice calls with JS. Going through the source, I’m wondering: > > 1. What’s the difference between the Verto source and the Verto Communicator source? > > 2. What’s the best way to start from the bottom up - by this I mean that it seems hugely comprehensive, but rather than just use grunt to set it all up, I’d like to start simply with the basics… is there for instance a list of the bare minimum scripts / file structure to use? Apologies if this is a silly question, I’m still relatively new to JS and I don’t want to blow my mind in one go ;) > > Rick > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From italo at freeswitch.org Mon Sep 18 15:48:19 2017 From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=) Date: Mon, 18 Sep 2017 15:48:19 +0000 Subject: [Freeswitch-users] Verto In-Reply-To: <8B3FB66F-E6AA-4B3A-A802-6CF79A24D1CE@magicmail.mooo.com> References: <03E75515-780B-451B-A30F-62C99020D8A7@magicmail.mooo.com> <8B3FB66F-E6AA-4B3A-A802-6CF79A24D1CE@magicmail.mooo.com> Message-ID: I don't think so, but you can always pastebin what you have so we can take a look Em seg, 18 de set de 2017 às 12:37, Rick Jarvis escreveu: > Italo, do you have a ‘complete code’ example for the really helpful verto > example at evoluxbr.github.io/verto-docs ? I just can’t seem to get all > the code in the right place, so that it all hangs together correctly. I’m > also trying to send commands via Verto but not getting anywhere > > On 7 Sep 2017, at 21:11, Ítalo Rossi wrote: > > Verto communicator is a example implementation on what can be done with > Verto. > > Check our tutorial to understand how to build a minimum app: > > https://evoluxbr.github.io/verto-docs/ > Em qui, 7 de set de 2017 às 17:02, Rick Jarvis > escreveu: > >> Looking to get into Verto, in particular handling voice calls with JS. >> Going through the source, I’m wondering: >> >> 1. What’s the difference between the Verto source and the Verto >> Communicator source? >> >> 2. What’s the best way to start from the bottom up - by this I mean that >> it seems hugely comprehensive, but rather than just use grunt to set it all >> up, I’d like to start simply with the basics… is there for instance a list >> of the bare minimum scripts / file structure to use? Apologies if this is a >> silly question, I’m still relatively new to JS and I don’t want to blow my >> mind in one go ;) >> >> Rick >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From rick at magicmail.mooo.com Mon Sep 18 15:58:04 2017 From: rick at magicmail.mooo.com (Rick Jarvis) Date: Mon, 18 Sep 2017 16:58:04 +0100 Subject: [Freeswitch-users] Verto In-Reply-To: References: <03E75515-780B-451B-A30F-62C99020D8A7@magicmail.mooo.com> <8B3FB66F-E6AA-4B3A-A802-6CF79A24D1CE@magicmail.mooo.com> Message-ID: <1B47A679-7BF9-4F11-9A73-462746931C0F@magicmail.mooo.com> Thanks Italo - here it is: https://pastebin.com/pv6DdPkR > On 18 Sep 2017, at 16:48, Ítalo Rossi wrote: > > I don't think so, but you can always pastebin what you have so we can take a look > Em seg, 18 de set de 2017 às 12:37, Rick Jarvis > escreveu: > Italo, do you have a ‘complete code’ example for the really helpful verto example at evoluxbr.github.io/verto-docs ? I just can’t seem to get all the code in the right place, so that it all hangs together correctly. I’m also trying to send commands via Verto but not getting anywhere > > >> On 7 Sep 2017, at 21:11, Ítalo Rossi > wrote: >> > >> Verto communicator is a example implementation on what can be done with Verto. >> >> Check our tutorial to understand how to build a minimum app: >> >> https://evoluxbr.github.io/verto-docs/ >> Em qui, 7 de set de 2017 às 17:02, Rick Jarvis > escreveu: >> Looking to get into Verto, in particular handling voice calls with JS. Going through the source, I’m wondering: >> >> 1. What’s the difference between the Verto source and the Verto Communicator source? >> >> 2. What’s the best way to start from the bottom up - by this I mean that it seems hugely comprehensive, but rather than just use grunt to set it all up, I’d like to start simply with the basics… is there for instance a list of the bare minimum scripts / file structure to use? Apologies if this is a silly question, I’m still relatively new to JS and I don’t want to blow my mind in one go ;) >> >> Rick >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From tculjaga at gmail.com Mon Sep 18 20:37:58 2017 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Mon, 18 Sep 2017 22:37:58 +0200 Subject: [Freeswitch-users] How to record User Voice and listen for the DTMF simultaneously In-Reply-To: References: Message-ID: on ESL i do it like this for calls i bridge... im positive you can find your way to do it as well. "api uuid_setvar d8a8589d-f16a-4caa-b079-f45d73d838b3 bridge_pre_execute_aleg_app record_session" "api uuid_setvar d8a8589d-f16a-4caa-b079-f45d73d838b3 bridge_pre_execute_aleg_arg /home/nexios/recordings/2/queue_recordings/2017-06-12.10.02.13-385996863242.wav" "api uuid_setvar d8a8589d-f16a-4caa-b079-f45d73d838b3 bridge_pre_execute_aleg_data /home/nexios/recordings/2/queue_recordings/2017-06-12.10.02.13-385996863242.wav" "bgapi uuid_bridge d8a8589d-f16a-4caa-b079-f45d73d838b3 33d2d6f5-b88c-45c3-8d2a-187366b2bb0b" On 18 September 2017 at 10:08, Giovanni Maruzzelli wrote: > You may use the set-meta* apps. > > Check local extension in demo configuration > > sent from mobile > cell: +39 347 266 56 18 > Giovanni Maruzzelli > OpenTelecom.IT > > On Sep 18, 2017 09:12, "Deepika Yadav" wrote: > >> Playandgetdigits() - plays a file and gets digits at the same time. >> However I need to initiate recording of the user voice and get the digits >> for quitting >> >> >> >> On Mon, Sep 18, 2017 at 12:02 PM, Giovanni Maruzzelli >> wrote: >> >>> Playandgetdigits() >>> >>> sent from mobile >>> cell: +39 347 266 56 18 >>> Giovanni Maruzzelli >>> OpenTelecom.IT >>> >>> On Sep 17, 2017 11:28, "Deepika Yadav" wrote: >>> >>>> Hi, >>>> >>>> I want to give users an option to record their voice by presssing some >>>> digit in the IVR menu through my python script. I use the code as follows: >>>> >>>> session.recordFile(filename, 120, 500, 3) >>>> >>>> However, this code will block for 2 minutes and then proceed. In case a >>>> user has little to say and after 1 minute wants to stop the recording, how >>>> I can do that. Is there any function similar to "session.playAndGetDigits" >>>> to also hear for DTMFs at the same time? >>>> >>>> >>>> -- >>>> Regards, >>>> Deepika >>>> https://deepikay.wixsite.com/deepika >>>> >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>> switch-users >>>> http://www.freeswitch.org >>>> >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Regards, >> Deepika >> https://deepikay.wixsite.com/deepika >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From ynasida at gmail.com Tue Sep 19 08:07:54 2017 From: ynasida at gmail.com (Yuriy Nasida) Date: Tue, 19 Sep 2017 11:07:54 +0300 Subject: [Freeswitch-users] vm_cc weird behavior In-Reply-To: References: Message-ID: Updated to 1.6.19-36 Same issue. On 18 September 2017 at 10:03, Yuriy Nasida wrote: > Update. > > It isn't just a lot of emails. FS got segfault afte 69th attampt... > > Sep 18 02:59:07 fs1 kernel: [15421422.922960] freeswitch[43697]: segfault > at 7f30e35249b8 ip 00007f313e6c3ef3 sp 00007f30e35249a0 error 6 in > libcurl.so.4.3.0[7f313e69d000+70000] > > > On 16 September 2017 at 18:46, Yuriy Nasida wrote: > >> Any advice? >> >> 14 Сен 2017 г. 16:00 пользователь "Yuriy Nasida" >> написал: >> >> Hi, >>> >>> I noted that FS sends a lot of emails to my email box instead of only >>> one in case I use vm_cc. >>> >>> It's very simple configuration. I just set vm_cc before mod_voicemail >>> is called (according wiki page) and leave voicemail message. >>> For main emails box I recieve only one message as expected. >>> But I have ~60 exactly same messages at email box from vm_cc. >>> >>> In console I also noted that FS really sends emails many times without >>> obvios reason. >>> >>> mod_voicemail.c:2832 Deliver VM to yuriytest2 at mydomain.com >>> mod_voicemail.c:1925 Update MWI: Processing for yuriytest2 at mydomain.com >>> in inbox >>> mod_voicemail.c:1950 Update MWI: Messages Waiting yes >>> mod_voicemail.c:1951 Update MWI: Update Reason NEW >>> mod_voicemail.c:1952 Update MWI: Message Account yuriytest2 at mydomain.com >>> mod_voicemail.c:1953 Update MWI: Voice Message 69/0 >>> switch_utils.c:1180 Emailed file [/tmp/mail.1505386894e5a8] to [ >>> myemailbox at gmail.com] >>> mod_voicemail.c:3042 Sending message to myemailbox at gmail.com >>> >>> Next FS does this again and tries to Deliver VM to >>> yuriytest2 at mydomain.com. >>> >>> Please advice. >>> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From rick at magicmail.mooo.com Tue Sep 19 09:38:16 2017 From: rick at magicmail.mooo.com (Rick Jarvis) Date: Tue, 19 Sep 2017 10:38:16 +0100 Subject: [Freeswitch-users] Verto In-Reply-To: <1B47A679-7BF9-4F11-9A73-462746931C0F@magicmail.mooo.com> References: <03E75515-780B-451B-A30F-62C99020D8A7@magicmail.mooo.com> <8B3FB66F-E6AA-4B3A-A802-6CF79A24D1CE@magicmail.mooo.com> <1B47A679-7BF9-4F11-9A73-462746931C0F@magicmail.mooo.com> Message-ID: I’ve spotted that the live array (which is what I’m currently having problems with) is already in a function expression (initLiveArray). so I’ve taken it out of the function declaration (subscribeArray) that I had put it in before I spotted this. I’ve also moved it out of the main function. So I’m calling the function expression with: initLiveArray(); but it’s looking for three arguments, and properties thereof, so I’m not really clear on how I should be initialising it... > On 18 Sep 2017, at 16:58, Rick Jarvis wrote: > > Thanks Italo - here it is: > > https://pastebin.com/pv6DdPkR > > >> On 18 Sep 2017, at 16:48, Ítalo Rossi > wrote: >> >> I don't think so, but you can always pastebin what you have so we can take a look >> Em seg, 18 de set de 2017 às 12:37, Rick Jarvis > escreveu: >> Italo, do you have a ‘complete code’ example for the really helpful verto example at evoluxbr.github.io/verto-docs ? I just can’t seem to get all the code in the right place, so that it all hangs together correctly. I’m also trying to send commands via Verto but not getting anywhere >> >> >>> On 7 Sep 2017, at 21:11, Ítalo Rossi > wrote: >>> >> >>> Verto communicator is a example implementation on what can be done with Verto. >>> >>> Check our tutorial to understand how to build a minimum app: >>> >>> https://evoluxbr.github.io/verto-docs/ >>> Em qui, 7 de set de 2017 às 17:02, Rick Jarvis > escreveu: >>> Looking to get into Verto, in particular handling voice calls with JS. Going through the source, I’m wondering: >>> >>> 1. What’s the difference between the Verto source and the Verto Communicator source? >>> >>> 2. What’s the best way to start from the bottom up - by this I mean that it seems hugely comprehensive, but rather than just use grunt to set it all up, I’d like to start simply with the basics… is there for instance a list of the bare minimum scripts / file structure to use? Apologies if this is a silly question, I’m still relatively new to JS and I don’t want to blow my mind in one go ;) >>> >>> Rick >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From infos at madovsky.org Tue Sep 19 10:07:04 2017 From: infos at madovsky.org (Madovsky) Date: Tue, 19 Sep 2017 03:07:04 -0700 Subject: [Freeswitch-users] rtmp and user registrations Message-ID: <19f68aea-af68-80d8-852a-3f7d5ef5bbd6@madovsky.org> Hi everybody, is mod_rtmp registered users can be managed like sip profile with a postgresql table like rtmp_registrations? thanks Franck From rick at magicmail.mooo.com Tue Sep 19 13:56:53 2017 From: rick at magicmail.mooo.com (Rick Jarvis) Date: Tue, 19 Sep 2017 14:56:53 +0100 Subject: [Freeswitch-users] Verto In-Reply-To: References: <03E75515-780B-451B-A30F-62C99020D8A7@magicmail.mooo.com> <8B3FB66F-E6AA-4B3A-A802-6CF79A24D1CE@magicmail.mooo.com> <1B47A679-7BF9-4F11-9A73-462746931C0F@magicmail.mooo.com> Message-ID: <839B080D-BD32-4F51-BAB6-E82CF2EC781A@magicmail.mooo.com> Still battling with this, trying to get anything out of the livearray. Any help would be really appreciated, I really want to get this working and I don’t know what it is I’m just not getting :( My understanding is that this stripped down code should at least join me to the conference 00001 and subscribe me to updates: https://pastebin.com/bphHdxGm I don’t want to sound desperate, but I am, embarrassingly so ;) > On 19 Sep 2017, at 10:38, Rick Jarvis wrote: > > I’ve spotted that the live array (which is what I’m currently having problems with) is already in a function expression (initLiveArray). so I’ve taken it out of the function declaration (subscribeArray) that I had put it in before I spotted this. I’ve also moved it out of the main function. > > So I’m calling the function expression with: > > initLiveArray(); > > but it’s looking for three arguments, and properties thereof, so I’m not really clear on how I should be initialising it... > >> On 18 Sep 2017, at 16:58, Rick Jarvis > wrote: >> >> Thanks Italo - here it is: >> >> https://pastebin.com/pv6DdPkR >> >> >>> On 18 Sep 2017, at 16:48, Ítalo Rossi > wrote: >>> >>> I don't think so, but you can always pastebin what you have so we can take a look >>> Em seg, 18 de set de 2017 às 12:37, Rick Jarvis > escreveu: >>> Italo, do you have a ‘complete code’ example for the really helpful verto example at evoluxbr.github.io/verto-docs ? I just can’t seem to get all the code in the right place, so that it all hangs together correctly. I’m also trying to send commands via Verto but not getting anywhere >>> >>> >>>> On 7 Sep 2017, at 21:11, Ítalo Rossi > wrote: >>>> >>> >>>> Verto communicator is a example implementation on what can be done with Verto. >>>> >>>> Check our tutorial to understand how to build a minimum app: >>>> >>>> https://evoluxbr.github.io/verto-docs/ >>>> Em qui, 7 de set de 2017 às 17:02, Rick Jarvis > escreveu: >>>> Looking to get into Verto, in particular handling voice calls with JS. Going through the source, I’m wondering: >>>> >>>> 1. What’s the difference between the Verto source and the Verto Communicator source? >>>> >>>> 2. What’s the best way to start from the bottom up - by this I mean that it seems hugely comprehensive, but rather than just use grunt to set it all up, I’d like to start simply with the basics… is there for instance a list of the bare minimum scripts / file structure to use? Apologies if this is a silly question, I’m still relatively new to JS and I don’t want to blow my mind in one go ;) >>>> >>>> Rick >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From nerad.peter at gmail.com Tue Sep 19 09:45:11 2017 From: nerad.peter at gmail.com (=?iso-8859-2?Q?Peter_Ner=E1d?=) Date: Tue, 19 Sep 2017 11:45:11 +0200 Subject: [Freeswitch-users] mod_callcenter events variables Message-ID: <000c01d3312b$fd1430f0$f73c92d0$@gmail.com> Hi, Im parsing events from mod_callcenter but I have problem with variables like CC-*-Time .. what is meaning of those numbers ? CC-Member-Joined-Time, CC-Member-Leaving-Time .. I have no idea. ( they are not unix timestamps in microseconds ) Sorry for me bad English :-D -------------- next part -------------- An HTML attachment was scrubbed... URL: From freeswitch940 at gmail.com Mon Sep 18 06:16:10 2017 From: freeswitch940 at gmail.com (Freeswitch user) Date: Mon, 18 Sep 2017 11:46:10 +0530 Subject: [Freeswitch-users] JAVA CODE ERROR Message-ID: Error:- [ERR] switch_cpp.cpp:683 object is not initalized import org.freeswitch.*; import org.freeswitch.swig.*; public class PhoneTest implements FreeswitchScript, DTMFCallback, HangupHook { public PhoneTest() { } public String onDTMF(Object object, int i, String arg) { if (object instanceof String) freeswitch.console_log("notice", "DTMF: " + (String)object + " ARG: " + arg + "\n"); else freeswitch.console_log("notice", "WOW GOT AN EVENT: " + object.toString()); return "true"; } public void onHangup() { freeswitch.console_log("notice", "HANGUP!\n"); } public void run(String sessionUuid, String args) { freeswitch.console_log("notice", "UUID: " + sessionUuid + " ARGS: " + args + "\n"); JavaSession session = null; try { session = new JavaSession(sessionUuid); session.setDTMFCallback(this, "TEST"); // Getting error in this line. session object not initilizing session.setHangupHook(this); session.answer(); session.streamFile(args, 0); session.hangup(""); } finally { if (session != null) session.delete(); } } } -------------- next part -------------- An HTML attachment was scrubbed... URL: From Bjorn.Bylander at loxysoft.com Tue Sep 19 12:06:13 2017 From: Bjorn.Bylander at loxysoft.com (=?iso-8859-1?Q?Bj=F6rn_Bylander?=) Date: Tue, 19 Sep 2017 12:06:13 +0000 Subject: [Freeswitch-users] Bug? Recording pauses when the recorded leg is held Message-ID: Hello, I've got an application using mod_erlang_event which does the following: 1. Originates a call to A and waits for A to answer. 2. Parks A. 3. Originates a call to B and waits for B to answer. 4. Bridges A to B. 5. Starts recording B using uuid_record (to a file with the "wav" extension). Audio from both A and B is recorded, as expected. 6. Stops recording when either leg is terminated. If B holds its call using a SIP re-INVITE with recvonly in the SDP, FreeSWITCH pauses recording instead of recording just the audio from the A leg. This seems like a bug to me given the documentation for uuid_record (from https://freeswitch.org/confluence/display/FREESWITCH/mod_commands): "The start command causes FreeSWITCH to start mixing all call legs together and saves the result as a file in the format that the file's extension dictates." Am I right in considering this behaviour to be a bug? I'm using FS v1.6.19. Regards, Björn Bylander From akshaya18j at gmail.com Mon Sep 18 07:31:36 2017 From: akshaya18j at gmail.com (Akshaya j) Date: Mon, 18 Sep 2017 13:01:36 +0530 Subject: [Freeswitch-users] Dynamic Conference Calls Message-ID: Hello FS users, I'm a fresher working around with FreeSwitch features and came across this conference room feature. This conference room actually acts like a rooms where anyone can join the room by dialling some number. There is a also moderator who can add any user into the conference room. What i need to know is whether the conventional conference calls is available in FreeSwitch? The conventional conference calls is what we use in our normal mobile phones where any person in a call can add any other person by dialling his number and then all 3 persons can talk together. That is anybody in a call can act as a moderator to add a person to join. I don't find any content regarding this conference in web, is there any other terminology for this feature in Freeswitch which i dont see or whether this is possible? If so how to proceed with this ?? Any help would be highly appreciated.!!! Thanks Advance, PurpleWinged -------------- next part -------------- An HTML attachment was scrubbed... URL: From 11666by at gmail.com Tue Sep 19 10:57:29 2017 From: 11666by at gmail.com (=?UTF-8?B?0JzQsNC60LDRgNC10LLQuNGHINCcLtCQLg==?=) Date: Tue, 19 Sep 2017 13:57:29 +0300 Subject: [Freeswitch-users] install source error In-Reply-To: References: Message-ID: <24f6a772-9ca9-c096-a368-f9a2d5709415@gmail.com> Hi, I am installing freeswitch 1.6.19 with RPI-3,but getting some errors Linux raspberrypi 4.9.41-v7+ #1023 SMP Tue Aug 8 16:00:15 BST 2017 armv7l GNU/Linux. pi at raspberrypi:~/freeswitch-1.6.19 $ make -j3 touch /home/pi/freeswitch-1.6.19/src/include/switch.h CC libfreeswitch_spandsp_la-plc.lo make[1]: Entering directory '/home/pi/freeswitch-1.6.19/libs/srtp' /bin/bash ./libtool --tag=CC --mode=compile gcc -DHAVE_CONFIG_H -I. -I./crypto/include -I./src -Icrypto/include -I./include -I./crypto/include -I./src -Icrypto/include -I./include -I./crypto/include -g -O2 -Wall -std=c99 -pedantic -MT srtp.lo -MD -MP -MF .deps/srtp.Tpo -c -o srtp.lo `test -f 'srtp/srtp.c' || echo './'`srtp/srtp.c libtool: compile: gcc -DHAVE_CONFIG_H -I. -I./crypto/include -I./src -Icrypto/include -I./include -I./crypto/include -I./src -Icrypto/include -I./include -I./crypto/include -g -O2 -Wall -std=c99 -pedantic -MT srtp.lo -MD -MP -MF .deps/srtp.Tpo -c srtp/srtp.c -fPIC -DPIC -o srtp.o In file included from ./crypto/include/prng.h:17:0, from ./crypto/include/crypto_kernel.h:50, from ./include/srtp.h:53, from srtp/srtp.c:46: ./crypto/include/aes_icm_ossl.h:66:20: error: field ‘ctx’ has incomplete type EVP_CIPHER_CTX ctx; ^~~ In file included from srtp/srtp.c:50:0: ./crypto/include/aes_gcm_ossl.h:58:18: error: field ‘ctx’ has incomplete type EVP_CIPHER_CTX ctx; ^~~ config.status: creating Makefile config.status: creating include/apr.h config.status: creating build/apr_rules.mk Makefile:513: recipe for target 'srtp.lo' failed make[1]: *** [srtp.lo] Error 1 Please suggest me. From 11666by at gmail.com Tue Sep 19 10:00:24 2017 From: 11666by at gmail.com (=?UTF-8?B?0JzQsNC60LDRgNC10LLQuNGHINCcLtCQLg==?=) Date: Tue, 19 Sep 2017 13:00:24 +0300 Subject: [Freeswitch-users] install error Message-ID: <37cbd080-8bc6-1737-3557-8b20e15b5903@gmail.com> I am installing freeswitch 1.6.19 with RPI-3,but getting some errors. Linux raspberrypi 4.9.41-v7+ #1023 SMP Tue Aug 8 16:00:15 BST 2017 armv7l GNU/Linux pi at raspberrypi:~/freeswitch-1.6.19 $ make -j3 touch /home/pi/freeswitch-1.6.19/src/include/switch.h CC libfreeswitch_spandsp_la-plc.lo make[1]: Entering directory '/home/pi/freeswitch-1.6.19/libs/srtp' /bin/bash ./libtool --tag=CC --mode=compile gcc -DHAVE_CONFIG_H -I. -I./crypto/include -I./src -Icrypto/include -I./include -I./crypto/include -I./src -Icrypto/include -I./include -I./crypto/include -g -O2 -Wall -std=c99 -pedantic -MT srtp.lo -MD -MP -MF .deps/srtp.Tpo -c -o srtp.lo `test -f 'srtp/srtp.c' || echo './'`srtp/srtp.c libtool: compile: gcc -DHAVE_CONFIG_H -I. -I./crypto/include -I./src -Icrypto/include -I./include -I./crypto/include -I./src -Icrypto/include -I./include -I./crypto/include -g -O2 -Wall -std=c99 -pedantic -MT srtp.lo -MD -MP -MF .deps/srtp.Tpo -c srtp/srtp.c -fPIC -DPIC -o srtp.o In file included from ./crypto/include/prng.h:17:0, from ./crypto/include/crypto_kernel.h:50, from ./include/srtp.h:53, from srtp/srtp.c:46: ./crypto/include/aes_icm_ossl.h:66:20: error: field ‘ctx’ has incomplete type EVP_CIPHER_CTX ctx; ^~~ In file included from srtp/srtp.c:50:0: ./crypto/include/aes_gcm_ossl.h:58:18: error: field ‘ctx’ has incomplete type EVP_CIPHER_CTX ctx; ^~~ config.status: creating Makefile config.status: creating include/apr.h config.status: creating build/apr_rules.mk Makefile:513: recipe for target 'srtp.lo' failed make[1]: *** [srtp.lo] Error 1 make[1]: Leaving directory '/home/pi/freeswitch-1.6.19/libs/srtp' Makefile:3719: recipe for target 'libs/srtp/libsrtp.la' failed make: *** [libs/srtp/libsrtp.la] Error 2 Please suggest me. From freeswitch940 at gmail.com Tue Sep 19 15:48:53 2017 From: freeswitch940 at gmail.com (Freeswitch user) Date: Tue, 19 Sep 2017 15:48:53 +0000 Subject: [Freeswitch-users] Dynamic Conference Calls In-Reply-To: References: Message-ID: Use mod_conference module for setup a conference... On Tue, 19 Sep 2017 at 9:11 PM, Akshaya j wrote: > Hello FS users, > I'm a fresher working around with FreeSwitch features and came across > this conference room feature. This conference room actually acts like a > rooms where anyone can join the room by dialling some number. There is a > also moderator who can add any user into the conference room. > What i need to know is whether the conventional conference calls is > available in FreeSwitch? The conventional conference calls is what we use > in our normal mobile phones where any person in a call can add any other > person by dialling his number and then all 3 persons can talk together. > That is anybody in a call can act as a moderator to add a person to join. > I don't find any content regarding this conference in web, is > there any other terminology for this feature in Freeswitch which i dont see > or whether this is possible? If so how to proceed with this ?? > Any help would be highly appreciated.!!! > > Thanks Advance, > PurpleWinged > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Tue Sep 19 15:49:16 2017 From: mike at jerris.com (Michael Jerris) Date: Tue, 19 Sep 2017 11:49:16 -0400 Subject: [Freeswitch-users] install source error In-Reply-To: <24f6a772-9ca9-c096-a368-f9a2d5709415@gmail.com> References: <24f6a772-9ca9-c096-a368-f9a2d5709415@gmail.com> Message-ID: <4E0A221A-40A2-4CAA-A45D-A7E45D8C7D8B@jerris.com> building 1.6 against newer openssl? newer openssl requires master. > On Sep 19, 2017, at 6:57 AM, Макаревич М.А. <11666by at gmail.com> wrote: > > > > > > > > > > Hi, > > I am installing freeswitch 1.6.19 with RPI-3,but getting some errors > > Linux raspberrypi 4.9.41-v7+ #1023 SMP Tue Aug 8 16:00:15 BST 2017 > armv7l GNU/Linux. > > > pi at raspberrypi:~/freeswitch-1.6.19 $ make -j3 > touch /home/pi/freeswitch-1.6.19/src/include/switch.h > CC libfreeswitch_spandsp_la-plc.lo > make[1]: Entering directory '/home/pi/freeswitch-1.6.19/libs/srtp' > /bin/bash ./libtool --tag=CC --mode=compile gcc -DHAVE_CONFIG_H -I. > -I./crypto/include -I./src -Icrypto/include -I./include > -I./crypto/include -I./src -Icrypto/include -I./include > -I./crypto/include -g -O2 -Wall -std=c99 -pedantic -MT srtp.lo -MD -MP > -MF .deps/srtp.Tpo -c -o srtp.lo `test -f 'srtp/srtp.c' || echo > './'`srtp/srtp.c > libtool: compile: gcc -DHAVE_CONFIG_H -I. -I./crypto/include -I./src > -Icrypto/include -I./include -I./crypto/include -I./src -Icrypto/include > -I./include -I./crypto/include -g -O2 -Wall -std=c99 -pedantic -MT > srtp.lo -MD -MP -MF .deps/srtp.Tpo -c srtp/srtp.c -fPIC -DPIC -o srtp.o > In file included from ./crypto/include/prng.h:17:0, > from ./crypto/include/crypto_kernel.h:50, > from ./include/srtp.h:53, > from srtp/srtp.c:46: > ./crypto/include/aes_icm_ossl.h:66:20: error: field ‘ctx’ has incomplete > type > EVP_CIPHER_CTX ctx; > ^~~ > In file included from srtp/srtp.c:50:0: > ./crypto/include/aes_gcm_ossl.h:58:18: error: field ‘ctx’ has incomplete > type > EVP_CIPHER_CTX ctx; > ^~~ > config.status: creating Makefile > config.status: creating include/apr.h > config.status: creating build/apr_rules.mk > Makefile:513: recipe for target 'srtp.lo' failed > make[1]: *** [srtp.lo] Error 1 > > > Please suggest me. From anthony.minessale at gmail.com Tue Sep 19 16:11:05 2017 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 19 Sep 2017 11:11:05 -0500 Subject: [Freeswitch-users] JAVA CODE ERROR In-Reply-To: References: Message-ID: sessionUuid is probably not a valid existing call uuid, kid of a rude post BTW On Mon, Sep 18, 2017 at 1:16 AM, Freeswitch user wrote: > > Error:- [ERR] switch_cpp.cpp:683 object is not initalized > > import org.freeswitch.*; > import org.freeswitch.swig.*; > > public class PhoneTest implements FreeswitchScript, DTMFCallback, > HangupHook > { > public PhoneTest() > { > } > > public String onDTMF(Object object, int i, String arg) > { > if (object instanceof String) > freeswitch.console_log("notice", "DTMF: " + (String)object + " > ARG: " + arg + "\n"); > else > freeswitch.console_log("notice", "WOW GOT AN EVENT: " + > object.toString()); > return "true"; > } > > public void onHangup() > { > freeswitch.console_log("notice", "HANGUP!\n"); > } > > public void run(String sessionUuid, String args) > { > freeswitch.console_log("notice", "UUID: " + sessionUuid + " ARGS: > " + args + "\n"); > JavaSession session = null; > try > { > session = new JavaSession(sessionUuid); > session.setDTMFCallback(this, "TEST"); // Getting error > in this line. session object not initilizing > session.setHangupHook(this); > > session.answer(); > session.streamFile(args, 0); > session.hangup(""); > } > finally > { > if (session != null) > session.delete(); > } > } > } > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ * ClueCon Weekly Development Call ☎ sip:888 at conference.freeswitch.org ☎ +19193869900 https://www.youtube.com/watch?v=oAxXgyx5jUw https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: From chad at apartmentlines.com Tue Sep 19 16:14:28 2017 From: chad at apartmentlines.com (Chad Phillips) Date: Tue, 19 Sep 2017 09:14:28 -0700 Subject: [Freeswitch-users] Verto In-Reply-To: <839B080D-BD32-4F51-BAB6-E82CF2EC781A@magicmail.mooo.com> References: <03E75515-780B-451B-A30F-62C99020D8A7@magicmail.mooo.com> <8B3FB66F-E6AA-4B3A-A802-6CF79A24D1CE@magicmail.mooo.com> <1B47A679-7BF9-4F11-9A73-462746931C0F@magicmail.mooo.com> <839B080D-BD32-4F51-BAB6-E82CF2EC781A@magicmail.mooo.com> Message-ID: Perhaps my ClueCon 2016 talk slides and example code will help you: https://gist.github.com/thehunmonkgroup/446370910266f006cdcf25df5e28df7b A lot of that got integrated into http://evoluxbr.github.io/verto-docs/, but it might provide some additional insight. On Tue, Sep 19, 2017 at 6:56 AM, Rick Jarvis wrote: > Still battling with this, trying to get anything out of the livearray. Any > help would be really appreciated, I really want to get this working and I > don’t know what it is I’m just not getting :( > > My understanding is that this stripped down code should at least join me > to the conference 00001 and subscribe me to updates: https://pastebin.com/ > bphHdxGm > > I don’t want to sound desperate, but I am, embarrassingly so ;) > > On 19 Sep 2017, at 10:38, Rick Jarvis wrote: > > I’ve spotted that the live array (which is what I’m currently having > problems with) is already in a function expression (initLiveArray). so I’ve > taken it out of the function declaration (subscribeArray) that I had put it > in before I spotted this. I’ve also moved it out of the main function. > > So I’m calling the function expression with: > > initLiveArray(); > > but it’s looking for three arguments, and properties thereof, so I’m not > really clear on how I should be initialising it... > > On 18 Sep 2017, at 16:58, Rick Jarvis wrote: > > Thanks Italo - here it is: > > https://pastebin.com/pv6DdPkR > > > On 18 Sep 2017, at 16:48, Ítalo Rossi wrote: > > I don't think so, but you can always pastebin what you have so we can take > a look > Em seg, 18 de set de 2017 às 12:37, Rick Jarvis > escreveu: > >> Italo, do you have a ‘complete code’ example for the really helpful verto >> example at evoluxbr.github.io/verto-docs ? I just can’t seem to get all >> the code in the right place, so that it all hangs together correctly. I’m >> also trying to send commands via Verto but not getting anywhere >> >> On 7 Sep 2017, at 21:11, Ítalo Rossi wrote: >> >> Verto communicator is a example implementation on what can be done with >> Verto. >> >> Check our tutorial to understand how to build a minimum app: >> >> https://evoluxbr.github.io/verto-docs/ >> Em qui, 7 de set de 2017 às 17:02, Rick Jarvis >> escreveu: >> >>> Looking to get into Verto, in particular handling voice calls with JS. >>> Going through the source, I’m wondering: >>> >>> 1. What’s the difference between the Verto source and the Verto >>> Communicator source? >>> >>> 2. What’s the best way to start from the bottom up - by this I mean that >>> it seems hugely comprehensive, but rather than just use grunt to set it all >>> up, I’d like to start simply with the basics… is there for instance a list >>> of the bare minimum scripts / file structure to use? Apologies if this is a >>> silly question, I’m still relatively new to JS and I don’t want to blow my >>> mind in one go ;) >>> >>> Rick >>> >>> >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From freeswitch940 at gmail.com Tue Sep 19 16:19:24 2017 From: freeswitch940 at gmail.com (Freeswitch user) Date: Tue, 19 Sep 2017 16:19:24 +0000 Subject: [Freeswitch-users] JAVA CODE ERROR In-Reply-To: References: Message-ID: Thanks for reply.Can you please suggest how can setup session object in java and get channel variables.? On Tue, 19 Sep 2017 at 9:41 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > sessionUuid is probably not a valid existing call uuid, kid of a rude post > BTW > > On Mon, Sep 18, 2017 at 1:16 AM, Freeswitch user > wrote: > >> >> Error:- [ERR] switch_cpp.cpp:683 object is not initalized >> >> import org.freeswitch.*; >> import org.freeswitch.swig.*; >> >> public class PhoneTest implements FreeswitchScript, DTMFCallback, >> HangupHook >> { >> public PhoneTest() >> { >> } >> >> public String onDTMF(Object object, int i, String arg) >> { >> if (object instanceof String) >> freeswitch.console_log("notice", "DTMF: " + (String)object + " >> ARG: " + arg + "\n"); >> else >> freeswitch.console_log("notice", "WOW GOT AN EVENT: " + >> object.toString()); >> return "true"; >> } >> >> public void onHangup() >> { >> freeswitch.console_log("notice", "HANGUP!\n"); >> } >> >> public void run(String sessionUuid, String args) >> { >> freeswitch.console_log("notice", "UUID: " + sessionUuid + " >> ARGS: " + args + "\n"); >> JavaSession session = null; >> try >> { >> session = new JavaSession(sessionUuid); >> session.setDTMFCallback(this, "TEST"); // Getting error >> in this line. session object not initilizing >> session.setHangupHook(this); >> >> session.answer(); >> session.streamFile(args, 0); >> session.hangup(""); >> } >> finally >> { >> if (session != null) >> session.delete(); >> } >> } >> } >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ > > ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ > http://twitter.com/FreeSWITCH > ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ > * > > ClueCon Weekly Development Call > ☎ sip:888 at conference.freeswitch.org ☎ +19193869900 > > https://www.youtube.com/watch?v=oAxXgyx5jUw > https://www.youtube.com/watch?v=9XXgW34t40s > https://www.youtube.com/watch?v=NLaDpGQuZDA > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From tahir at ictinnovations.com Tue Sep 19 16:33:43 2017 From: tahir at ictinnovations.com (Tahir Almas) Date: Tue, 19 Sep 2017 09:33:43 -0700 Subject: [Freeswitch-users] FreeSWITCH on Raspberry Pi In-Reply-To: References: <59B7F678.1050403@telefaks.de> Message-ID: I will be interested to know how many g.711 concurrent calls , the latest respberri pi can support *Tahir Almas* ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT On Wed, Sep 13, 2017 at 1:35 AM, Tihomir Culjaga wrote: > if you need to do a lot of writes, sdcard is not a good option :=) > > this is why most embedded systems use RO fs and they do write something > only when needed. > > I tried this one: http://khadas.com/vim/ and i can say its really good. > > As for RPi i was able to compile FS from source without no issues. Of > course you need to reconfigure your OS (i used centos ) properly. I moved > FS database on a ram disk and i can say it works really nice. > FS logging is disabled.. i enable it only if needed when i debug somethig > > T. > > On 13 September 2017 at 00:28, Stanislav Sinyagin > wrote: > >> >> >> On 12 Sep 2017 22:13, "jungle Boogie" wrote: >> >> On 12 September 2017 at 12:50, Stanislav Sinyagin >> wrote: >> > Alix is too old. The company has a much better board already: >> > http://pcengines.ch/apu2.htm >> > It can accommodate a real SSD with a much longer life cycle. >> > Also I made a debian installer for it: >> > https://github.com/ssinyagin/pcengines-apu-debian-cd >> > >> > in the ARM world, here's a box that houses a 2.5" disk: >> > http://www.friendlyarm.com/index.php?route=product/product&p >> roduct_id=192 >> > >> >> This looks neat. Do you think it would be faster than an SD card on a >> raspberry pi3? >> >> >> I guess so, although it's SATA over USB 2.0 >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From anthony.minessale at gmail.com Tue Sep 19 16:56:25 2017 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 19 Sep 2017 11:56:25 -0500 Subject: [Freeswitch-users] Verto In-Reply-To: References: <03E75515-780B-451B-A30F-62C99020D8A7@magicmail.mooo.com> <8B3FB66F-E6AA-4B3A-A802-6CF79A24D1CE@magicmail.mooo.com> <1B47A679-7BF9-4F11-9A73-462746931C0F@magicmail.mooo.com> <839B080D-BD32-4F51-BAB6-E82CF2EC781A@magicmail.mooo.com> Message-ID: Easy Button would be: https://freeswitch.org/pages/vm.html On Tue, Sep 19, 2017 at 11:14 AM, Chad Phillips wrote: > Perhaps my ClueCon 2016 talk slides and example code will help you: > https://gist.github.com/thehunmonkgroup/446370910266f006cdcf25df5e28df7b > > A lot of that got integrated into http://evoluxbr.github.io/verto-docs/, > but it might provide some additional insight. > > On Tue, Sep 19, 2017 at 6:56 AM, Rick Jarvis > wrote: > >> Still battling with this, trying to get anything out of the livearray. >> Any help would be really appreciated, I really want to get this working and >> I don’t know what it is I’m just not getting :( >> >> My understanding is that this stripped down code should at least join me >> to the conference 00001 and subscribe me to updates: >> https://pastebin.com/bphHdxGm >> >> I don’t want to sound desperate, but I am, embarrassingly so ;) >> >> On 19 Sep 2017, at 10:38, Rick Jarvis wrote: >> >> I’ve spotted that the live array (which is what I’m currently having >> problems with) is already in a function expression (initLiveArray). so I’ve >> taken it out of the function declaration (subscribeArray) that I had put it >> in before I spotted this. I’ve also moved it out of the main function. >> >> So I’m calling the function expression with: >> >> initLiveArray(); >> >> but it’s looking for three arguments, and properties thereof, so I’m not >> really clear on how I should be initialising it... >> >> On 18 Sep 2017, at 16:58, Rick Jarvis wrote: >> >> Thanks Italo - here it is: >> >> https://pastebin.com/pv6DdPkR >> >> >> On 18 Sep 2017, at 16:48, Ítalo Rossi wrote: >> >> I don't think so, but you can always pastebin what you have so we can >> take a look >> Em seg, 18 de set de 2017 às 12:37, Rick Jarvis >> escreveu: >> >>> Italo, do you have a ‘complete code’ example for the really helpful >>> verto example at evoluxbr.github.io/verto-docs ? I just can’t seem to >>> get all the code in the right place, so that it all hangs together >>> correctly. I’m also trying to send commands via Verto but not getting >>> anywhere >>> >>> On 7 Sep 2017, at 21:11, Ítalo Rossi wrote: >>> >>> Verto communicator is a example implementation on what can be done with >>> Verto. >>> >>> Check our tutorial to understand how to build a minimum app: >>> >>> https://evoluxbr.github.io/verto-docs/ >>> Em qui, 7 de set de 2017 às 17:02, Rick Jarvis >>> escreveu: >>> >>>> Looking to get into Verto, in particular handling voice calls with JS. >>>> Going through the source, I’m wondering: >>>> >>>> 1. What’s the difference between the Verto source and the Verto >>>> Communicator source? >>>> >>>> 2. What’s the best way to start from the bottom up - by this I mean >>>> that it seems hugely comprehensive, but rather than just use grunt to set >>>> it all up, I’d like to start simply with the basics… is there for instance >>>> a list of the bare minimum scripts / file structure to use? Apologies if >>>> this is a silly question, I’m still relatively new to JS and I don’t want >>>> to blow my mind in one go ;) >>>> >>>> Rick >>>> >>>> >>>> >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ >>>> freeswitch-users >>>> http://www.freeswitch.org >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ * ClueCon Weekly Development Call ☎ sip:888 at conference.freeswitch.org ☎ +19193869900 https://www.youtube.com/watch?v=oAxXgyx5jUw https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: From rick at magicmail.mooo.com Tue Sep 19 18:12:57 2017 From: rick at magicmail.mooo.com (Rick Jarvis) Date: Tue, 19 Sep 2017 19:12:57 +0100 Subject: [Freeswitch-users] Verto In-Reply-To: References: <03E75515-780B-451B-A30F-62C99020D8A7@magicmail.mooo.com> <8B3FB66F-E6AA-4B3A-A802-6CF79A24D1CE@magicmail.mooo.com> <1B47A679-7BF9-4F11-9A73-462746931C0F@magicmail.mooo.com> <839B080D-BD32-4F51-BAB6-E82CF2EC781A@magicmail.mooo.com> Message-ID: Progress! I’ve now got messages coming back with conference joins etc! Not sure what I was missing before, but, hey. So now I have two main challenges: 1. How to get a full list of all conference members 2. How to check if there is a current verto session / call in progress, and make the call if not. Thought about doing this through unique IDs and cookies, but I’d guess this is built in already somehow, so don’t want to reinvent the wheel? > On 19 Sep 2017, at 17:14, Chad Phillips wrote: > > Perhaps my ClueCon 2016 talk slides and example code will help you: https://gist.github.com/thehunmonkgroup/446370910266f006cdcf25df5e28df7b > > A lot of that got integrated into http://evoluxbr.github.io/verto-docs/ , but it might provide some additional insight. > > On Tue, Sep 19, 2017 at 6:56 AM, Rick Jarvis > wrote: > Still battling with this, trying to get anything out of the livearray. Any help would be really appreciated, I really want to get this working and I don’t know what it is I’m just not getting :( > > My understanding is that this stripped down code should at least join me to the conference 00001 and subscribe me to updates: https://pastebin.com/bphHdxGm > > I don’t want to sound desperate, but I am, embarrassingly so ;) > >> On 19 Sep 2017, at 10:38, Rick Jarvis > wrote: >> >> I’ve spotted that the live array (which is what I’m currently having problems with) is already in a function expression (initLiveArray). so I’ve taken it out of the function declaration (subscribeArray) that I had put it in before I spotted this. I’ve also moved it out of the main function. >> >> So I’m calling the function expression with: >> >> initLiveArray(); >> >> but it’s looking for three arguments, and properties thereof, so I’m not really clear on how I should be initialising it... >> >>> On 18 Sep 2017, at 16:58, Rick Jarvis > wrote: >>> >>> Thanks Italo - here it is: >>> >>> https://pastebin.com/pv6DdPkR >>> >>> >>>> On 18 Sep 2017, at 16:48, Ítalo Rossi > wrote: >>>> >>>> I don't think so, but you can always pastebin what you have so we can take a look >>>> Em seg, 18 de set de 2017 às 12:37, Rick Jarvis > escreveu: >>>> Italo, do you have a ‘complete code’ example for the really helpful verto example at evoluxbr.github.io/verto-docs ? I just can’t seem to get all the code in the right place, so that it all hangs together correctly. I’m also trying to send commands via Verto but not getting anywhere >>>> >>>> >>>>> On 7 Sep 2017, at 21:11, Ítalo Rossi > wrote: >>>>> >>>> >>>>> Verto communicator is a example implementation on what can be done with Verto. >>>>> >>>>> Check our tutorial to understand how to build a minimum app: >>>>> >>>>> https://evoluxbr.github.io/verto-docs/ >>>>> Em qui, 7 de set de 2017 às 17:02, Rick Jarvis > escreveu: >>>>> Looking to get into Verto, in particular handling voice calls with JS. Going through the source, I’m wondering: >>>>> >>>>> 1. What’s the difference between the Verto source and the Verto Communicator source? >>>>> >>>>> 2. What’s the best way to start from the bottom up - by this I mean that it seems hugely comprehensive, but rather than just use grunt to set it all up, I’d like to start simply with the basics… is there for instance a list of the bare minimum scripts / file structure to use? Apologies if this is a silly question, I’m still relatively new to JS and I don’t want to blow my mind in one go ;) >>>>> >>>>> Rick >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From rick at magicmail.mooo.com Tue Sep 19 19:20:52 2017 From: rick at magicmail.mooo.com (Rick Jarvis) Date: Tue, 19 Sep 2017 20:20:52 +0100 Subject: [Freeswitch-users] Verto In-Reply-To: References: <03E75515-780B-451B-A30F-62C99020D8A7@magicmail.mooo.com> <8B3FB66F-E6AA-4B3A-A802-6CF79A24D1CE@magicmail.mooo.com> <1B47A679-7BF9-4F11-9A73-462746931C0F@magicmail.mooo.com> <839B080D-BD32-4F51-BAB6-E82CF2EC781A@magicmail.mooo.com> Message-ID: <9A41F452-DE2A-48BD-ADEF-F4D4D227E5EF@magicmail.mooo.com> Thanks Anthony, that looks v interesting! Also, having trouble with sending a command, EG if I want to unmute a member. Is the ‘member_id’ the UUID of that call leg? Using the included ‘sendCommand’ function, if I do something like: sendCommand('unmute', 'ef5af1ea-829d-4dbb-98be-16a5f9d3ca94’); It’s not enough args, and in any case I get: {message: "eventChannel not specified.", code: -32002} > On 19 Sep 2017, at 17:56, Anthony Minessale wrote: > > Easy Button would be: > > https://freeswitch.org/pages/vm.html > > > On Tue, Sep 19, 2017 at 11:14 AM, Chad Phillips > wrote: > Perhaps my ClueCon 2016 talk slides and example code will help you: https://gist.github.com/thehunmonkgroup/446370910266f006cdcf25df5e28df7b > > A lot of that got integrated into http://evoluxbr.github.io/verto-docs/ , but it might provide some additional insight. > > On Tue, Sep 19, 2017 at 6:56 AM, Rick Jarvis > wrote: > Still battling with this, trying to get anything out of the livearray. Any help would be really appreciated, I really want to get this working and I don’t know what it is I’m just not getting :( > > My understanding is that this stripped down code should at least join me to the conference 00001 and subscribe me to updates: https://pastebin.com/bphHdxGm > > I don’t want to sound desperate, but I am, embarrassingly so ;) > >> On 19 Sep 2017, at 10:38, Rick Jarvis > wrote: >> >> I’ve spotted that the live array (which is what I’m currently having problems with) is already in a function expression (initLiveArray). so I’ve taken it out of the function declaration (subscribeArray) that I had put it in before I spotted this. I’ve also moved it out of the main function. >> >> So I’m calling the function expression with: >> >> initLiveArray(); >> >> but it’s looking for three arguments, and properties thereof, so I’m not really clear on how I should be initialising it... >> >>> On 18 Sep 2017, at 16:58, Rick Jarvis > wrote: >>> >>> Thanks Italo - here it is: >>> >>> https://pastebin.com/pv6DdPkR >>> >>> >>>> On 18 Sep 2017, at 16:48, Ítalo Rossi > wrote: >>>> >>>> I don't think so, but you can always pastebin what you have so we can take a look >>>> Em seg, 18 de set de 2017 às 12:37, Rick Jarvis > escreveu: >>>> Italo, do you have a ‘complete code’ example for the really helpful verto example at evoluxbr.github.io/verto-docs ? I just can’t seem to get all the code in the right place, so that it all hangs together correctly. I’m also trying to send commands via Verto but not getting anywhere >>>> >>>> >>>>> On 7 Sep 2017, at 21:11, Ítalo Rossi > wrote: >>>>> >>>> >>>>> Verto communicator is a example implementation on what can be done with Verto. >>>>> >>>>> Check our tutorial to understand how to build a minimum app: >>>>> >>>>> https://evoluxbr.github.io/verto-docs/ >>>>> Em qui, 7 de set de 2017 às 17:02, Rick Jarvis > escreveu: >>>>> Looking to get into Verto, in particular handling voice calls with JS. Going through the source, I’m wondering: >>>>> >>>>> 1. What’s the difference between the Verto source and the Verto Communicator source? >>>>> >>>>> 2. What’s the best way to start from the bottom up - by this I mean that it seems hugely comprehensive, but rather than just use grunt to set it all up, I’d like to start simply with the basics… is there for instance a list of the bare minimum scripts / file structure to use? Apologies if this is a silly question, I’m still relatively new to JS and I don’t want to blow my mind in one go ;) >>>>> >>>>> Rick >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ > > ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH > ☞ irc.freenode.net #freeswitch ☞ http://freeswitch.org/g+ > > ClueCon Weekly Development Call > ☎ sip:888 at conference.freeswitch.org ☎ +19193869900 > > https://www.youtube.com/watch?v=oAxXgyx5jUw > https://www.youtube.com/watch?v=9XXgW34t40s > https://www.youtube.com/watch?v=NLaDpGQuZDA > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From bipin at xbipin.com Tue Sep 19 20:05:44 2017 From: bipin at xbipin.com (Bipin Patel) Date: Wed, 20 Sep 2017 00:05:44 +0400 Subject: [Freeswitch-users] FreeSWITCH on Raspberry Pi In-Reply-To: References: <59B7F678.1050403@telefaks.de> Message-ID: <15e9bbf11c0.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com> I believe 10-12 is no issues unless your recording then on raspbian lite, anything higher there is slight latency but not much but if you go over 20 then I believe you will face issues. Main issue is the nic runs on the USB bus so you don't get much performance on that. On September 19, 2017 8:35:56 PM Tahir Almas wrote: > I will be interested to know how many g.711 concurrent calls , the > latest respberri pi can support > > *Tahir Almas* > > ICT Innovations > http://www.ictinnovations.com > Leveraging open source in ICT > > > On Wed, Sep 13, 2017 at 1:35 AM, Tihomir Culjaga wrote: > >> if you need to do a lot of writes, sdcard is not a good option :=) >> >> this is why most embedded systems use RO fs and they do write something >> only when needed. >> >> I tried this one: http://khadas.com/vim/ and i can say its really good. >> >> As for RPi i was able to compile FS from source without no issues. Of >> course you need to reconfigure your OS (i used centos ) properly. I moved >> FS database on a ram disk and i can say it works really nice. >> FS logging is disabled.. i enable it only if needed when i debug somethig >> >> T. >> >> On 13 September 2017 at 00:28, Stanislav Sinyagin >> wrote: >> >>> >>> >>> On 12 Sep 2017 22:13, "jungle Boogie" wrote: >>> >>> On 12 September 2017 at 12:50, Stanislav Sinyagin >>> wrote: >>> > Alix is too old. The company has a much better board already: >>> > http://pcengines.ch/apu2.htm >>> > It can accommodate a real SSD with a much longer life cycle. >>> > Also I made a debian installer for it: >>> > https://github.com/ssinyagin/pcengines-apu-debian-cd >>> > >>> > in the ARM world, here's a box that houses a 2.5" disk: >>> > http://www.friendlyarm.com/index.php?route=product/product&p >>> roduct_id=192 >>> > >>> >>> This looks neat. Do you think it would be faster than an SD card on a >>> raspberry pi3? >>> >>> >>> I guess so, although it's SATA over USB 2.0 >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > ---------- > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From italo at freeswitch.org Wed Sep 20 00:22:24 2017 From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=) Date: Tue, 19 Sep 2017 21:22:24 -0300 Subject: [Freeswitch-users] mod_callcenter events variables In-Reply-To: <000c01d3312b$fd1430f0$f73c92d0$@gmail.com> References: <000c01d3312b$fd1430f0$f73c92d0$@gmail.com> Message-ID: They are Unix Timestamps in seconds, get a sample one and convert here: https://www.epochconverter.com/ They indicate the time the event occurs, CC-Member-Joined-Time when the member (caller) entered the queue, etc. On Tue, Sep 19, 2017 at 6:45 AM, Peter Nerád wrote: > Hi, > > > > Im parsing events from mod_callcenter but I have problem with variables > like CC-*-Time .. what is meaning of those numbers ? > > > > CC-Member-Joined-Time, CC-Member-Leaving-Time …. I have no idea… ( they > are not unix timestamps in microseconds ) > > > > Sorry for me bad English :-D > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Ítalo Rossi italo at freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From nerad.peter at gmail.com Wed Sep 20 03:46:51 2017 From: nerad.peter at gmail.com (=?UTF-8?B?UGV0ZXIgTmVyw6Fk?=) Date: Wed, 20 Sep 2017 05:46:51 +0200 Subject: [Freeswitch-users] mod_callcenter events variables In-Reply-To: References: <000c01d3312b$fd1430f0$f73c92d0$@gmail.com> Message-ID: <15e9d653bf8.281d.fa38bb7e8bf5b691e25471faaca20b8b@gmail.com> Ok Thanks Dňa 20. septembra 2017 2:23:14 používateľ Ítalo Rossi napísal: > They are Unix Timestamps in seconds, get a sample one and convert here: > https://www.epochconverter.com/ > > They indicate the time the event occurs, CC-Member-Joined-Time when the > member (caller) entered the queue, etc. > > On Tue, Sep 19, 2017 at 6:45 AM, Peter Nerád wrote: > >> Hi, >> >> >> >> Im parsing events from mod_callcenter but I have problem with variables >> like CC-*-Time .. what is meaning of those numbers ? >> >> >> >> CC-Member-Joined-Time, CC-Member-Leaving-Time …. I have no idea… ( they >> are not unix timestamps in microseconds ) >> >> >> >> Sorry for me bad English :-D >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Ítalo Rossi > italo at freeswitch.org > > > > ---------- > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From Harry.Z at iv66.net Wed Sep 20 05:39:51 2017 From: Harry.Z at iv66.net (Harry.Z) Date: Wed, 20 Sep 2017 05:39:51 +0000 Subject: [Freeswitch-users] FreeSWITCH Channel Variables Message-ID: <6b0742ef919f48f1988c68ff7429ade0@iv66.net> Dear All, FreeSWITCH Version 1.6.17-34-0fc0946~64bit (-34-0fc0946 64bit) How can I get below channel variables in dialplan? Caller-Callee-ID-Number: Other-Leg-Callee-ID-Number: Other-Leg-Destination-Number: variable_last_sent_callee_id_number: Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: From jeff.chua.linux at gmail.com Wed Sep 20 09:47:06 2017 From: jeff.chua.linux at gmail.com (Jeff Chua) Date: Wed, 20 Sep 2017 17:47:06 +0800 Subject: [Freeswitch-users] gsmopen one-way voice issue In-Reply-To: References: Message-ID: On Wed, Dec 28, 2016 at 6:58 PM, Jeff Chua wrote: > Any help is very much appreciated! > > Thanks, > Jeff > > On Tue, Dec 27, 2016 at 4:38 PM Jeff Chua wrote: >> >> On Tue, Dec 27, 2016 at 7:03 AM, Brian West wrote: >> >> > Any of those variables with dashes are invalid, you should look those >> > up, >> >> > those are params. >> >> >> >> Ok, I set all these to false ... >> >> >> >> # sip_profiles/internal.xml >> >> >> >> >> >> >> >> >> >> # ./sip_profiles/external.xml >> >> >> >> >> >> >> >> >> >> Still no sound from GSM to SIP. >> >> >> >> What else can I try? I still can't fix the one-way gamopen audio issue on freeswitch. It's running the latest freeswitch git version. I'm using Huaweil USB modem. The modem works fine in with Asterisk/chan_dongle (audio works both ways). SIP phone calling from wifi -> to freeswitch -> to another SIP phone on the same wfil -> audio works both ways. SIP phone calling from wifl -> to freeswitch -> to gsmopen -> Mobile Phone ... -> one-way audio only from SIP phone. GSM to SIP no audio. Thanks, Jeff From gmaruzz at gmail.com Wed Sep 20 10:44:19 2017 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Wed, 20 Sep 2017 12:44:19 +0200 Subject: [Freeswitch-users] gsmopen one-way voice issue In-Reply-To: References: Message-ID: try to change the interface devices that are used by gsopen, many times modem (dongle) vendor are messing with them. Rerea the documentation, and check out how to change those devices, eg, in various permutations. This is the only thing that comes to my mind. If this do not work, you will need to debug it deeply. -giovanni On 20 September 2017 at 11:47, Jeff Chua wrote: > On Wed, Dec 28, 2016 at 6:58 PM, Jeff Chua > wrote: > > Any help is very much appreciated! > > > > Thanks, > > Jeff > > > > On Tue, Dec 27, 2016 at 4:38 PM Jeff Chua > wrote: > >> > >> On Tue, Dec 27, 2016 at 7:03 AM, Brian West > wrote: > >> > >> > Any of those variables with dashes are invalid, you should look those > >> > up, > >> > >> > those are params. > >> > >> > >> > >> Ok, I set all these to false ... > >> > >> > >> > >> # sip_profiles/internal.xml > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> # ./sip_profiles/external.xml > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> Still no sound from GSM to SIP. > >> > >> > >> > >> What else can I try? > > I still can't fix the one-way gamopen audio issue on freeswitch. It's > running the latest freeswitch git version. > > I'm using Huaweil USB modem. The modem works fine in with > Asterisk/chan_dongle (audio works both ways). > > SIP phone calling from wifi -> to freeswitch -> to another SIP phone > on the same wfil -> audio works both ways. > > SIP phone calling from wifl -> to freeswitch -> to gsmopen -> Mobile > Phone ... -> one-way audio only from SIP phone. GSM to SIP no audio. > > > Thanks, > Jeff > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From rick at magicmail.mooo.com Wed Sep 20 11:20:25 2017 From: rick at magicmail.mooo.com (Rick Jarvis) Date: Wed, 20 Sep 2017 12:20:25 +0100 Subject: [Freeswitch-users] Verto Qs Message-ID: Ok I’ve asked a lot of questions about Verto, so I thought I’d summarise my current confusions here - if anyone can give me any pointers on any of these inline, that would be amazing! :) 1. Correct way to list all conference members via wss? 2. How to check if there is a current verto session / call in progress, and make the call if not. 3. Difference between :7443 and :8082 (is one for signalling and one for audio?) I have just purchased the latest FS book, 1.8, so hoping that will help also :) Thanks R From rick at magicmail.mooo.com Wed Sep 20 12:09:37 2017 From: rick at magicmail.mooo.com (Rick Jarvis) Date: Wed, 20 Sep 2017 13:09:37 +0100 Subject: [Freeswitch-users] Verto Qs In-Reply-To: References: Message-ID: So it seems the reason sendCommand isn’t working, is this line: "eventChannel": vertoConf.params.laData.modChannel ‘modChannel’ isn’t defined, vertoConf.params.laData is… when does the modChannel property get defined? > On 20 Sep 2017, at 12:20, Rick Jarvis wrote: > > Ok I’ve asked a lot of questions about Verto, so I thought I’d summarise my current confusions here - if anyone can give me any pointers on any of these inline, that would be amazing! :) > > 1. Correct way to list all conference members via wss? > > 2. How to check if there is a current verto session / call in progress, and make the call if not. > > 3. Difference between :7443 and :8082 (is one for signalling and one for audio?) > > I have just purchased the latest FS book, 1.8, so hoping that will help also :) > > Thanks > R > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From aqsyounas at gmail.com Wed Sep 20 12:45:19 2017 From: aqsyounas at gmail.com (Aqs Younas) Date: Wed, 20 Sep 2017 17:45:19 +0500 Subject: [Freeswitch-users] Unable to catch extra sip headers Message-ID: Greetings list, I am trying to catch extra sip header in dialplan but of no avail. I have tried info app and uuid_dump on the call to make show variables are appearing in console, but they were not. Though I can see header in sip trace when i run "sofia global siptrace on". This is how it is appearing in sip trace. 2017/09/20 11:22:37.025662 35.189.47.13:5060 -> 10.240.0.23:5063 INVITE sip:0061862083277 at sip99.mondotalk.com:5060;transport=udp SIP/2.0 Record-Route: From: "8300005";tag=9d3880-6709294b-13c4-50029-f25-3460e079-f25 To: *SRC-IP: 103.9.41.75* *Customer: 8300005* *Fax: out* Call-ID: 9eda40-6709294b-13c4-50029-f25-31be0b2b-f25 CSeq: 32590 INVITE Via: SIP/2.0/UDP 35.189.47.13:5060;branch=z9hG4bK9fed.42448b27.0 Via: SIP/2.0/UDP 103.9.41.75:5060 ;rport=5060;received=103.9.41.75;branch=z9hG4bK-f25-3b2a1d-491db0d7 Max-Forwards: 69 Supported: replaces,100rel Contact: User-Agent: TITAN Allow: INVITE,ACK,CANCEL,BYE,OPTIONS,REFER,INFO,NOTIFY,PRACK,MESSAGE,UPDATE Content-Type: application/sdp Content-Length: 461 v=0 o=a0000 8966 6672 IN IP4 103.9.41.75 s=SIP Call c=IN IP4 103.9.41.75 t=0 0 m=audio 53930 RTP/AVP 8 0 101 103 100 104 9 4 102 18 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:103 G726-32/8000 a=rtpmap:100 G726-16/8000 a=rtpmap:104 G726-40/8000 a=rtpmap:9 G722/16000 a=rtpmap:4 G723-63/8000 a=rtpmap:102 G726-24/8000 a=rtpmap:18 G729/8000 a=ptime:10 a=sendrecv a=rtcp-xr:voip-metrics and freeswitch at fax-syd-dev-01> uuid_dump 01784ffa-9df6-11e7-9716-291629b5fb60 Event-Name: CHANNEL_DATA Core-UUID: 499e114c-8330-11e7-b5e4-291629b5fb60 FreeSWITCH-Hostname: fax-syd-dev-01 FreeSWITCH-Switchname: fax-syd-dev-01 FreeSWITCH-IPv4: 10.240.0.23 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2017-09-20%2011%3A22%3A50 Event-Date-GMT: Wed,%2020%20Sep%202017%2011%3A22%3A50%20GMT Event-Date-Timestamp: 1505906570912351 Event-Calling-File: mod_commands.c Event-Calling-Function: uuid_dump_function Event-Calling-Line-Number: 6138 Event-Sequence: 461193 Channel-State: CS_EXECUTE Channel-Call-State: ACTIVE Channel-State-Number: 4 Channel-Name: sofia/external/8300005%40sip99.mondotalk.com%3A5060 Unique-ID: 01784ffa-9df6-11e7-9716-291629b5fb60 Call-Direction: inbound Presence-Call-Direction: inbound Channel-HIT-Dialplan: true Channel-Call-UUID: 01784ffa-9df6-11e7-9716-291629b5fb60 Answer-State: answered Channel-Read-Codec-Name: L16 Channel-Read-Codec-Rate: 8000 Channel-Read-Codec-Bit-Rate: 128000 Channel-Write-Codec-Name: PCMA Channel-Write-Codec-Rate: 8000 Channel-Write-Codec-Bit-Rate: 64000 Caller-Direction: inbound Caller-Logical-Direction: inbound Caller-Username: 8300005 Caller-Dialplan: XML Caller-Caller-ID-Name: 8300005 Caller-Caller-ID-Number: 8300005 Caller-Orig-Caller-ID-Name: 8300005 Caller-Orig-Caller-ID-Number: 8300005 Caller-Network-Addr: 35.189.47.13 Caller-ANI: 8300005 Caller-Destination-Number: 0061862083277 Caller-Unique-ID: 01784ffa-9df6-11e7-9716-291629b5fb60 Caller-Source: mod_sofia Caller-Context: public Caller-Channel-Name: sofia/external/8300005%40sip99.mondotalk.com%3A5060 Caller-Profile-Index: 1 Caller-Profile-Created-Time: 1505906554292369 Caller-Channel-Created-Time: 1505906554292369 Caller-Channel-Answered-Time: 1505906557192357 Caller-Channel-Progress-Time: 0 Caller-Channel-Progress-Media-Time: 1505906557192357 Caller-Channel-Hangup-Time: 0 Caller-Channel-Transfer-Time: 0 Caller-Channel-Resurrect-Time: 0 Caller-Channel-Bridged-Time: 0 Caller-Channel-Last-Hold: 0 Caller-Channel-Hold-Accum: 0 Caller-Screen-Bit: true Caller-Privacy-Hide-Name: false Caller-Privacy-Hide-Number: false variable_direction: inbound variable_uuid: 01784ffa-9df6-11e7-9716-291629b5fb60 variable_session_id: 4193 variable_video_media_flow: sendrecv variable_channel_name: sofia/external/8300005%40sip99.mondotalk.com%3A5060 variable_sip_invite_stamp: 1505906554292369 variable_sip_received_ip: 35.189.47.13 variable_sip_received_port: 5060 variable_sip_via_protocol: udp variable_sip_from_user_stripped: 8300005 variable_sofia_profile_name: external variable_recovery_profile_name: external variable_sip_invite_route_uri: %3Csip%3A35.189.47.13%3Blr%3Bftag%3D9d3880-6709294b-13c4-50029-f25-3460e079-f25%3Bdid%3D985.187c7a27%3E variable_sip_invite_record_route: %3Csip%3A35.189.47.13%3Blr%3Bftag%3D9d3880-6709294b-13c4-50029-f25-3460e079-f25%3Bdid%3D985.187c7a27%3E variable_sip_req_params: transport%3Dudp variable_sip_req_user: 0061862083277 variable_sip_req_port: 5060 variable_sip_req_uri: 0061862083277%40sip99.mondotalk.com%3A5060 variable_sip_req_host: sip99.mondotalk.com variable_sip_via_host: 35.189.47.13 variable_sip_via_port: 5060 variable_max_forwards: 69 variable_call_uuid: 01784ffa-9df6-11e7-9716-291629b5fb60 variable_audio_media_flow: sendrecv variable_rtp_audio_recv_pt: 8 variable_rtp_use_codec_name: PCMA variable_rtp_use_codec_rate: 8000 variable_rtp_use_codec_ptime: 10 variable_rtp_use_codec_channels: 1 variable_rtp_last_audio_codec_string: PCMA%408000h%4010i%401c variable_original_read_codec: PCMA variable_original_read_rate: 8000 variable_write_codec: PCMA variable_write_rate: 8000 variable_dtmf_type: rfc2833 variable_local_media_ip: 10.240.0.23 variable_local_media_port: 18268 variable_advertised_media_ip: 35.197.168.74 variable_rtp_use_timer_name: soft variable_rtp_use_pt: 8 variable_rtp_use_ssrc: 1844781053 variable_rtp_2833_send_payload: 101 variable_rtp_2833_recv_payload: 101 variable_remote_media_ip: 103.9.41.75 variable_remote_media_port: 53930 variable_endpoint_disposition: ANSWER variable_api_hangup_hook: lua%20fax_hangup.lua variable_fax_enable_t38_request: true variable_fax_enable_t38: true variable_current_application_data: / variable_current_application: rxfax variable_read_codec: L16 variable_read_rate: 8000 variable_rtp_local_sdp_str: v%3D0%0D%0Ao%3DFreeSWITCH%201505888289%201505888291%20IN%20IP4%2035.197.168.74%0D%0As%3DFreeSWITCH%0D%0Ac%3DIN%20IP4%2035.197.168.74%0D%0At%3D0%200%0D%0Am%3Dimage%2018268%20udptl%20t38%0D%0Aa%3DT38FaxVersion%3A0%0D%0Aa%3DT38MaxBitRate%3A14400%0D%0Aa%3DT38FaxFillBitRemoval%0D%0Aa%3DT38FaxRateManagement%3AtransferredTCF%0D%0Aa%3DT38FaxMaxBuffer%3A2000%0D%0Aa%3DT38FaxMaxDatagram%3A400%0D%0Aa%3DT38FaxUdpEC%3At38UDPRedundancy%0D%0A variable_sip_local_network_addr: 35.197.168.74 variable_sip_reply_host: 35.189.47.13 variable_sip_reply_port: 5060 variable_sip_network_ip: 35.189.47.13 variable_sip_network_port: 5060 variable_ep_codec_string: t38 variable_sip_user_agent: TITAN variable_sip_allow: INVITE,%20ACK,%20CANCEL,%20BYE,%20OPTIONS,%20REFER,%20INFO,%20NOTIFY,%20PRACK,%20MESSAGE,%20UPDATE variable_sip_full_via: SIP/2.0/UDP%2035.197.168.74%3A5063%3Breceived%3D35.197.168.74%3Brport%3D5063%3Bbranch%3Dz9hG4bKmeU9KQHKjjS3m variable_sip_full_from: %3Csip%3A0061862083277%40sip99.mondotalk.com %3A5060%3E%3Btag%3D2HQ8NmB9v6Uca variable_sip_to_display: 8300005 variable_sip_full_to: %228300005%22%20%3Csip%3A8300005%40sip99.mondotalk.com %3A5060%3E%3Btag%3D9d3880-6709294b-13c4-50029-f25-3460e079-f25 variable_sip_from_user: 0061862083277 variable_sip_from_port: 5060 variable_sip_from_uri: 0061862083277%40sip99.mondotalk.com%3A5060 variable_sip_from_host: sip99.mondotalk.com variable_sip_to_user: 8300005 variable_sip_to_port: 5060 variable_sip_to_uri: 8300005%40sip99.mondotalk.com%3A5060 variable_sip_to_host: sip99.mondotalk.com variable_sip_contact_params: transport%3Dudp variable_sip_contact_user: 8300005 variable_sip_contact_port: 5060 variable_sip_contact_uri: 8300005%40103.9.41.75%3A5060 variable_sip_contact_host: 103.9.41.75 variable_sip_to_tag: 9d3880-6709294b-13c4-50029-f25-3460e079-f25 variable_sip_from_tag: 2HQ8NmB9v6Uca variable_sip_cseq: 112617216 variable_sip_call_id: 9eda40-6709294b-13c4-50029-f25-31be0b2b-f25 variable_switch_r_sdp: v%3D0%0D%0Ao%3Da0000%208966%206673%20IN%20IP4%20103.9.41.75%0D%0As%3DSIP%20Call%0D%0Ac%3DIN%20IP4%20103.9.41.75%0D%0At%3D0%200%0D%0Am%3Dimage%2053930%20udptl%20t38%0D%0Aa%3DT38FaxVersion%3A0%0D%0Aa%3DT38MaxBitRate%3A14400%0D%0Aa%3DT38FaxRateManagement%3AtransferredTCF%0D%0Aa%3DT38FaxMaxBuffer%3A1800%0D%0Aa%3DT38FaxMaxDatagram%3A256%0D%0Aa%3DT38FaxUdpEC%3At38UDPRedundancy%0D%0Aa%3DT38Fax%0D%0A variable_rtp_use_codec_string: OPUS,G722,PCMU,PCMA,VP8 variable_has_t38: true variable_fax_v17_disabled: 0 variable_fax_ecm_requested: 1 variable_fax_filename: / I am stuck here, please advise. Best Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: From jurijs.ivolga at gmail.com Wed Sep 20 12:56:23 2017 From: jurijs.ivolga at gmail.com (Jurijs Ivolga) Date: Wed, 20 Sep 2017 15:56:23 +0300 Subject: [Freeswitch-users] Unable to catch extra sip headers In-Reply-To: References: Message-ID: Hi, To catch this sip headers, it should have "X-" in front. So instead "Fax", please try "X-Fax". With kind regards, Jurijs On Wed, Sep 20, 2017 at 3:45 PM, Aqs Younas wrote: > Greetings list, > > I am trying to catch extra sip header in dialplan but of no avail. > I have tried info app and uuid_dump on the call to make show variables are > appearing in console, but they were not. Though I can see header in sip > trace when i run "sofia global siptrace on". > > This is how it is appearing in sip trace. > > 2017/09/20 11:22:37.025662 35.189.47.13:5060 -> 10.240.0.23:5063 > INVITE sip:0061862083277 at sip99.mondotalk.com:5060;transport=udp SIP/2.0 > Record-Route: f25-3460e079-f25;did=985.187c7a27> > From: "8300005";tag= > 9d3880-6709294b-13c4-50029-f25-3460e079-f25 > To: > *SRC-IP: 103.9.41.75* > *Customer: 8300005* > *Fax: out* > Call-ID: 9eda40-6709294b-13c4-50029-f25-31be0b2b-f25 > CSeq: 32590 INVITE > Via: SIP/2.0/UDP 35.189.47.13:5060;branch=z9hG4bK9fed.42448b27.0 > Via: SIP/2.0/UDP 103.9.41.75:5060;rport=5060;received=103.9.41.75;branch= > z9hG4bK-f25-3b2a1d-491db0d7 > Max-Forwards: 69 > Supported: replaces,100rel > Contact: > User-Agent: TITAN > Allow: INVITE,ACK,CANCEL,BYE,OPTIONS,REFER,INFO,NOTIFY,PRACK, > MESSAGE,UPDATE > Content-Type: application/sdp > Content-Length: 461 > > v=0 > o=a0000 8966 6672 IN IP4 103.9.41.75 > s=SIP Call > c=IN IP4 103.9.41.75 > t=0 0 > m=audio 53930 RTP/AVP 8 0 101 103 100 104 9 4 102 18 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=rtpmap:103 G726-32/8000 > a=rtpmap:100 G726-16/8000 > a=rtpmap:104 G726-40/8000 > a=rtpmap:9 G722/16000 > a=rtpmap:4 G723-63/8000 > a=rtpmap:102 G726-24/8000 > a=rtpmap:18 G729/8000 > a=ptime:10 > a=sendrecv > a=rtcp-xr:voip-metrics > > > and > > freeswitch at fax-syd-dev-01> uuid_dump 01784ffa-9df6-11e7-9716-291629b5fb60 > Event-Name: CHANNEL_DATA > Core-UUID: 499e114c-8330-11e7-b5e4-291629b5fb60 > FreeSWITCH-Hostname: fax-syd-dev-01 > FreeSWITCH-Switchname: fax-syd-dev-01 > FreeSWITCH-IPv4: 10.240.0.23 > FreeSWITCH-IPv6: %3A%3A1 > Event-Date-Local: 2017-09-20%2011%3A22%3A50 > Event-Date-GMT: Wed,%2020%20Sep%202017%2011%3A22%3A50%20GMT > Event-Date-Timestamp: 1505906570912351 > Event-Calling-File: mod_commands.c > Event-Calling-Function: uuid_dump_function > Event-Calling-Line-Number: 6138 > Event-Sequence: 461193 > Channel-State: CS_EXECUTE > Channel-Call-State: ACTIVE > Channel-State-Number: 4 > Channel-Name: sofia/external/8300005%40sip99.mondotalk.com%3A5060 > Unique-ID: 01784ffa-9df6-11e7-9716-291629b5fb60 > Call-Direction: inbound > Presence-Call-Direction: inbound > Channel-HIT-Dialplan: true > Channel-Call-UUID: 01784ffa-9df6-11e7-9716-291629b5fb60 > Answer-State: answered > Channel-Read-Codec-Name: L16 > Channel-Read-Codec-Rate: 8000 > Channel-Read-Codec-Bit-Rate: 128000 > Channel-Write-Codec-Name: PCMA > Channel-Write-Codec-Rate: 8000 > Channel-Write-Codec-Bit-Rate: 64000 > Caller-Direction: inbound > Caller-Logical-Direction: inbound > Caller-Username: 8300005 > Caller-Dialplan: XML > Caller-Caller-ID-Name: 8300005 > Caller-Caller-ID-Number: 8300005 > Caller-Orig-Caller-ID-Name: 8300005 > Caller-Orig-Caller-ID-Number: 8300005 > Caller-Network-Addr: 35.189.47.13 > Caller-ANI: 8300005 > Caller-Destination-Number: 0061862083277 > Caller-Unique-ID: 01784ffa-9df6-11e7-9716-291629b5fb60 > Caller-Source: mod_sofia > Caller-Context: public > Caller-Channel-Name: sofia/external/8300005%40sip99.mondotalk.com%3A5060 > Caller-Profile-Index: 1 > Caller-Profile-Created-Time: 1505906554292369 > Caller-Channel-Created-Time: 1505906554292369 > Caller-Channel-Answered-Time: 1505906557192357 > Caller-Channel-Progress-Time: 0 > Caller-Channel-Progress-Media-Time: 1505906557192357 > Caller-Channel-Hangup-Time: 0 > Caller-Channel-Transfer-Time: 0 > Caller-Channel-Resurrect-Time: 0 > Caller-Channel-Bridged-Time: 0 > Caller-Channel-Last-Hold: 0 > Caller-Channel-Hold-Accum: 0 > Caller-Screen-Bit: true > Caller-Privacy-Hide-Name: false > Caller-Privacy-Hide-Number: false > variable_direction: inbound > variable_uuid: 01784ffa-9df6-11e7-9716-291629b5fb60 > variable_session_id: 4193 > variable_video_media_flow: sendrecv > variable_channel_name: sofia/external/8300005%40sip99.mondotalk.com%3A5060 > variable_sip_invite_stamp: 1505906554292369 > variable_sip_received_ip: 35.189.47.13 > variable_sip_received_port: 5060 > variable_sip_via_protocol: udp > variable_sip_from_user_stripped: 8300005 > variable_sofia_profile_name: external > variable_recovery_profile_name: external > variable_sip_invite_route_uri: %3Csip%3A35.189.47.13%3Blr% > 3Bftag%3D9d3880-6709294b-13c4-50029-f25-3460e079-f25%3Bdid% > 3D985.187c7a27%3E > variable_sip_invite_record_route: %3Csip%3A35.189.47.13%3Blr% > 3Bftag%3D9d3880-6709294b-13c4-50029-f25-3460e079-f25%3Bdid% > 3D985.187c7a27%3E > variable_sip_req_params: transport%3Dudp > variable_sip_req_user: 0061862083277 > variable_sip_req_port: 5060 > variable_sip_req_uri: 0061862083277%40sip99.mondotalk.com%3A5060 > variable_sip_req_host: sip99.mondotalk.com > variable_sip_via_host: 35.189.47.13 > variable_sip_via_port: 5060 > variable_max_forwards: 69 > variable_call_uuid: 01784ffa-9df6-11e7-9716-291629b5fb60 > variable_audio_media_flow: sendrecv > variable_rtp_audio_recv_pt: 8 > variable_rtp_use_codec_name: PCMA > variable_rtp_use_codec_rate: 8000 > variable_rtp_use_codec_ptime: 10 > variable_rtp_use_codec_channels: 1 > variable_rtp_last_audio_codec_string: PCMA%408000h%4010i%401c > variable_original_read_codec: PCMA > variable_original_read_rate: 8000 > variable_write_codec: PCMA > variable_write_rate: 8000 > variable_dtmf_type: rfc2833 > variable_local_media_ip: 10.240.0.23 > variable_local_media_port: 18268 > variable_advertised_media_ip: 35.197.168.74 > variable_rtp_use_timer_name: soft > variable_rtp_use_pt: 8 > variable_rtp_use_ssrc: 1844781053 > variable_rtp_2833_send_payload: 101 > variable_rtp_2833_recv_payload: 101 > variable_remote_media_ip: 103.9.41.75 > variable_remote_media_port: 53930 > variable_endpoint_disposition: ANSWER > variable_api_hangup_hook: lua%20fax_hangup.lua > variable_fax_enable_t38_request: true > variable_fax_enable_t38: true > variable_current_application_data: / > variable_current_application: rxfax > variable_read_codec: L16 > variable_read_rate: 8000 > variable_rtp_local_sdp_str: v%3D0%0D%0Ao%3DFreeSWITCH% > 201505888289%201505888291%20IN%20IP4%2035.197.168.74%0D% > 0As%3DFreeSWITCH%0D%0Ac%3DIN%20IP4%2035.197.168.74%0D%0At% > 3D0%200%0D%0Am%3Dimage%2018268%20udptl%20t38%0D%0Aa% > 3DT38FaxVersion%3A0%0D%0Aa%3DT38MaxBitRate%3A14400%0D% > 0Aa%3DT38FaxFillBitRemoval%0D%0Aa%3DT38FaxRateManagement% > 3AtransferredTCF%0D%0Aa%3DT38FaxMaxBuffer%3A2000%0D% > 0Aa%3DT38FaxMaxDatagram%3A400%0D%0Aa%3DT38FaxUdpEC% > 3At38UDPRedundancy%0D%0A > variable_sip_local_network_addr: 35.197.168.74 > variable_sip_reply_host: 35.189.47.13 > variable_sip_reply_port: 5060 > variable_sip_network_ip: 35.189.47.13 > variable_sip_network_port: 5060 > variable_ep_codec_string: t38 > variable_sip_user_agent: TITAN > variable_sip_allow: INVITE,%20ACK,%20CANCEL,%20BYE,%20OPTIONS,%20REFER,% > 20INFO,%20NOTIFY,%20PRACK,%20MESSAGE,%20UPDATE > variable_sip_full_via: SIP/2.0/UDP%2035.197.168.74% > 3A5063%3Breceived%3D35.197.168.74%3Brport%3D5063%3Bbranch% > 3Dz9hG4bKmeU9KQHKjjS3m > variable_sip_full_from: %3Csip%3A0061862083277%40sip99.mondotalk.com > %3A5060%3E%3Btag%3D2HQ8NmB9v6Uca > variable_sip_to_display: 8300005 > variable_sip_full_to: %228300005%22%20%3Csip%3A8300005%40sip99.mondotalk. > com%3A5060%3E%3Btag%3D9d3880-6709294b-13c4-50029-f25-3460e079-f25 > variable_sip_from_user: 0061862083277 > variable_sip_from_port: 5060 > variable_sip_from_uri: 0061862083277%40sip99.mondotalk.com%3A5060 > variable_sip_from_host: sip99.mondotalk.com > variable_sip_to_user: 8300005 > variable_sip_to_port: 5060 > variable_sip_to_uri: 8300005%40sip99.mondotalk.com%3A5060 > variable_sip_to_host: sip99.mondotalk.com > variable_sip_contact_params: transport%3Dudp > variable_sip_contact_user: 8300005 > variable_sip_contact_port: 5060 > variable_sip_contact_uri: 8300005%40103.9.41.75%3A5060 > variable_sip_contact_host: 103.9.41.75 > variable_sip_to_tag: 9d3880-6709294b-13c4-50029-f25-3460e079-f25 > variable_sip_from_tag: 2HQ8NmB9v6Uca > variable_sip_cseq: 112617216 > variable_sip_call_id: 9eda40-6709294b-13c4-50029-f25-31be0b2b-f25 > variable_switch_r_sdp: v%3D0%0D%0Ao%3Da0000%208966% > 206673%20IN%20IP4%20103.9.41.75%0D%0As%3DSIP%20Call%0D%0Ac% > 3DIN%20IP4%20103.9.41.75%0D%0At%3D0%200%0D%0Am%3Dimage% > 2053930%20udptl%20t38%0D%0Aa%3DT38FaxVersion%3A0%0D%0Aa% > 3DT38MaxBitRate%3A14400%0D%0Aa%3DT38FaxRateManagement% > 3AtransferredTCF%0D%0Aa%3DT38FaxMaxBuffer%3A1800%0D% > 0Aa%3DT38FaxMaxDatagram%3A256%0D%0Aa%3DT38FaxUdpEC% > 3At38UDPRedundancy%0D%0Aa%3DT38Fax%0D%0A > variable_rtp_use_codec_string: OPUS,G722,PCMU,PCMA,VP8 > variable_has_t38: true > variable_fax_v17_disabled: 0 > variable_fax_ecm_requested: 1 > variable_fax_filename: / > > > I am stuck here, please advise. > > Best Regards, > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From donguyenha at gmail.com Wed Sep 20 12:59:02 2017 From: donguyenha at gmail.com (Do Nguyen Ha) Date: Wed, 20 Sep 2017 19:59:02 +0700 Subject: [Freeswitch-users] Verto Qs In-Reply-To: References: Message-ID: Hi Rick I did get confused like you for question number 3 3. Difference between :7443 and :8082 >>>> The port 7443 is used for sip over web socket protocol(RFC 7118) - you can use command "sofia status profile internal reg", the port 8082 is used for verto protocol- you can use command"verto status - not sure it is right command", both ports are used for signaling Hope this help Ha` On Sep 20, 2017 18:23, "Rick Jarvis" wrote: Ok I’ve asked a lot of questions about Verto, so I thought I’d summarise my current confusions here - if anyone can give me any pointers on any of these inline, that would be amazing! :) 1. Correct way to list all conference members via wss? 2. How to check if there is a current verto session / call in progress, and make the call if not. 3. Difference between :7443 and :8082 (is one for signalling and one for audio?) I have just purchased the latest FS book, 1.8, so hoping that will help also :) Thanks R _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From ssinyagin at gmail.com Wed Sep 20 13:45:57 2017 From: ssinyagin at gmail.com (Stanislav Sinyagin) Date: Wed, 20 Sep 2017 15:45:57 +0200 Subject: [Freeswitch-users] Session recording performance Message-ID: I'm building an environment for voice quality testing, and found a strange behavior. Maybe someone has an explanation. Two identical physical computers (PC Engines APU2), connected back to back (A and B). A initiates the test calls to B's SIP address, and then both ends go through a small dialplan (less than 100 lines of XML, similar to https://github.com/voxserv/fsqa/blob/master/fsqa.xml and start recording the session. The dialplan initiates session recording to a Raw file, and plays back a 20-second audio sample, and then inserts the filename into a Redis queue, then hangs up. Then the files get processed by Sevana AQuA software in lowest CPU priority, and deleted. The files are written into a tmpfs filesystem. With 2 calls per second, I get 48 simultaneous calls, and B indicates frame drops in the audio recordings, while A shows good results. The CPU average idle time is always above 50% on both computers. Why is the B side (the one receiving INVITEs) getting more frame drops than the A (the one originating the calls)? I flipped the sides, and then the other machine started to get more drops. At first I started recording in WAV format, and similar effect started to appear at a call every 2 seconds, resulting in 12 simultaneous calls. With raw recording, the performance is much better, but still it's a mistery why there are dropped frames in the audio. The CPU is always running at its highest frequency, under "performance" governor. From ssinyagin at gmail.com Wed Sep 20 13:51:26 2017 From: ssinyagin at gmail.com (Stanislav Sinyagin) Date: Wed, 20 Sep 2017 15:51:26 +0200 Subject: [Freeswitch-users] FreeSWITCH on Raspberry Pi In-Reply-To: <15e9bbf11c0.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com> References: <59B7F678.1050403@telefaks.de> <15e9bbf11c0.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com> Message-ID: this ARM board has a Gigabit NIC, directly handled by the SOC: http://www.friendlyarm.com/index.php?route=product/product&path=69&product_id=196 I made a short test, and you can produce a gigabit of TCP traffic with it. Didn't try it with FreeSWITCH though. On Tue, Sep 19, 2017 at 10:05 PM, Bipin Patel wrote: > I believe 10-12 is no issues unless your recording then on raspbian lite, > anything higher there is slight latency but not much but if you go over 20 > then I believe you will face issues. Main issue is the nic runs on the USB > bus so you don't get much performance on that. > > On September 19, 2017 8:35:56 PM Tahir Almas > wrote: >> >> I will be interested to know how many g.711 concurrent calls , the >> latest respberri pi can support >> >> Tahir Almas >> >> ICT Innovations >> http://www.ictinnovations.com >> Leveraging open source in ICT >> >> >> On Wed, Sep 13, 2017 at 1:35 AM, Tihomir Culjaga >> wrote: >>> >>> if you need to do a lot of writes, sdcard is not a good option :=) >>> >>> this is why most embedded systems use RO fs and they do write something >>> only when needed. >>> >>> I tried this one: http://khadas.com/vim/ and i can say its really good. >>> >>> As for RPi i was able to compile FS from source without no issues. Of >>> course you need to reconfigure your OS (i used centos ) properly. I moved FS >>> database on a ram disk and i can say it works really nice. >>> FS logging is disabled.. i enable it only if needed when i debug somethig >>> >>> T. >>> >>> On 13 September 2017 at 00:28, Stanislav Sinyagin >>> wrote: >>>> >>>> >>>> >>>> On 12 Sep 2017 22:13, "jungle Boogie" wrote: >>>> >>>> On 12 September 2017 at 12:50, Stanislav Sinyagin >>>> wrote: >>>> > Alix is too old. The company has a much better board already: >>>> > http://pcengines.ch/apu2.htm >>>> > It can accommodate a real SSD with a much longer life cycle. >>>> > Also I made a debian installer for it: >>>> > https://github.com/ssinyagin/pcengines-apu-debian-cd >>>> > >>>> > in the ARM world, here's a box that houses a 2.5" disk: >>>> > >>>> > http://www.friendlyarm.com/index.php?route=product/product&product_id=192 >>>> > >>>> >>>> This looks neat. Do you think it would be faster than an SD card on a >>>> raspberry pi3? >>>> >>>> >>>> I guess so, although it's SATA over USB 2.0 >>>> >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From rick at magicmail.mooo.com Wed Sep 20 13:54:42 2017 From: rick at magicmail.mooo.com (Rick Jarvis) Date: Wed, 20 Sep 2017 14:54:42 +0100 Subject: [Freeswitch-users] Verto Qs In-Reply-To: References: Message-ID: <430A784F-3570-433E-A7F7-53644DD2F113@magicmail.mooo.com> Thanks Ha, that makes sense. I’m still trying to work out how to send jsonapi commands via 8082 with Javascript. Any ideas? > On 20 Sep 2017, at 13:59, Do Nguyen Ha wrote: > > Hi Rick > I did get confused like you for question number 3 > 3. Difference between :7443 and :8082 > >>>> The port 7443 is used for sip over web socket protocol(RFC 7118) - you can use command "sofia status profile internal reg", the port 8082 is used for verto protocol- you can use command"verto status - not sure it is right command", both ports are used for signaling > > Hope this help > Ha` > > On Sep 20, 2017 18:23, "Rick Jarvis" > wrote: > Ok I’ve asked a lot of questions about Verto, so I thought I’d summarise my current confusions here - if anyone can give me any pointers on any of these inline, that would be amazing! :) > > 1. Correct way to list all conference members via wss? > > 2. How to check if there is a current verto session / call in progress, and make the call if not. > > 3. Difference between :7443 and :8082 (is one for signalling and one for audio?) > > I have just purchased the latest FS book, 1.8, so hoping that will help also :) > > Thanks > R > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From aqsyounas at gmail.com Wed Sep 20 14:29:34 2017 From: aqsyounas at gmail.com (Aqs Younas) Date: Wed, 20 Sep 2017 19:29:34 +0500 Subject: [Freeswitch-users] Unable to catch extra sip headers In-Reply-To: References: Message-ID: Thanks for reply, may i ask why it is necessary to prefix extra header with "X-" to make it visible in freeswitch? On 20 September 2017 at 17:56, Jurijs Ivolga wrote: > Hi, > > To catch this sip headers, it should have "X-" in front. > > So instead "Fax", please try "X-Fax". > > With kind regards, > > Jurijs > > On Wed, Sep 20, 2017 at 3:45 PM, Aqs Younas wrote: > >> Greetings list, >> >> I am trying to catch extra sip header in dialplan but of no avail. >> I have tried info app and uuid_dump on the call to make show variables >> are appearing in console, but they were not. Though I can see header in sip >> trace when i run "sofia global siptrace on". >> >> This is how it is appearing in sip trace. >> >> 2017/09/20 11:22:37.025662 35.189.47.13:5060 -> 10.240.0.23:5063 >> INVITE sip:0061862083277 at sip99.mondotalk.com:5060;transport=udp SIP/2.0 >> Record-Route: > 3460e079-f25;did=985.187c7a27> >> From: "8300005";tag=9d3880- >> 6709294b-13c4-50029-f25-3460e079-f25 >> To: >> *SRC-IP: 103.9.41.75* >> *Customer: 8300005* >> *Fax: out* >> Call-ID: 9eda40-6709294b-13c4-50029-f25-31be0b2b-f25 >> CSeq: 32590 INVITE >> Via: SIP/2.0/UDP 35.189.47.13:5060;branch=z9hG4bK9fed.42448b27.0 >> Via: SIP/2.0/UDP 103.9.41.75:5060;rport=5060;re >> ceived=103.9.41.75;branch=z9hG4bK-f25-3b2a1d-491db0d7 >> Max-Forwards: 69 >> Supported: replaces,100rel >> Contact: >> User-Agent: TITAN >> Allow: INVITE,ACK,CANCEL,BYE,OPTIONS,REFER,INFO,NOTIFY,PRACK,MESSAG >> E,UPDATE >> Content-Type: application/sdp >> Content-Length: 461 >> >> v=0 >> o=a0000 8966 6672 IN IP4 103.9.41.75 >> s=SIP Call >> c=IN IP4 103.9.41.75 >> t=0 0 >> m=audio 53930 RTP/AVP 8 0 101 103 100 104 9 4 102 18 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> a=rtpmap:103 G726-32/8000 >> a=rtpmap:100 G726-16/8000 >> a=rtpmap:104 G726-40/8000 >> a=rtpmap:9 G722/16000 >> a=rtpmap:4 G723-63/8000 >> a=rtpmap:102 G726-24/8000 >> a=rtpmap:18 G729/8000 >> a=ptime:10 >> a=sendrecv >> a=rtcp-xr:voip-metrics >> >> >> and >> >> freeswitch at fax-syd-dev-01> uuid_dump 01784ffa-9df6-11e7-9716-291629b5fb60 >> Event-Name: CHANNEL_DATA >> Core-UUID: 499e114c-8330-11e7-b5e4-291629b5fb60 >> FreeSWITCH-Hostname: fax-syd-dev-01 >> FreeSWITCH-Switchname: fax-syd-dev-01 >> FreeSWITCH-IPv4: 10.240.0.23 >> FreeSWITCH-IPv6: %3A%3A1 >> Event-Date-Local: 2017-09-20%2011%3A22%3A50 >> Event-Date-GMT: Wed,%2020%20Sep%202017%2011%3A22%3A50%20GMT >> Event-Date-Timestamp: 1505906570912351 >> Event-Calling-File: mod_commands.c >> Event-Calling-Function: uuid_dump_function >> Event-Calling-Line-Number: 6138 >> Event-Sequence: 461193 >> Channel-State: CS_EXECUTE >> Channel-Call-State: ACTIVE >> Channel-State-Number: 4 >> Channel-Name: sofia/external/8300005%40sip99.mondotalk.com%3A5060 >> Unique-ID: 01784ffa-9df6-11e7-9716-291629b5fb60 >> Call-Direction: inbound >> Presence-Call-Direction: inbound >> Channel-HIT-Dialplan: true >> Channel-Call-UUID: 01784ffa-9df6-11e7-9716-291629b5fb60 >> Answer-State: answered >> Channel-Read-Codec-Name: L16 >> Channel-Read-Codec-Rate: 8000 >> Channel-Read-Codec-Bit-Rate: 128000 >> Channel-Write-Codec-Name: PCMA >> Channel-Write-Codec-Rate: 8000 >> Channel-Write-Codec-Bit-Rate: 64000 >> Caller-Direction: inbound >> Caller-Logical-Direction: inbound >> Caller-Username: 8300005 >> Caller-Dialplan: XML >> Caller-Caller-ID-Name: 8300005 >> Caller-Caller-ID-Number: 8300005 >> Caller-Orig-Caller-ID-Name: 8300005 >> Caller-Orig-Caller-ID-Number: 8300005 >> Caller-Network-Addr: 35.189.47.13 >> Caller-ANI: 8300005 >> Caller-Destination-Number: 0061862083277 >> Caller-Unique-ID: 01784ffa-9df6-11e7-9716-291629b5fb60 >> Caller-Source: mod_sofia >> Caller-Context: public >> Caller-Channel-Name: sofia/external/8300005%40sip99.mondotalk.com%3A5060 >> Caller-Profile-Index: 1 >> Caller-Profile-Created-Time: 1505906554292369 >> Caller-Channel-Created-Time: 1505906554292369 >> Caller-Channel-Answered-Time: 1505906557192357 >> Caller-Channel-Progress-Time: 0 >> Caller-Channel-Progress-Media-Time: 1505906557192357 >> Caller-Channel-Hangup-Time: 0 >> Caller-Channel-Transfer-Time: 0 >> Caller-Channel-Resurrect-Time: 0 >> Caller-Channel-Bridged-Time: 0 >> Caller-Channel-Last-Hold: 0 >> Caller-Channel-Hold-Accum: 0 >> Caller-Screen-Bit: true >> Caller-Privacy-Hide-Name: false >> Caller-Privacy-Hide-Number: false >> variable_direction: inbound >> variable_uuid: 01784ffa-9df6-11e7-9716-291629b5fb60 >> variable_session_id: 4193 >> variable_video_media_flow: sendrecv >> variable_channel_name: sofia/external/8300005%40sip99.mondotalk.com >> %3A5060 >> variable_sip_invite_stamp: 1505906554292369 >> variable_sip_received_ip: 35.189.47.13 >> variable_sip_received_port: 5060 >> variable_sip_via_protocol: udp >> variable_sip_from_user_stripped: 8300005 >> variable_sofia_profile_name: external >> variable_recovery_profile_name: external >> variable_sip_invite_route_uri: %3Csip%3A35.189.47.13%3Blr%3Bf >> tag%3D9d3880-6709294b-13c4-50029-f25-3460e079-f25%3Bdid%3D985.187c7a27%3E >> variable_sip_invite_record_route: %3Csip%3A35.189.47.13%3Blr%3Bf >> tag%3D9d3880-6709294b-13c4-50029-f25-3460e079-f25%3Bdid%3D985.187c7a27%3E >> variable_sip_req_params: transport%3Dudp >> variable_sip_req_user: 0061862083277 >> variable_sip_req_port: 5060 >> variable_sip_req_uri: 0061862083277%40sip99.mondotalk.com%3A5060 >> variable_sip_req_host: sip99.mondotalk.com >> variable_sip_via_host: 35.189.47.13 >> variable_sip_via_port: 5060 >> variable_max_forwards: 69 >> variable_call_uuid: 01784ffa-9df6-11e7-9716-291629b5fb60 >> variable_audio_media_flow: sendrecv >> variable_rtp_audio_recv_pt: 8 >> variable_rtp_use_codec_name: PCMA >> variable_rtp_use_codec_rate: 8000 >> variable_rtp_use_codec_ptime: 10 >> variable_rtp_use_codec_channels: 1 >> variable_rtp_last_audio_codec_string: PCMA%408000h%4010i%401c >> variable_original_read_codec: PCMA >> variable_original_read_rate: 8000 >> variable_write_codec: PCMA >> variable_write_rate: 8000 >> variable_dtmf_type: rfc2833 >> variable_local_media_ip: 10.240.0.23 >> variable_local_media_port: 18268 >> variable_advertised_media_ip: 35.197.168.74 >> variable_rtp_use_timer_name: soft >> variable_rtp_use_pt: 8 >> variable_rtp_use_ssrc: 1844781053 >> variable_rtp_2833_send_payload: 101 >> variable_rtp_2833_recv_payload: 101 >> variable_remote_media_ip: 103.9.41.75 >> variable_remote_media_port: 53930 >> variable_endpoint_disposition: ANSWER >> variable_api_hangup_hook: lua%20fax_hangup.lua >> variable_fax_enable_t38_request: true >> variable_fax_enable_t38: true >> variable_current_application_data: / >> variable_current_application: rxfax >> variable_read_codec: L16 >> variable_read_rate: 8000 >> variable_rtp_local_sdp_str: v%3D0%0D%0Ao%3DFreeSWITCH%2015 >> 05888289%201505888291%20IN%20IP4%2035.197.168.74%0D%0As% >> 3DFreeSWITCH%0D%0Ac%3DIN%20IP4%2035.197.168.74%0D%0At%3D0% >> 200%0D%0Am%3Dimage%2018268%20udptl%20t38%0D%0Aa%3DT38FaxV >> ersion%3A0%0D%0Aa%3DT38MaxBitRate%3A14400%0D%0Aa% >> 3DT38FaxFillBitRemoval%0D%0Aa%3DT38FaxRateManagement%3Atrans >> ferredTCF%0D%0Aa%3DT38FaxMaxBuffer%3A2000%0D%0Aa% >> 3DT38FaxMaxDatagram%3A400%0D%0Aa%3DT38FaxUdpEC%3At38UDPRedundancy%0D%0A >> variable_sip_local_network_addr: 35.197.168.74 >> variable_sip_reply_host: 35.189.47.13 >> variable_sip_reply_port: 5060 >> variable_sip_network_ip: 35.189.47.13 >> variable_sip_network_port: 5060 >> variable_ep_codec_string: t38 >> variable_sip_user_agent: TITAN >> variable_sip_allow: INVITE,%20ACK,%20CANCEL,%20BYE >> ,%20OPTIONS,%20REFER,%20INFO,%20NOTIFY,%20PRACK,%20MESSAGE,%20UPDATE >> variable_sip_full_via: SIP/2.0/UDP%2035.197.168.74%3A >> 5063%3Breceived%3D35.197.168.74%3Brport%3D5063%3Bbranch%3Dz >> 9hG4bKmeU9KQHKjjS3m >> variable_sip_full_from: %3Csip%3A0061862083277%40sip99.mondotalk.com >> %3A5060%3E%3Btag%3D2HQ8NmB9v6Uca >> variable_sip_to_display: 8300005 >> variable_sip_full_to: %228300005%22%20%3Csip%3A8300005% >> 40sip99.mondotalk.com%3A5060%3E%3Btag%3D9d3880-67092 >> 94b-13c4-50029-f25-3460e079-f25 >> variable_sip_from_user: 0061862083277 >> variable_sip_from_port: 5060 >> variable_sip_from_uri: 0061862083277%40sip99.mondotalk.com%3A5060 >> variable_sip_from_host: sip99.mondotalk.com >> variable_sip_to_user: 8300005 >> variable_sip_to_port: 5060 >> variable_sip_to_uri: 8300005%40sip99.mondotalk.com%3A5060 >> variable_sip_to_host: sip99.mondotalk.com >> variable_sip_contact_params: transport%3Dudp >> variable_sip_contact_user: 8300005 >> variable_sip_contact_port: 5060 >> variable_sip_contact_uri: 8300005%40103.9.41.75%3A5060 >> variable_sip_contact_host: 103.9.41.75 >> variable_sip_to_tag: 9d3880-6709294b-13c4-50029-f25-3460e079-f25 >> variable_sip_from_tag: 2HQ8NmB9v6Uca >> variable_sip_cseq: 112617216 >> variable_sip_call_id: 9eda40-6709294b-13c4-50029-f25-31be0b2b-f25 >> variable_switch_r_sdp: v%3D0%0D%0Ao%3Da0000%208966%20 >> 6673%20IN%20IP4%20103.9.41.75%0D%0As%3DSIP%20Call%0D%0Ac%3DI >> N%20IP4%20103.9.41.75%0D%0At%3D0%200%0D%0Am%3Dimage%2053930 >> %20udptl%20t38%0D%0Aa%3DT38FaxVersion%3A0%0D%0Aa%3DT38MaxBit >> Rate%3A14400%0D%0Aa%3DT38FaxRateManagement%3Atrans >> ferredTCF%0D%0Aa%3DT38FaxMaxBuffer%3A1800%0D%0Aa% >> 3DT38FaxMaxDatagram%3A256%0D%0Aa%3DT38FaxUdpEC%3At38UDPRedu >> ndancy%0D%0Aa%3DT38Fax%0D%0A >> variable_rtp_use_codec_string: OPUS,G722,PCMU,PCMA,VP8 >> variable_has_t38: true >> variable_fax_v17_disabled: 0 >> variable_fax_ecm_requested: 1 >> variable_fax_filename: / >> >> >> I am stuck here, please advise. >> >> Best Regards, >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From brandon at cryy.com Wed Sep 20 15:27:13 2017 From: brandon at cryy.com (Brandon Armstead) Date: Wed, 20 Sep 2017 15:27:13 +0000 Subject: [Freeswitch-users] Session recording performance In-Reply-To: References: Message-ID: Same kernel same everything? Are they "exact replicas" identical? Perhaps see if you can do this test again and receive the exact system calls A vs B are doing and then B to A should give you further insight to your answer. On Wed, Sep 20, 2017 at 6:46 AM Stanislav Sinyagin wrote: > I'm building an environment for voice quality testing, and found a > strange behavior. Maybe someone has an explanation. > > Two identical physical computers (PC Engines APU2), connected back to > back (A and B). > > A initiates the test calls to B's SIP address, and then both ends go > through a small dialplan (less than 100 lines of XML, similar to > https://github.com/voxserv/fsqa/blob/master/fsqa.xml > and start recording the session. > > The dialplan initiates session recording to a Raw file, and plays back > a 20-second audio sample, and then inserts the filename into a Redis > queue, then hangs up. Then the files get processed by Sevana AQuA > software in lowest CPU priority, and deleted. The files are written > into a tmpfs filesystem. > > With 2 calls per second, I get 48 simultaneous calls, and B indicates > frame drops in the audio recordings, while A shows good results. The > CPU average idle time is always above 50% on both computers. > > Why is the B side (the one receiving INVITEs) getting more frame drops > than the A (the one originating the calls)? I flipped the sides, and > then the other machine started to get more drops. > > At first I started recording in WAV format, and similar effect started > to appear at a call every 2 seconds, resulting in 12 simultaneous > calls. With raw recording, the performance is much better, but still > it's a mistery why there are dropped frames in the audio. > > The CPU is always running at its highest frequency, under > "performance" governor. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Sent from Gmail Mobile -------------- next part -------------- An HTML attachment was scrubbed... URL: From ssinyagin at gmail.com Wed Sep 20 15:53:25 2017 From: ssinyagin at gmail.com (Stanislav Sinyagin) Date: Wed, 20 Sep 2017 17:53:25 +0200 Subject: [Freeswitch-users] Session recording performance In-Reply-To: References: Message-ID: Yes, totally identical configs and software. And yes, the effect moves to the other party if I change their roles. Another hypothesis is that the calling party is sending bad RTP for some reason. I'll collect the statistics. On 20 Sep 2017 17:30, "Brandon Armstead" wrote: > Same kernel same everything? > > Are they "exact replicas" identical? > > Perhaps see if you can do this test again and receive the exact system > calls A vs B are doing and then B to A should give you further insight to > your answer. > > On Wed, Sep 20, 2017 at 6:46 AM Stanislav Sinyagin > wrote: > >> I'm building an environment for voice quality testing, and found a >> strange behavior. Maybe someone has an explanation. >> >> Two identical physical computers (PC Engines APU2), connected back to >> back (A and B). >> >> A initiates the test calls to B's SIP address, and then both ends go >> through a small dialplan (less than 100 lines of XML, similar to >> https://github.com/voxserv/fsqa/blob/master/fsqa.xml >> and start recording the session. >> >> The dialplan initiates session recording to a Raw file, and plays back >> a 20-second audio sample, and then inserts the filename into a Redis >> queue, then hangs up. Then the files get processed by Sevana AQuA >> software in lowest CPU priority, and deleted. The files are written >> into a tmpfs filesystem. >> >> With 2 calls per second, I get 48 simultaneous calls, and B indicates >> frame drops in the audio recordings, while A shows good results. The >> CPU average idle time is always above 50% on both computers. >> >> Why is the B side (the one receiving INVITEs) getting more frame drops >> than the A (the one originating the calls)? I flipped the sides, and >> then the other machine started to get more drops. >> >> At first I started recording in WAV format, and similar effect started >> to appear at a call every 2 seconds, resulting in 12 simultaneous >> calls. With raw recording, the performance is much better, but still >> it's a mistery why there are dropped frames in the audio. >> >> The CPU is always running at its highest frequency, under >> "performance" governor. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > -- > Sent from Gmail Mobile > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From ssinyagin at gmail.com Wed Sep 20 17:03:01 2017 From: ssinyagin at gmail.com (Stanislav Sinyagin) Date: Wed, 20 Sep 2017 19:03:01 +0200 Subject: [Freeswitch-users] sox L16 to G711, inconsistent results Message-ID: In order to utilize mod_native_file, I converted the original L16 8000Hz audio into G711 with sox: sox ${SRCFILE} -t raw -r 8k -b 8 -c 1 -e u-law ${DSTDIR}/${BASENAME}.PCMU I just discovered that if I do this command twice and compare resulting files, they differ. They also differ if I take 44100Hz source audio. Anyone got a clue what's going on? FreeSWITCH is always producing the same bitstream when converting L16 to G711. From ssinyagin at gmail.com Wed Sep 20 17:38:43 2017 From: ssinyagin at gmail.com (Stanislav Sinyagin) Date: Wed, 20 Sep 2017 19:38:43 +0200 Subject: [Freeswitch-users] Session recording performance In-Reply-To: References: Message-ID: RTP flows seem to be ok so far. Also using mod_native_file did not help the situation, although it improved the performance. On Wed, Sep 20, 2017 at 5:53 PM, Stanislav Sinyagin wrote: > Yes, totally identical configs and software. And yes, the effect moves to > the other party if I change their roles. > > Another hypothesis is that the calling party is sending bad RTP for some > reason. I'll collect the statistics. > > > > > On 20 Sep 2017 17:30, "Brandon Armstead" wrote: >> >> Same kernel same everything? >> >> Are they "exact replicas" identical? >> >> Perhaps see if you can do this test again and receive the exact system >> calls A vs B are doing and then B to A should give you further insight to >> your answer. >> >> On Wed, Sep 20, 2017 at 6:46 AM Stanislav Sinyagin >> wrote: >>> >>> I'm building an environment for voice quality testing, and found a >>> strange behavior. Maybe someone has an explanation. >>> >>> Two identical physical computers (PC Engines APU2), connected back to >>> back (A and B). >>> >>> A initiates the test calls to B's SIP address, and then both ends go >>> through a small dialplan (less than 100 lines of XML, similar to >>> https://github.com/voxserv/fsqa/blob/master/fsqa.xml >>> and start recording the session. >>> >>> The dialplan initiates session recording to a Raw file, and plays back >>> a 20-second audio sample, and then inserts the filename into a Redis >>> queue, then hangs up. Then the files get processed by Sevana AQuA >>> software in lowest CPU priority, and deleted. The files are written >>> into a tmpfs filesystem. >>> >>> With 2 calls per second, I get 48 simultaneous calls, and B indicates >>> frame drops in the audio recordings, while A shows good results. The >>> CPU average idle time is always above 50% on both computers. >>> >>> Why is the B side (the one receiving INVITEs) getting more frame drops >>> than the A (the one originating the calls)? I flipped the sides, and >>> then the other machine started to get more drops. >>> >>> At first I started recording in WAV format, and similar effect started >>> to appear at a call every 2 seconds, resulting in 12 simultaneous >>> calls. With raw recording, the performance is much better, but still >>> it's a mistery why there are dropped frames in the audio. >>> >>> The CPU is always running at its highest frequency, under >>> "performance" governor. >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> -- >> Sent from Gmail Mobile >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org From joel at gogii.net Wed Sep 20 18:29:39 2017 From: joel at gogii.net (Joel Serrano) Date: Wed, 20 Sep 2017 11:29:39 -0700 Subject: [Freeswitch-users] FreeSWITCH Channel Variables In-Reply-To: <6b0742ef919f48f1988c68ff7429ade0@iv66.net> References: <6b0742ef919f48f1988c68ff7429ade0@iv66.net> Message-ID: Check out: https://freeswitch.org/confluence/display/FREESWITCH/Channel+Variables#ChannelVariables-InfoApplicationVariableNames(variable_xxxx) On Tue, Sep 19, 2017 at 10:39 PM, Harry.Z wrote: > Dear All, > > > > FreeSWITCH Version 1.6.17-34-0fc0946~64bit (-34-0fc0946 64bit) > > > > How can I get below channel variables in dialplan? > > > > Caller-Callee-ID-Number: > > Other-Leg-Callee-ID-Number: > > Other-Leg-Destination-Number: > > variable_last_sent_callee_id_number: > > > > Thanks! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From ssinyagin at gmail.com Wed Sep 20 21:15:44 2017 From: ssinyagin at gmail.com (Stanislav Sinyagin) Date: Wed, 20 Sep 2017 23:15:44 +0200 Subject: [Freeswitch-users] Session recording performance In-Reply-To: References: Message-ID: I added and the drops reduced significantly (still testing to see if they're gone completely). So, probably the hardware, or NIC driver have added some jitter that caused the drops. On Wed, Sep 20, 2017 at 7:38 PM, Stanislav Sinyagin wrote: > RTP flows seem to be ok so far. Also using mod_native_file did not > help the situation, although it improved the performance. > > > > On Wed, Sep 20, 2017 at 5:53 PM, Stanislav Sinyagin wrote: >> Yes, totally identical configs and software. And yes, the effect moves to >> the other party if I change their roles. >> >> Another hypothesis is that the calling party is sending bad RTP for some >> reason. I'll collect the statistics. >> >> >> >> >> On 20 Sep 2017 17:30, "Brandon Armstead" wrote: >>> >>> Same kernel same everything? >>> >>> Are they "exact replicas" identical? >>> >>> Perhaps see if you can do this test again and receive the exact system >>> calls A vs B are doing and then B to A should give you further insight to >>> your answer. >>> >>> On Wed, Sep 20, 2017 at 6:46 AM Stanislav Sinyagin >>> wrote: >>>> >>>> I'm building an environment for voice quality testing, and found a >>>> strange behavior. Maybe someone has an explanation. >>>> >>>> Two identical physical computers (PC Engines APU2), connected back to >>>> back (A and B). >>>> >>>> A initiates the test calls to B's SIP address, and then both ends go >>>> through a small dialplan (less than 100 lines of XML, similar to >>>> https://github.com/voxserv/fsqa/blob/master/fsqa.xml >>>> and start recording the session. >>>> >>>> The dialplan initiates session recording to a Raw file, and plays back >>>> a 20-second audio sample, and then inserts the filename into a Redis >>>> queue, then hangs up. Then the files get processed by Sevana AQuA >>>> software in lowest CPU priority, and deleted. The files are written >>>> into a tmpfs filesystem. >>>> >>>> With 2 calls per second, I get 48 simultaneous calls, and B indicates >>>> frame drops in the audio recordings, while A shows good results. The >>>> CPU average idle time is always above 50% on both computers. >>>> >>>> Why is the B side (the one receiving INVITEs) getting more frame drops >>>> than the A (the one originating the calls)? I flipped the sides, and >>>> then the other machine started to get more drops. >>>> >>>> At first I started recording in WAV format, and similar effect started >>>> to appear at a call every 2 seconds, resulting in 12 simultaneous >>>> calls. With raw recording, the performance is much better, but still >>>> it's a mistery why there are dropped frames in the audio. >>>> >>>> The CPU is always running at its highest frequency, under >>>> "performance" governor. >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> -- >>> Sent from Gmail Mobile >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org From joelists at tm.net.uk Wed Sep 20 22:36:50 2017 From: joelists at tm.net.uk (Joseph Waite) Date: Wed, 20 Sep 2017 23:36:50 +0100 Subject: [Freeswitch-users] Routing calls to registered user. Message-ID: <995A631E-8C13-4E34-9735-806E20C5FA6C@tm.net.uk> Hi Guys Knocking my head against this. I am trying to route calls to registered endpoints. But it is not working. The log of the whole call is as follows 2017-09-20 23:31:49.230782 [NOTICE] switch_channel.c:1104 New Channel sofia/external/07966677711 at 185.8.92.3 [a923b79f-e080-416c-934c-af13b570f813] 2017-09-20 23:31:49.230782 [DEBUG] switch_core_state_machine.c:584 (sofia/external/07966677711 at 185.8.92.3) Running State Change CS_NEW (Cur 1 Tot 16430) 2017-09-20 23:31:49.230782 [DEBUG] sofia.c:9873 sofia/external/07966677711 at 185.8.92.3 receiving invite from 185.8.92.3:5062 version: 1.6.19 git febfb38 2017-09-14 23:14:02Z 64bit 2017-09-20 23:31:49.270606 [DEBUG] sofia.c:7084 Channel sofia/external/07966677711 at 185.8.92.3 entering state [received][100] 2017-09-20 23:31:49.270606 [DEBUG] sofia.c:7094 Remote SDP: v=0 o=FreeSWITCH 1505928870 1505928871 IN IP4 185.8.92.3 s=FreeSWITCH c=IN IP4 185.8.92.3 t=0 0 m=audio 17854 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 2017-09-20 23:31:49.270606 [DEBUG] sofia.c:7486 (sofia/external/07966677711 at 185.8.92.3) State Change CS_NEW -> CS_INIT 2017-09-20 23:31:49.270606 [DEBUG] switch_core_state_machine.c:603 (sofia/external/07966677711 at 185.8.92.3) State NEW 2017-09-20 23:31:49.270606 [DEBUG] switch_core_state_machine.c:584 (sofia/external/07966677711 at 185.8.92.3) Running State Change CS_INIT (Cur 1 Tot 16430) 2017-09-20 23:31:49.270606 [DEBUG] switch_core_state_machine.c:627 (sofia/external/07966677711 at 185.8.92.3) State INIT 2017-09-20 23:31:49.270606 [DEBUG] mod_sofia.c:90 sofia/external/07966677711 at 185.8.92.3 SOFIA INIT 2017-09-20 23:31:49.270606 [DEBUG] switch_core_state_machine.c:40 sofia/external/07966677711 at 185.8.92.3 Standard INIT 2017-09-20 23:31:49.270606 [DEBUG] switch_core_state_machine.c:48 (sofia/external/07966677711 at 185.8.92.3) State Change CS_INIT -> CS_ROUTING 2017-09-20 23:31:49.270606 [DEBUG] switch_core_state_machine.c:627 (sofia/external/07966677711 at 185.8.92.3) State INIT going to sleep 2017-09-20 23:31:49.270606 [DEBUG] switch_core_state_machine.c:584 (sofia/external/07966677711 at 185.8.92.3) Running State Change CS_ROUTING (Cur 1 Tot 16430) 2017-09-20 23:31:49.270606 [DEBUG] switch_channel.c:2249 (sofia/external/07966677711 at 185.8.92.3) Callstate Change DOWN -> RINGING 2017-09-20 23:31:49.270606 [DEBUG] switch_core_state_machine.c:643 (sofia/external/07966677711 at 185.8.92.3) State ROUTING 2017-09-20 23:31:49.270606 [DEBUG] mod_sofia.c:143 sofia/external/07966677711 at 185.8.92.3 SOFIA ROUTING 2017-09-20 23:31:49.270606 [ERR] mod_xml_radius.c:933 Result of true match: 185.35.229.6 == ^185\.35\.229\.30 2017-09-20 23:31:49.270606 [ERR] mod_xml_radius.c:930 Didn't match: inbound == ^outbound 2017-09-20 23:31:49.270606 [DEBUG] switch_core_state_machine.c:236 sofia/external/07966677711 at 185.8.92.3 Standard ROUTING 2017-09-20 23:31:49.270606 [INFO] mod_dialplan_xml.c:637 Processing 07966677711 <07966677711>->443307881011 in context public Dialplan: sofia/external/07966677711 at 185.8.92.3 parsing [public->unloop] continue=false Dialplan: sofia/external/07966677711 at 185.8.92.3 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/external/07966677711 at 185.8.92.3 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/external/07966677711 at 185.8.92.3 parsing [public->outside_call] continue=true Dialplan: sofia/external/07966677711 at 185.8.92.3 Absolute Condition [outside_call] Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(outside_call=true) Dialplan: sofia/external/07966677711 at 185.8.92.3 Action export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) Dialplan: sofia/external/07966677711 at 185.8.92.3 parsing [public->call_debug] continue=true Dialplan: sofia/external/07966677711 at 185.8.92.3 Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/external/07966677711 at 185.8.92.3 parsing [public->rejections] continue=false Dialplan: sofia/external/07966677711 at 185.8.92.3 Regex (FAIL) [rejections] ${radius_auth_result}() =~ /2/ break=on-false Dialplan: sofia/external/07966677711 at 185.8.92.3 parsing [public->timedouts] continue=false Dialplan: sofia/external/07966677711 at 185.8.92.3 Regex (FAIL) [timedouts] ${radius_auth_result}() =~ /1/ break=on-false Dialplan: sofia/external/07966677711 at 185.8.92.3 parsing [public->JeraSoft VCS Routing] continue=false Dialplan: sofia/external/07966677711 at 185.8.92.3 Regex (PASS) [JeraSoft VCS Routing] destination_number(443307881011) =~ /^(.+)$/ break=on-false Dialplan: sofia/external/07966677711 at 185.8.92.3 Action export(nolocal:h323-call-origin=originate) Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(sip_h_X-accountcode=${accountcode}) Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(call_direction=outbound) Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(hangup_after_bridge=true) Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(continue_on_fail=true) Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(inherit_codec=true) Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(call_timeout=20) Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(fail_on_single_reject=USER_BUSY) Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(origination_caller_id_name=${sip_req_user}) Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(origination_caller_id_number=${sip_from_user}) Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(SRCGWIP=src-gw-ip=${network_addr}) INLINE EXECUTE sofia/external/07966677711 at 185.8.92.3 set(SRCGWIP=src-gw-ip=185.8.92.3) 2017-09-20 23:31:49.270606 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [SRCGWIP]=[src-gw-ip=185.8.92.3] Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(CALLID=h323-conf-id=${uuid}) INLINE EXECUTE sofia/external/07966677711 at 185.8.92.3 set(CALLID=h323-conf-id=a923b79f-e080-416c-934c-af13b570f813) 2017-09-20 23:31:49.270606 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [CALLID]=[h323-conf-id=a923b79f-e080-416c-934c-af13b570f813] Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(CALLINGNUMBER=${caller_id_number}) INLINE EXECUTE sofia/external/07966677711 at 185.8.92.3 set(CALLINGNUMBER=07966677711) 2017-09-20 23:31:49.270606 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [CALLINGNUMBER]=[07966677711] Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(CALLEDNUMBER=${destination_number}) INLINE EXECUTE sofia/external/07966677711 at 185.8.92.3 set(CALLEDNUMBER=443307881011) 2017-09-20 23:31:49.270606 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [CALLEDNUMBER]=[443307881011] Dialplan: sofia/external/07966677711 at 185.8.92.3 Action auth_function(in ${DIALED_NUMBER}, in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT) Dialplan: sofia/external/07966677711 at 185.8.92.3 Action export(credit_time=${credit_time}) Dialplan: sofia/external/07966677711 at 185.8.92.3 Action export(cisco_service_info=${cisco_service_info) Dialplan: sofia/external/07966677711 at 185.8.92.3 Action export(sip_redirect_context=redirect) Dialplan: sofia/external/07966677711 at 185.8.92.3 Action transfer(443307881011 XML ipauth) Dialplan: sofia/external/07966677711 at 185.8.92.3 Action hangup(${bridge_hangup_cause}) 2017-09-20 23:31:49.270606 [DEBUG] switch_core_state_machine.c:286 (sofia/external/07966677711 at 185.8.92.3) State Change CS_ROUTING -> CS_EXECUTE 2017-09-20 23:31:49.270606 [DEBUG] switch_core_state_machine.c:643 (sofia/external/07966677711 at 185.8.92.3) State ROUTING going to sleep 2017-09-20 23:31:49.270606 [DEBUG] switch_core_state_machine.c:584 (sofia/external/07966677711 at 185.8.92.3) Running State Change CS_EXECUTE (Cur 1 Tot 16430) 2017-09-20 23:31:49.270606 [DEBUG] switch_core_state_machine.c:650 (sofia/external/07966677711 at 185.8.92.3) State EXECUTE 2017-09-20 23:31:49.270606 [DEBUG] mod_sofia.c:198 sofia/external/07966677711 at 185.8.92.3 SOFIA EXECUTE 2017-09-20 23:31:49.270606 [DEBUG] switch_core_state_machine.c:328 sofia/external/07966677711 at 185.8.92.3 Standard EXECUTE EXECUTE sofia/external/07966677711 at 185.8.92.3 set(outside_call=true) 2017-09-20 23:31:49.270606 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [outside_call]=[true] EXECUTE sofia/external/07966677711 at 185.8.92.3 export(RFC2822_DATE=Wed, 20 Sep 2017 23:31:49 +0100) 2017-09-20 23:31:49.270606 [DEBUG] switch_channel.c:1296 EXPORT (export_vars) [RFC2822_DATE]=[Wed, 20 Sep 2017 23:31:49 +0100] EXECUTE sofia/external/07966677711 at 185.8.92.3 export(nolocal:h323-call-origin=originate) 2017-09-20 23:31:49.270606 [DEBUG] switch_channel.c:1296 EXPORT (export_vars) (REMOTE ONLY) [h323-call-origin]=[originate] EXECUTE sofia/external/07966677711 at 185.8.92.3 set(sip_h_X-accountcode=) 2017-09-20 23:31:49.270606 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [sip_h_X-accountcode]=[UNDEF] EXECUTE sofia/external/07966677711 at 185.8.92.3 set(call_direction=outbound) 2017-09-20 23:31:49.270606 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [call_direction]=[outbound] EXECUTE sofia/external/07966677711 at 185.8.92.3 set(hangup_after_bridge=true) 2017-09-20 23:31:49.270606 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [hangup_after_bridge]=[true] EXECUTE sofia/external/07966677711 at 185.8.92.3 set(continue_on_fail=true) 2017-09-20 23:31:49.270606 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [continue_on_fail]=[true] EXECUTE sofia/external/07966677711 at 185.8.92.3 set(inherit_codec=true) 2017-09-20 23:31:49.270606 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [inherit_codec]=[true] EXECUTE sofia/external/07966677711 at 185.8.92.3 set(call_timeout=20) 2017-09-20 23:31:49.270606 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [call_timeout]=[20] EXECUTE sofia/external/07966677711 at 185.8.92.3 set(fail_on_single_reject=USER_BUSY) 2017-09-20 23:31:49.270606 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [fail_on_single_reject]=[USER_BUSY] EXECUTE sofia/external/07966677711 at 185.8.92.3 set(origination_caller_id_name=443307881011) 2017-09-20 23:31:49.270606 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [origination_caller_id_name]=[443307881011] EXECUTE sofia/external/07966677711 at 185.8.92.3 set(origination_caller_id_number=07966677711) 2017-09-20 23:31:49.270606 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [origination_caller_id_number]=[07966677711] EXECUTE sofia/external/07966677711 at 185.8.92.3 auth_function(in , in 07966677711, in , out AUTH_RESULT) 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:301 allocate initial structure. 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:313 initialzed configuration. 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:318 set authserver := 185.35.229.30:1812:h4nn4h. 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:318 set dictionary := /usr/local/freeswitch/conf/dictionaries/dictionary. 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:318 set seqfile := /var/run/radius.seq. 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:318 set mapfile := /usr/local/etc/radiusclient/port-id-map. 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:318 set default_realm := . 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:318 set radius_timeout := 3. 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:318 set radius_retries := 2. 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:318 set radius_deadtime := 0. 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:318 set bindaddr := 185.35.229.6. 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:371 ... radius: User-Name: 07966677711 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:380 ... radius: User-Password: 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:391 ... radius: Called-station-Id is empty, ignoring... 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:413 Handle attribute: h323-conf-id 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:439 ... dictionary data: id:24, vendor id:9, attr type:0, attr name:h323-conf-id (589848) 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:444 ... radius: key:h323-conf-id, value:CALLID (h323-conf-id=a923b79f-e080-416c-934c-af13b570f813) as string 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:413 Handle attribute: Cisco-AVPair 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:439 ... dictionary data: id:1, vendor id:9, attr type:0, attr name:Cisco-AVPair (589825) 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:444 ... radius: key:Cisco-AVPair, value:SRCGWIP (src-gw-ip=185.8.92.3) as string 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:413 Handle attribute: Cisco-AVPair 2017-09-20 23:31:49.270606 [ERR] mod_rad_auth.c:178 Undefined channel variable: SRCGWNAME. 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:439 ... dictionary data: id:1, vendor id:9, attr type:0, attr name:Cisco-AVPair (589825) 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:444 ... radius: key:Cisco-AVPair, value:SRCGWNAME () as string 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:413 Handle attribute: Cisco-AVPair 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:439 ... dictionary data: id:1, vendor id:9, attr type:0, attr name:Cisco-AVPair (589825) 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:444 ... radius: key:Cisco-AVPair, value:request-type=number (request-type=number) as string 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:413 Handle attribute: Calling-Station-Id 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:439 ... dictionary data: id:31, vendor id:0, attr type:0, attr name:Calling-Station-Id (31) 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:444 ... radius: key:Calling-Station-Id, value:CALLINGNUMBER (07966677711) as string 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:413 Handle attribute: Called-Station-Id 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:439 ... dictionary data: id:30, vendor id:0, attr type:0, attr name:Called-Station-Id (30) 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:444 ... radius: key:Called-Station-Id, value:CALLEDNUMBER (443307881011) as string 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:491 sending radius packet ... 2017-09-20 23:31:49.290658 [DEBUG] mod_rad_auth.c:497 RADIUS Authentication OK 2017-09-20 23:31:49.290658 [DEBUG] mod_rad_auth.c:519 Handle attribute: CREDIT_TIME 2017-09-20 23:31:49.290658 [DEBUG] mod_rad_auth.c:522 attribute (CREDIT_TIME) found in radius packet 2017-09-20 23:31:49.290658 [DEBUG] mod_rad_auth.c:523 set variable credit_time := 7199 2017-09-20 23:31:49.290658 [DEBUG] mod_rad_auth.c:519 Handle attribute: CISCO_SERVICE_INFO 2017-09-20 23:31:49.290658 [DEBUG] mod_rad_auth.c:522 attribute (CISCO_SERVICE_INFO) found in radius packet 2017-09-20 23:31:49.290658 [DEBUG] mod_rad_auth.c:523 set variable cisco_service_info := 200 2017-09-20 23:31:49.290658 [DEBUG] mod_rad_auth.c:519 Handle attribute: RADIUS_RETURN_CODE 2017-09-20 23:31:49.290658 [DEBUG] mod_rad_auth.c:522 attribute (RADIUS_RETURN_CODE) found in radius packet 2017-09-20 23:31:49.290658 [DEBUG] mod_rad_auth.c:523 set variable return_code := 16 EXECUTE sofia/external/07966677711 at 185.8.92.3 export(credit_time=7199) 2017-09-20 23:31:49.290658 [DEBUG] switch_channel.c:1296 EXPORT (export_vars) [credit_time]=[7199] EXECUTE sofia/external/07966677711 at 185.8.92.3 export(cisco_service_info=200) 2017-09-20 23:31:49.290658 [DEBUG] switch_channel.c:1296 EXPORT (export_vars) [cisco_service_info]=[200] EXECUTE sofia/external/07966677711 at 185.8.92.3 export(sip_redirect_context=redirect) 2017-09-20 23:31:49.290658 [DEBUG] switch_channel.c:1296 EXPORT (export_vars) [sip_redirect_context]=[redirect] EXECUTE sofia/external/07966677711 at 185.8.92.3 transfer(443307881011 XML ipauth) 2017-09-20 23:31:49.290658 [DEBUG] switch_ivr.c:2165 (sofia/external/07966677711 at 185.8.92.3) State Change CS_EXECUTE -> CS_ROUTING 2017-09-20 23:31:49.290658 [NOTICE] switch_ivr.c:2172 Transfer sofia/external/07966677711 at 185.8.92.3 to XML[443307881011 at ipauth] 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:650 (sofia/external/07966677711 at 185.8.92.3) State EXECUTE going to sleep 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:584 (sofia/external/07966677711 at 185.8.92.3) Running State Change CS_ROUTING (Cur 1 Tot 16430) 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:643 (sofia/external/07966677711 at 185.8.92.3) State ROUTING 2017-09-20 23:31:49.290658 [DEBUG] mod_sofia.c:143 sofia/external/07966677711 at 185.8.92.3 SOFIA ROUTING 2017-09-20 23:31:49.290658 [ERR] mod_xml_radius.c:933 Result of true match: 185.35.229.6 == ^185\.35\.229\.30 2017-09-20 23:31:49.290658 [ERR] mod_xml_radius.c:930 Didn't match: inbound == ^outbound 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:236 sofia/external/07966677711 at 185.8.92.3 Standard ROUTING 2017-09-20 23:31:49.290658 [INFO] mod_dialplan_xml.c:637 Processing 07966677711 <07966677711>->443307881011 in context ipauth Dialplan: sofia/external/07966677711 at 185.8.92.3 parsing [ipauth->Auth Calls] continue=false Dialplan: sofia/external/07966677711 at 185.8.92.3 Regex (PASS) [Auth Calls] ${cisco_service_info}(200) =~ /^200$/ break=never Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(execute_on_answer=sched_hangup +${credit_time} alloted_timeout) Dialplan: sofia/external/07966677711 at 185.8.92.3 Action bridge({sip_invite_from_uri=sip:${sip_from_user}@${sip_network_ip}}sofia/external/${destination_number}@185.35.229.30:5060) Dialplan: sofia/external/07966677711 at 185.8.92.3 Action hangup(${bridge_hangup_cause}) Dialplan: sofia/external/07966677711 at 185.8.92.3 Regex (FAIL) [Auth Calls] ${cisco_service_info}(200) =~ /^101$/ break=on-true Dialplan: sofia/external/07966677711 at 185.8.92.3 Regex (FAIL) [Auth Calls] ${cisco_service_info}(200) =~ /^221$/ break=on-true Dialplan: sofia/external/07966677711 at 185.8.92.3 Regex (FAIL) [Auth Calls] ${cisco_service_info}(200) =~ /^201$/ break=on-true Dialplan: sofia/external/07966677711 at 185.8.92.3 Absolute Condition [Auth Calls] Dialplan: sofia/external/07966677711 at 185.8.92.3 Action log(INFO Were in the IPauth Context!!) Dialplan: sofia/external/07966677711 at 185.8.92.3 Action log(INFO credit_time=${credit_time}) Dialplan: sofia/external/07966677711 at 185.8.92.3 Action log(INFO cisco_service_info=${cisco_service_info}) 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:286 (sofia/external/07966677711 at 185.8.92.3) State Change CS_ROUTING -> CS_EXECUTE 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:643 (sofia/external/07966677711 at 185.8.92.3) State ROUTING going to sleep 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:584 (sofia/external/07966677711 at 185.8.92.3) Running State Change CS_EXECUTE (Cur 1 Tot 16430) 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:650 (sofia/external/07966677711 at 185.8.92.3) State EXECUTE 2017-09-20 23:31:49.290658 [DEBUG] mod_sofia.c:198 sofia/external/07966677711 at 185.8.92.3 SOFIA EXECUTE 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:328 sofia/external/07966677711 at 185.8.92.3 Standard EXECUTE EXECUTE sofia/external/07966677711 at 185.8.92.3 set(execute_on_answer=sched_hangup +7199 alloted_timeout) 2017-09-20 23:31:49.290658 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [execute_on_answer]=[sched_hangup +7199 alloted_timeout] EXECUTE sofia/external/07966677711 at 185.8.92.3 bridge({sip_invite_from_uri=sip:07966677711 at 185.8.92.3}sofia/external/443307881011 at 185.35.229.30:5060) 2017-09-20 23:31:49.290658 [DEBUG] switch_channel.c:1250 sofia/external/07966677711 at 185.8.92.3 EXPORTING[export_vars] [RFC2822_DATE]=[Wed, 20 Sep 2017 23:31:49 +0100] to event 2017-09-20 23:31:49.290658 [DEBUG] switch_channel.c:1250 sofia/external/07966677711 at 185.8.92.3 EXPORTING[export_vars] [h323-call-origin]=[originate] to event 2017-09-20 23:31:49.290658 [DEBUG] switch_channel.c:1250 sofia/external/07966677711 at 185.8.92.3 EXPORTING[export_vars] [credit_time]=[7199] to event 2017-09-20 23:31:49.290658 [DEBUG] switch_channel.c:1250 sofia/external/07966677711 at 185.8.92.3 EXPORTING[export_vars] [cisco_service_info]=[200] to event 2017-09-20 23:31:49.290658 [DEBUG] switch_channel.c:1250 sofia/external/07966677711 at 185.8.92.3 EXPORTING[export_vars] [sip_redirect_context]=[redirect] to event 2017-09-20 23:31:49.290658 [DEBUG] switch_ivr_originate.c:2142 Parsing global variables 2017-09-20 23:31:49.290658 [NOTICE] switch_channel.c:1104 New Channel sofia/external/443307881011 at 185.35.229.30:5060 [848843db-cad9-48a8-a4bd-2ac6b6b77dff] 2017-09-20 23:31:49.290658 [DEBUG] mod_sofia.c:4819 (sofia/external/443307881011 at 185.35.229.30:5060) State Change CS_NEW -> CS_INIT 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:584 (sofia/external/443307881011 at 185.35.229.30:5060) Running State Change CS_INIT (Cur 2 Tot 16431) 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:627 (sofia/external/443307881011 at 185.35.229.30:5060) State INIT 2017-09-20 23:31:49.290658 [DEBUG] mod_sofia.c:90 sofia/external/443307881011 at 185.35.229.30:5060 SOFIA INIT 2017-09-20 23:31:49.290658 [DEBUG] sofia_glue.c:1295 sofia/external/443307881011 at 185.35.229.30:5060 sending invite version: 1.6.19 git febfb38 2017-09-14 23:14:02Z 64bit Local SDP: v=0 o=FreeSWITCH 1505929783 1505929784 IN IP4 185.35.229.6 s=FreeSWITCH c=IN IP4 185.35.229.6 t=0 0 m=audio 16926 RTP/AVP 8 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 a=sendrecv 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:40 sofia/external/443307881011 at 185.35.229.30:5060 Standard INIT 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:48 (sofia/external/443307881011 at 185.35.229.30:5060) State Change CS_INIT -> CS_ROUTING 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:627 (sofia/external/443307881011 at 185.35.229.30:5060) State INIT going to sleep 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:584 (sofia/external/443307881011 at 185.35.229.30:5060) Running State Change CS_ROUTING (Cur 2 Tot 16431) 2017-09-20 23:31:49.290658 [DEBUG] sofia.c:7084 Channel sofia/external/443307881011 at 185.35.229.30:5060 entering state [calling][0] 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:643 (sofia/external/443307881011 at 185.35.229.30:5060) State ROUTING 2017-09-20 23:31:49.290658 [DEBUG] mod_sofia.c:143 sofia/external/443307881011 at 185.35.229.30:5060 SOFIA ROUTING 2017-09-20 23:31:49.290658 [DEBUG] switch_ivr_originate.c:67 (sofia/external/443307881011 at 185.35.229.30:5060) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2017-09-20 23:31:49.290658 [ERR] mod_xml_radius.c:930 Didn't match: 185.35.229.30:5060 == ^185\.35\.229\.30 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:643 (sofia/external/443307881011 at 185.35.229.30:5060) State ROUTING going to sleep 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:584 (sofia/external/443307881011 at 185.35.229.30:5060) Running State Change CS_CONSUME_MEDIA (Cur 2 Tot 16431) 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:662 (sofia/external/443307881011 at 185.35.229.30:5060) State CONSUME_MEDIA 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:662 (sofia/external/443307881011 at 185.35.229.30:5060) State CONSUME_MEDIA going to sleep 2017-09-20 23:31:49.350696 [DEBUG] sofia.c:6540 Redirect: Transfering to joehouse XML redirect 2017-09-20 23:31:49.350696 [DEBUG] switch_ivr.c:2165 (sofia/external/07966677711 at 185.8.92.3) State Change CS_EXECUTE -> CS_ROUTING 2017-09-20 23:31:49.350696 [NOTICE] switch_ivr.c:2172 Transfer sofia/external/07966677711 at 185.8.92.3 to XML[joehouse at redirect] 2017-09-20 23:31:49.350696 [NOTICE] sofia.c:6549 Hangup sofia/external/443307881011 at 185.35.229.30:5060 [CS_CONSUME_MEDIA] [REDIRECTION_TO_NEW_DESTINATION] 2017-09-20 23:31:49.350696 [DEBUG] sofia.c:1453 Channel is already hungup. 2017-09-20 23:31:49.350696 [DEBUG] sofia.c:1453 Channel is already hungup. 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:584 (sofia/external/443307881011 at 185.35.229.30:5060) Running State Change CS_HANGUP (Cur 2 Tot 16431) 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:850 (sofia/external/443307881011 at 185.35.229.30:5060) Callstate Change DOWN -> HANGUP 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:852 (sofia/external/443307881011 at 185.35.229.30:5060) State HANGUP 2017-09-20 23:31:49.350696 [DEBUG] mod_sofia.c:438 Channel sofia/external/443307881011 at 185.35.229.30:5060 hanging up, cause: REDIRECTION_TO_NEW_DESTINATION 2017-09-20 23:31:49.350696 [DEBUG] mod_sofia.c:502 Sending CANCEL to sofia/external/443307881011 at 185.35.229.30:5060 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:60 sofia/external/443307881011 at 185.35.229.30:5060 Standard HANGUP, cause: REDIRECTION_TO_NEW_DESTINATION 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:852 (sofia/external/443307881011 at 185.35.229.30:5060) State HANGUP going to sleep 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:619 (sofia/external/443307881011 at 185.35.229.30:5060) State Change CS_HANGUP -> CS_REPORTING 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:584 (sofia/external/443307881011 at 185.35.229.30:5060) Running State Change CS_REPORTING (Cur 2 Tot 16431) 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:938 (sofia/external/443307881011 at 185.35.229.30:5060) State REPORTING 2017-09-20 23:31:49.350696 [ERR] mod_xml_radius.c:930 Didn't match: 185.35.229.30:5060 == ^185\.35\.229\.30 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:174 sofia/external/443307881011 at 185.35.229.30:5060 Standard REPORTING, cause: REDIRECTION_TO_NEW_DESTINATION 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:938 (sofia/external/443307881011 at 185.35.229.30:5060) State REPORTING going to sleep 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:610 (sofia/external/443307881011 at 185.35.229.30:5060) State Change CS_REPORTING -> CS_DESTROY 2017-09-20 23:31:49.350696 [DEBUG] switch_core_session.c:1665 Session 16431 (sofia/external/443307881011 at 185.35.229.30:5060) Locked, Waiting on external entities 2017-09-20 23:31:49.350696 [DEBUG] switch_ivr_originate.c:3837 Originate Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL] 2017-09-20 23:31:49.350696 [NOTICE] switch_core_session.c:1683 Session 16431 (sofia/external/443307881011 at 185.35.229.30:5060) Ended 2017-09-20 23:31:49.350696 [NOTICE] switch_core_session.c:1687 Close Channel sofia/external/443307881011 at 185.35.229.30:5060 [CS_DESTROY] 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:741 (sofia/external/443307881011 at 185.35.229.30:5060) Running State Change CS_DESTROY (Cur 1 Tot 16431) 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:751 (sofia/external/443307881011 at 185.35.229.30:5060) State DESTROY 2017-09-20 23:31:49.350696 [DEBUG] mod_sofia.c:343 sofia/external/443307881011 at 185.35.229.30:5060 SOFIA DESTROY 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:181 sofia/external/443307881011 at 185.35.229.30:5060 Standard DESTROY 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:751 (sofia/external/443307881011 at 185.35.229.30:5060) State DESTROY going to sleep 2017-09-20 23:31:49.350696 [INFO] mod_dptools.c:3436 Originate Failed. Cause: ORIGINATOR_CANCEL 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:650 (sofia/external/07966677711 at 185.8.92.3) State EXECUTE going to sleep 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:584 (sofia/external/07966677711 at 185.8.92.3) Running State Change CS_ROUTING (Cur 1 Tot 16431) 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:643 (sofia/external/07966677711 at 185.8.92.3) State ROUTING 2017-09-20 23:31:49.350696 [DEBUG] mod_sofia.c:143 sofia/external/07966677711 at 185.8.92.3 SOFIA ROUTING 2017-09-20 23:31:49.350696 [ERR] mod_xml_radius.c:933 Result of true match: 185.35.229.6 == ^185\.35\.229\.30 2017-09-20 23:31:49.350696 [ERR] mod_xml_radius.c:930 Didn't match: inbound == ^outbound 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:236 sofia/external/07966677711 at 185.8.92.3 Standard ROUTING 2017-09-20 23:31:49.350696 [INFO] mod_dialplan_xml.c:637 Processing 07966677711 <07966677711>->joehouse in context redirect Dialplan: sofia/external/07966677711 at 185.8.92.3 parsing [redirect->Redirect Calls] continue=false Dialplan: sofia/external/07966677711 at 185.8.92.3 Regex (PASS) [Redirect Calls] ${sip_redirect_contact_0}(;src_number=07966677711;q=1.00) =~ /^ CS_EXECUTE 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:643 (sofia/external/07966677711 at 185.8.92.3) State ROUTING going to sleep 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:584 (sofia/external/07966677711 at 185.8.92.3) Running State Change CS_EXECUTE (Cur 1 Tot 16431) 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:650 (sofia/external/07966677711 at 185.8.92.3) State EXECUTE 2017-09-20 23:31:49.350696 [DEBUG] mod_sofia.c:198 sofia/external/07966677711 at 185.8.92.3 SOFIA EXECUTE 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:328 sofia/external/07966677711 at 185.8.92.3 Standard EXECUTE EXECUTE sofia/external/07966677711 at 185.8.92.3 set(process_cdr=true) 2017-09-20 23:31:49.350696 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [process_cdr]=[true] EXECUTE sofia/external/07966677711 at 185.8.92.3 log(ALERT Local User) 2017-09-20 23:31:49.350696 [ALERT] mod_dptools.c:1742 Local User EXECUTE sofia/external/07966677711 at 185.8.92.3 set(accountcode=jerasoftoutbound) 2017-09-20 23:31:49.350696 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [accountcode]=[jerasoftoutbound] EXECUTE sofia/external/07966677711 at 185.8.92.3 log(ALERT joehouse) 2017-09-20 23:31:49.350696 [ALERT] mod_dptools.c:1742 joehouse EXECUTE sofia/external/07966677711 at 185.8.92.3 bridge(user/joehouse at sip.biznetuk.com) 2017-09-20 23:31:49.350696 [DEBUG] switch_channel.c:1250 sofia/external/07966677711 at 185.8.92.3 EXPORTING[export_vars] [RFC2822_DATE]=[Wed, 20 Sep 2017 23:31:49 +0100] to event 2017-09-20 23:31:49.350696 [DEBUG] switch_channel.c:1250 sofia/external/07966677711 at 185.8.92.3 EXPORTING[export_vars] [h323-call-origin]=[originate] to event 2017-09-20 23:31:49.350696 [DEBUG] switch_channel.c:1250 sofia/external/07966677711 at 185.8.92.3 EXPORTING[export_vars] [credit_time]=[7199] to event 2017-09-20 23:31:49.350696 [DEBUG] switch_channel.c:1250 sofia/external/07966677711 at 185.8.92.3 EXPORTING[export_vars] [cisco_service_info]=[200] to event 2017-09-20 23:31:49.350696 [DEBUG] switch_channel.c:1250 sofia/external/07966677711 at 185.8.92.3 EXPORTING[export_vars] [sip_redirect_context]=[redirect] to event 2017-09-20 23:31:49.350696 [DEBUG] switch_ivr_originate.c:2142 Parsing global variables 2017-09-20 23:31:49.350696 [WARNING] mod_dptools.c:4184 Can't find user [joehouse at sip.biznetuk.com] 2017-09-20 23:31:49.350696 [NOTICE] switch_ivr_originate.c:2851 Cannot create outgoing channel of type [user] cause: [SUBSCRIBER_ABSENT] 2017-09-20 23:31:49.350696 [DEBUG] switch_ivr_originate.c:3848 Originate Resulted in Error Cause: 20 [SUBSCRIBER_ABSENT] 2017-09-20 23:31:49.350696 [INFO] mod_dptools.c:3436 Originate Failed. Cause: SUBSCRIBER_ABSENT 2017-09-20 23:31:49.350696 [NOTICE] switch_core_state_machine.c:385 sofia/external/07966677711 at 185.8.92.3 has executed the last dialplan instruction, hanging up. 2017-09-20 23:31:49.350696 [NOTICE] switch_core_state_machine.c:387 Hangup sofia/external/07966677711 at 185.8.92.3 [CS_EXECUTE] [NORMAL_CLEARING] 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:650 (sofia/external/07966677711 at 185.8.92.3) State EXECUTE going to sleep 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:584 (sofia/external/07966677711 at 185.8.92.3) Running State Change CS_HANGUP (Cur 1 Tot 16431) 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:850 (sofia/external/07966677711 at 185.8.92.3) Callstate Change RINGING -> HANGUP 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:852 (sofia/external/07966677711 at 185.8.92.3) State HANGUP 2017-09-20 23:31:49.350696 [DEBUG] mod_sofia.c:438 Channel sofia/external/07966677711 at 185.8.92.3 hanging up, cause: NORMAL_CLEARING 2017-09-20 23:31:49.350696 [DEBUG] mod_sofia.c:577 Responding to INVITE with: 480 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:60 sofia/external/07966677711 at 185.8.92.3 Standard HANGUP, cause: NORMAL_CLEARING 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:852 (sofia/external/07966677711 at 185.8.92.3) State HANGUP going to sleep 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:619 (sofia/external/07966677711 at 185.8.92.3) State Change CS_HANGUP -> CS_REPORTING 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:584 (sofia/external/07966677711 at 185.8.92.3) Running State Change CS_REPORTING (Cur 1 Tot 16431) 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:938 (sofia/external/07966677711 at 185.8.92.3) State REPORTING 2017-09-20 23:31:49.350696 [ERR] mod_xml_radius.c:933 Result of true match: 185.35.229.6 == ^185\.35\.229\.30 2017-09-20 23:31:49.350696 [ERR] mod_xml_radius.c:930 Didn't match: inbound == ^outbound 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:174 sofia/external/07966677711 at 185.8.92.3 Standard REPORTING, cause: NORMAL_CLEARING 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:938 (sofia/external/07966677711 at 185.8.92.3) State REPORTING going to sleep 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:610 (sofia/external/07966677711 at 185.8.92.3) State Change CS_REPORTING -> CS_DESTROY 2017-09-20 23:31:49.350696 [DEBUG] switch_core_session.c:1665 Session 16430 (sofia/external/07966677711 at 185.8.92.3) Locked, Waiting on external entities 2017-09-20 23:31:49.350696 [NOTICE] switch_core_session.c:1683 Session 16430 (sofia/external/07966677711 at 185.8.92.3) Ended 2017-09-20 23:31:49.350696 [NOTICE] switch_core_session.c:1687 Close Channel sofia/external/07966677711 at 185.8.92.3 [CS_DESTROY] 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:741 (sofia/external/07966677711 at 185.8.92.3) Running State Change CS_DESTROY (Cur 0 Tot 16431) 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:751 (sofia/external/07966677711 at 185.8.92.3) State DESTROY 2017-09-20 23:31:49.350696 [DEBUG] mod_sofia.c:343 sofia/external/07966677711 at 185.8.92.3 SOFIA DESTROY 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:181 sofia/external/07966677711 at 185.8.92.3 Standard DESTROY 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:751 (sofia/external/07966677711 at 185.8.92.3) State DESTROY going to sleep its saying there is no endpoint registered yet the output of show registrations shows there is freeswitch at BiznetukSBC1> show registrations reg_user,realm,token,url,expires,network_ip,network_port,network_proto,hostname,metadata,registration_uuid joehouse,sip.biznetuk.com,1030250472 at 192_168_1_101,sofia/internal/sip:joehouse at 185.35.230.2:5382,1505947110,185.35.230.2,5382,udp,BiznetukSBC1,,60484000-aa2d-4b21-9d58-11e7d4155f1e 1 total. Any ideas? -------------- next part -------------- An HTML attachment was scrubbed... URL: From Harry.Z at iv66.net Thu Sep 21 01:36:41 2017 From: Harry.Z at iv66.net (Harry.Z) Date: Thu, 21 Sep 2017 01:36:41 +0000 Subject: [Freeswitch-users] =?utf-8?b?562U5aSNOiAgRnJlZVNXSVRDSCBDaGFubmVs?= =?utf-8?q?_Variables?= In-Reply-To: References: <6b0742ef919f48f1988c68ff7429ade0@iv66.net> Message-ID: I can’t find these variables in this link. That means I can’t use these variables? 发件人: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] 代表 Joel Serrano 发送时间: 2017年9月21日 2:30 收件人: FreeSWITCH Users Help 主题: Re: [Freeswitch-users] FreeSWITCH Channel Variables Check out: https://freeswitch.org/confluence/display/FREESWITCH/Channel+Variables#ChannelVariables-InfoApplicationVariableNames(variable_xxxx) On Tue, Sep 19, 2017 at 10:39 PM, Harry.Z > wrote: Dear All, FreeSWITCH Version 1.6.17-34-0fc0946~64bit (-34-0fc0946 64bit) How can I get below channel variables in dialplan? Caller-Callee-ID-Number: Other-Leg-Callee-ID-Number: Other-Leg-Destination-Number: variable_last_sent_callee_id_number: Thanks! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From jungleboogie0 at gmail.com Thu Sep 21 01:43:40 2017 From: jungleboogie0 at gmail.com (jungle Boogie) Date: Wed, 20 Sep 2017 18:43:40 -0700 Subject: [Freeswitch-users] Session recording performance In-Reply-To: References: Message-ID: What occurs if you send identical audio for the same duration? IOW, play the sound file for each call. -------------- next part -------------- An HTML attachment was scrubbed... URL: From donguyenha at gmail.com Thu Sep 21 01:52:49 2017 From: donguyenha at gmail.com (Do Nguyen Ha) Date: Thu, 21 Sep 2017 08:52:49 +0700 Subject: [Freeswitch-users] Verto Qs In-Reply-To: <430A784F-3570-433E-A7F7-53644DD2F113@magicmail.mooo.com> References: <430A784F-3570-433E-A7F7-53644DD2F113@magicmail.mooo.com> Message-ID: I have no idea, sorry cannot help On Sep 20, 2017 20:56, "Rick Jarvis" wrote: > Thanks Ha, that makes sense. > > I’m still trying to work out how to send jsonapi commands via 8082 with > Javascript. Any ideas? > > On 20 Sep 2017, at 13:59, Do Nguyen Ha wrote: > > Hi Rick > I did get confused like you for question number 3 > 3. Difference between :7443 and :8082 > >>>> The port 7443 is used for sip over web socket protocol(RFC 7118) - > you can use command "sofia status profile internal reg", the port 8082 is > used for verto protocol- you can use command"verto status - not sure it is > right command", both ports are used for signaling > > Hope this help > Ha` > > On Sep 20, 2017 18:23, "Rick Jarvis" wrote: > > Ok I’ve asked a lot of questions about Verto, so I thought I’d summarise > my current confusions here - if anyone can give me any pointers on any of > these inline, that would be amazing! :) > > 1. Correct way to list all conference members via wss? > > 2. How to check if there is a current verto session / call in progress, > and make the call if not. > > 3. Difference between :7443 and :8082 (is one for signalling and one for > audio?) > > I have just purchased the latest FS book, 1.8, so hoping that will help > also :) > > Thanks > R > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From jeff.chua.linux at gmail.com Thu Sep 21 02:09:52 2017 From: jeff.chua.linux at gmail.com (Jeff Chua) Date: Thu, 21 Sep 2017 10:09:52 +0800 Subject: [Freeswitch-users] gsmopen one-way voice issue In-Reply-To: References: Message-ID: On Wed, Sep 20, 2017 at 6:44 PM, Giovanni Maruzzelli wrote: > try to change the interface devices that are used by gsopen, many times > modem (dongle) vendor are messing with them. Rerea the documentation, and > check out how to change those devices, eg, in various permutations. > This is the only thing that comes to my mind. This is what I used ... And in asterisk ... audio=/dev/ttyUSB1 ; tty port for audio connection; data=/dev/ttyUSB2 ; tty port for AT commands; There are only 3 tty's listed for this device (0 1 2). Again, asterisk works on the same system using the same device. > If this do not work, you will need to debug it deeply. Looks like I an out of options, so will start the debug. How to start this? Thanks, Jeff From ssinyagin at gmail.com Thu Sep 21 05:46:02 2017 From: ssinyagin at gmail.com (Stanislav Sinyagin) Date: Thu, 21 Sep 2017 07:46:02 +0200 Subject: [Freeswitch-users] Session recording performance In-Reply-To: References: Message-ID: with jitterbuffer, it now works as expected. So, the sending party was generating extra jitter. I am playing the same audio file in both directions, and then compare the recorded audio with the original by Sevana AQuA software. This allows me to detect packet loss in a heterogeneous environment where the voice goes through RTP proxies and TDM. On Thu, Sep 21, 2017 at 3:43 AM, jungle Boogie wrote: > What occurs if you send identical audio for the same duration? IOW, play the > sound file for each call. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mirkobrankovic at gmail.com Thu Sep 21 11:53:22 2017 From: mirkobrankovic at gmail.com (Mirko Brankovic) Date: Thu, 21 Sep 2017 13:53:22 +0200 Subject: [Freeswitch-users] ICE/DTLS handshake Message-ID: HI, Has anyone experienced DTLS handshake takes 5s to get to SETUP state: > 2017-09-21 11:40:40.021178 [INFO] switch_rtp.c:3515 Changing audio DTLS > state from OFF to HANDSHAKE > 2017-09-21 11:40:45.319457 [INFO] switch_rtp.c:3172 Changing audio DTLS > state from HANDSHAKE to SETUP In network dump i see that answering side is not sending STUN for this 5s and then suddenly answers last 5 STUNs from A side. Has anyone encountered this kind of problem ? I have a pcap if necessary... Thanks, -- Regards, Mirko -------------- next part -------------- An HTML attachment was scrubbed... URL: From ssinyagin at gmail.com Thu Sep 21 12:30:30 2017 From: ssinyagin at gmail.com (Stanislav Sinyagin) Date: Thu, 21 Sep 2017 14:30:30 +0200 Subject: [Freeswitch-users] Session recording performance In-Reply-To: References: Message-ID: but the interesting thing is, that it's the device that originates the SIP call is producing a significantly higher jitter than the one receiving the call. The originating command looks like this here, with a SIP address as destination, and a number in local XML dialplan as transfer application: https://freeswitch.org/confluence/display/FREESWITCH/Originate+Example On Thu, Sep 21, 2017 at 7:46 AM, Stanislav Sinyagin wrote: > with jitterbuffer, it now works as expected. So, the sending party was > generating extra jitter. > > I am playing the same audio file in both directions, and then compare > the recorded audio with the original by Sevana AQuA software. This > allows me to detect packet loss in a heterogeneous environment where > the voice goes through RTP proxies and TDM. > > > > > On Thu, Sep 21, 2017 at 3:43 AM, jungle Boogie wrote: >> What occurs if you send identical audio for the same duration? IOW, play the >> sound file for each call. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org From vikas452 at gmail.com Wed Sep 20 13:27:22 2017 From: vikas452 at gmail.com (vikas sharma) Date: Wed, 20 Sep 2017 18:57:22 +0530 Subject: [Freeswitch-users] RTP Loss with Freeswitch in Default Mode Message-ID: Hi I am using Freeswitch in Default mode for video calls.But while i am making video calls using two linphone applications on android mobile phone with H264 codec support ,video is getting stuck in between the calls on linphone i.e video is not continuous. *aleg*:*bleg*: Here is my analysis on RTP packet dump taken on the Freeswitch server side: 1. Audio RTP received on *A leg*= *5184* but* forwarded *on *B leg *= 5005 + 35(*private port*) =* 5040*. 2. Audio RTP received on* B leg* = *5178* but forwarded on* A leg* = 4763 + 68(*private port*) = *4831*. 3. Video RTP received on* A leg* = *3145* but forwarded on* B leg* = 3050 + 77(*private port)* = *3127*. 4. Video RTP received on *B leg* = *6402* but forwarded on *A leg* = *6252*. I have also attached here the screenshot of RTP streams from wireshark. This RTP Loss is causing video call quality very bad if network is not too good.Kindly help me to improve the video call quality and reduce RTP loss. Questions: 1. Why some of the RTP packets are being forwarded to private ports? 2. Why there is so much difference in the RTP packets received on the Freeswitch and sent out from the Freeswitch Kindly let me know if i am doing something wrong. ​ -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: A-leg-Ports.png Type: image/png Size: 16487 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: B-leg-Ports.png Type: image/png Size: 15456 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: Call1.png Type: image/png Size: 18284 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: rtp_streams.png Type: image/png Size: 116425 bytes Desc: not available URL: From rmundkowsky at ets.org Thu Sep 21 14:08:58 2017 From: rmundkowsky at ets.org (Mundkowsky, Robert) Date: Thu, 21 Sep 2017 14:08:58 +0000 Subject: [Freeswitch-users] Session recording performance In-Reply-To: References: Message-ID: Just curious, have you tried switching the boxes and seeing if you get same situation? Robert -----Original Message----- From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Stanislav Sinyagin Sent: Thursday, September 21, 2017 8:31 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Session recording performance but the interesting thing is, that it's the device that originates the SIP call is producing a significantly higher jitter than the one receiving the call. The originating command looks like this here, with a SIP address as destination, and a number in local XML dialplan as transfer application: https://na01.safelinks.protection.outlook.com/?url=https%3A%2F%2Ffreeswitch.org%2Fconfluence%2Fdisplay%2FFREESWITCH%2FOriginate%2BExample&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636415940392006303&sdata=M5OEPvRrB9JkM5fodeKscJakpsvZn42LhZs2jQle9Ro%3D&reserved=0 On Thu, Sep 21, 2017 at 7:46 AM, Stanislav Sinyagin wrote: > with jitterbuffer, it now works as expected. So, the sending party was > generating extra jitter. > > I am playing the same audio file in both directions, and then compare > the recorded audio with the original by Sevana AQuA software. This > allows me to detect packet loss in a heterogeneous environment where > the voice goes through RTP proxies and TDM. > > > > > On Thu, Sep 21, 2017 at 3:43 AM, jungle Boogie wrote: >> What occurs if you send identical audio for the same duration? IOW, >> play the sound file for each call. >> >> >> _____________________________________________________________________ >> ____ Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.f >> reeswitchsolutions.com&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846 >> ba37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0% >> 7C636415940392006303&sdata=nGh9In%2FvTyg37f7nJph6TXJFaz2nMMjZ0NXzIZr3 >> hWg%3D&reserved=0 >> >> Official FreeSWITCH Sites >> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.f >> reeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba >> 852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C6364159 >> 40392006303&sdata=wFbEKkGrG49JjArYiD7D1mo6icJFqaxinnS3%2F5xq6hs%3D&re >> served=0 >> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fconfl >> uence.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba >> 37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C >> 636415940392006303&sdata=ehMZuD9vMm325yuExibRBY6G9ysSDmvqoBQaXcHXFww% >> 3D&reserved=0 >> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.c >> luecon.com&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852 >> 608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C6364159403 >> 92006303&sdata=J2p7ztQidG%2FLf4c6prqao3hhOnrmHHl2pxlwod3LNKM%3D&reser >> ved=0 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists >> .freeswitch.org%2Fmailman%2Flistinfo%2Ffreeswitch-users&data=02%7C01% >> 7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852608d500ed0775%7C0ba6e9b7 >> 60b34fae92f37e6ddd9e9b65%7C0%7C0%7C636415940392006303&sdata=kpqcuibvT >> vN3zMcS54w5CemHeLiKgj0OrMMTq1xiMUo%3D&reserved=0 >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us >> ers >> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.f >> reeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba >> 852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C6364159 >> 40392006303&sdata=wFbEKkGrG49JjArYiD7D1mo6icJFqaxinnS3%2F5xq6hs%3D&re >> served=0 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitchsolutions.com&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636415940392006303&sdata=nGh9In%2FvTyg37f7nJph6TXJFaz2nMMjZ0NXzIZr3hWg%3D&reserved=0 Official FreeSWITCH Sites https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636415940392006303&sdata=wFbEKkGrG49JjArYiD7D1mo6icJFqaxinnS3%2F5xq6hs%3D&reserved=0 https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fconfluence.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636415940392006303&sdata=ehMZuD9vMm325yuExibRBY6G9ysSDmvqoBQaXcHXFww%3D&reserved=0 https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.cluecon.com&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636415940392006303&sdata=J2p7ztQidG%2FLf4c6prqao3hhOnrmHHl2pxlwod3LNKM%3D&reserved=0 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists.freeswitch.org%2Fmailman%2Flistinfo%2Ffreeswitch-users&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636415940392006303&sdata=kpqcuibvTvN3zMcS54w5CemHeLiKgj0OrMMTq1xiMUo%3D&reserved=0 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636415940392006303&sdata=wFbEKkGrG49JjArYiD7D1mo6icJFqaxinnS3%2F5xq6hs%3D&reserved=0 ________________________________ This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. Thank you for your compliance. ________________________________ From infos at madovsky.org Thu Sep 21 14:22:37 2017 From: infos at madovsky.org (Madovsky) Date: Thu, 21 Sep 2017 07:22:37 -0700 Subject: [Freeswitch-users] mod_rtmp channel variables Message-ID: <8644d0bc-69e4-ddbb-4aa2-0c656a7ee0ae@madovsky.org> Hi, is there any example of how to create custom channel variables with mod_rtmp makeCall() function? I made many tries but no custom variables appear in the dialplan. looking at the source code it seems that rtmp_u_ prefix is the trick but not working apparently Am I missing something? Thanks Franck From ssinyagin at gmail.com Thu Sep 21 14:49:32 2017 From: ssinyagin at gmail.com (Stanislav Sinyagin) Date: Thu, 21 Sep 2017 16:49:32 +0200 Subject: [Freeswitch-users] Session recording performance In-Reply-To: References: Message-ID: I actually mentioned it in the thread. Yes, after switching the roles of the boxes, the problem switches too. It's only the call origination box that is producing the jitter on outbound RTP. I also verified it with packet capture on both sides. The RTP leaves the box with jitter already. I'm still analyzing it, trying to find the reason for the jitter. On Thu, Sep 21, 2017 at 4:08 PM, Mundkowsky, Robert wrote: > Just curious, have you tried switching the boxes and seeing if you get same situation? > > Robert > > -----Original Message----- > From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Stanislav Sinyagin > Sent: Thursday, September 21, 2017 8:31 AM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Session recording performance > > but the interesting thing is, that it's the device that originates the SIP call is producing a significantly higher jitter than the one receiving the call. > > The originating command looks like this here, with a SIP address as destination, and a number in local XML dialplan as transfer > application: > https://na01.safelinks.protection.outlook.com/?url=https%3A%2F%2Ffreeswitch.org%2Fconfluence%2Fdisplay%2FFREESWITCH%2FOriginate%2BExample&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636415940392006303&sdata=M5OEPvRrB9JkM5fodeKscJakpsvZn42LhZs2jQle9Ro%3D&reserved=0 > > > > On Thu, Sep 21, 2017 at 7:46 AM, Stanislav Sinyagin wrote: >> with jitterbuffer, it now works as expected. So, the sending party was >> generating extra jitter. >> >> I am playing the same audio file in both directions, and then compare >> the recorded audio with the original by Sevana AQuA software. This >> allows me to detect packet loss in a heterogeneous environment where >> the voice goes through RTP proxies and TDM. >> >> >> >> >> On Thu, Sep 21, 2017 at 3:43 AM, jungle Boogie wrote: >>> What occurs if you send identical audio for the same duration? IOW, >>> play the sound file for each call. >>> >>> >>> _____________________________________________________________________ >>> ____ Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.f >>> reeswitchsolutions.com&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846 >>> ba37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0% >>> 7C636415940392006303&sdata=nGh9In%2FvTyg37f7nJph6TXJFaz2nMMjZ0NXzIZr3 >>> hWg%3D&reserved=0 >>> >>> Official FreeSWITCH Sites >>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.f >>> reeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba >>> 852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C6364159 >>> 40392006303&sdata=wFbEKkGrG49JjArYiD7D1mo6icJFqaxinnS3%2F5xq6hs%3D&re >>> served=0 >>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fconfl >>> uence.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba >>> 37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C >>> 636415940392006303&sdata=ehMZuD9vMm325yuExibRBY6G9ysSDmvqoBQaXcHXFww% >>> 3D&reserved=0 >>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.c >>> luecon.com&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852 >>> 608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C6364159403 >>> 92006303&sdata=J2p7ztQidG%2FLf4c6prqao3hhOnrmHHl2pxlwod3LNKM%3D&reser >>> ved=0 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists >>> .freeswitch.org%2Fmailman%2Flistinfo%2Ffreeswitch-users&data=02%7C01% >>> 7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852608d500ed0775%7C0ba6e9b7 >>> 60b34fae92f37e6ddd9e9b65%7C0%7C0%7C636415940392006303&sdata=kpqcuibvT >>> vN3zMcS54w5CemHeLiKgj0OrMMTq1xiMUo%3D&reserved=0 >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us >>> ers >>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.f >>> reeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba >>> 852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C6364159 >>> 40392006303&sdata=wFbEKkGrG49JjArYiD7D1mo6icJFqaxinnS3%2F5xq6hs%3D&re >>> served=0 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitchsolutions.com&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636415940392006303&sdata=nGh9In%2FvTyg37f7nJph6TXJFaz2nMMjZ0NXzIZr3hWg%3D&reserved=0 > > Official FreeSWITCH Sites > https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636415940392006303&sdata=wFbEKkGrG49JjArYiD7D1mo6icJFqaxinnS3%2F5xq6hs%3D&reserved=0 > https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fconfluence.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636415940392006303&sdata=ehMZuD9vMm325yuExibRBY6G9ysSDmvqoBQaXcHXFww%3D&reserved=0 > https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.cluecon.com&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636415940392006303&sdata=J2p7ztQidG%2FLf4c6prqao3hhOnrmHHl2pxlwod3LNKM%3D&reserved=0 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists.freeswitch.org%2Fmailman%2Flistinfo%2Ffreeswitch-users&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636415940392006303&sdata=kpqcuibvTvN3zMcS54w5CemHeLiKgj0OrMMTq1xiMUo%3D&reserved=0 > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636415940392006303&sdata=wFbEKkGrG49JjArYiD7D1mo6icJFqaxinnS3%2F5xq6hs%3D&reserved=0 > > ________________________________ > > This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. > > > Thank you for your compliance. > > ________________________________ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brandon at cryy.com Thu Sep 21 15:02:20 2017 From: brandon at cryy.com (Brandon Armstead) Date: Thu, 21 Sep 2017 08:02:20 -0700 Subject: [Freeswitch-users] Session recording performance In-Reply-To: References: Message-ID: Any leads maybe under ethtool -S [interface] , possible some kind of checksum / network "interference" , possibly the tx side is doing more work? https://www.kernel.org/doc/Documentation/networking/checksum-offloads.txt On Thu, Sep 21, 2017 at 7:49 AM, Stanislav Sinyagin wrote: > I actually mentioned it in the thread. Yes, after switching the roles > of the boxes, the problem switches too. It's only the call origination > box that is producing the jitter on outbound RTP. I also verified it > with packet capture on both sides. The RTP leaves the box with jitter > already. > > I'm still analyzing it, trying to find the reason for the jitter. > > > > On Thu, Sep 21, 2017 at 4:08 PM, Mundkowsky, Robert wrote: >> Just curious, have you tried switching the boxes and seeing if you get same situation? >> >> Robert >> >> -----Original Message----- >> From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Stanislav Sinyagin >> Sent: Thursday, September 21, 2017 8:31 AM >> To: FreeSWITCH Users Help >> Subject: Re: [Freeswitch-users] Session recording performance >> >> but the interesting thing is, that it's the device that originates the SIP call is producing a significantly higher jitter than the one receiving the call. >> >> The originating command looks like this here, with a SIP address as destination, and a number in local XML dialplan as transfer >> application: >> https://na01.safelinks.protection.outlook.com/?url=https%3A%2F%2Ffreeswitch.org%2Fconfluence%2Fdisplay%2FFREESWITCH%2FOriginate%2BExample&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636415940392006303&sdata=M5OEPvRrB9JkM5fodeKscJakpsvZn42LhZs2jQle9Ro%3D&reserved=0 >> >> >> >> On Thu, Sep 21, 2017 at 7:46 AM, Stanislav Sinyagin wrote: >>> with jitterbuffer, it now works as expected. So, the sending party was >>> generating extra jitter. >>> >>> I am playing the same audio file in both directions, and then compare >>> the recorded audio with the original by Sevana AQuA software. This >>> allows me to detect packet loss in a heterogeneous environment where >>> the voice goes through RTP proxies and TDM. >>> >>> >>> >>> >>> On Thu, Sep 21, 2017 at 3:43 AM, jungle Boogie wrote: >>>> What occurs if you send identical audio for the same duration? IOW, >>>> play the sound file for each call. >>>> >>>> >>>> _____________________________________________________________________ >>>> ____ Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.f >>>> reeswitchsolutions.com&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846 >>>> ba37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0% >>>> 7C636415940392006303&sdata=nGh9In%2FvTyg37f7nJph6TXJFaz2nMMjZ0NXzIZr3 >>>> hWg%3D&reserved=0 >>>> >>>> Official FreeSWITCH Sites >>>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.f >>>> reeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba >>>> 852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C6364159 >>>> 40392006303&sdata=wFbEKkGrG49JjArYiD7D1mo6icJFqaxinnS3%2F5xq6hs%3D&re >>>> served=0 >>>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fconfl >>>> uence.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba >>>> 37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C >>>> 636415940392006303&sdata=ehMZuD9vMm325yuExibRBY6G9ysSDmvqoBQaXcHXFww% >>>> 3D&reserved=0 >>>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.c >>>> luecon.com&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852 >>>> 608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C6364159403 >>>> 92006303&sdata=J2p7ztQidG%2FLf4c6prqao3hhOnrmHHl2pxlwod3LNKM%3D&reser >>>> ved=0 >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists >>>> .freeswitch.org%2Fmailman%2Flistinfo%2Ffreeswitch-users&data=02%7C01% >>>> 7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852608d500ed0775%7C0ba6e9b7 >>>> 60b34fae92f37e6ddd9e9b65%7C0%7C0%7C636415940392006303&sdata=kpqcuibvT >>>> vN3zMcS54w5CemHeLiKgj0OrMMTq1xiMUo%3D&reserved=0 >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us >>>> ers >>>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.f >>>> reeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba >>>> 852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C6364159 >>>> 40392006303&sdata=wFbEKkGrG49JjArYiD7D1mo6icJFqaxinnS3%2F5xq6hs%3D&re >>>> served=0 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitchsolutions.com&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636415940392006303&sdata=nGh9In%2FvTyg37f7nJph6TXJFaz2nMMjZ0NXzIZr3hWg%3D&reserved=0 >> >> Official FreeSWITCH Sites >> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636415940392006303&sdata=wFbEKkGrG49JjArYiD7D1mo6icJFqaxinnS3%2F5xq6hs%3D&reserved=0 >> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fconfluence.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636415940392006303&sdata=ehMZuD9vMm325yuExibRBY6G9ysSDmvqoBQaXcHXFww%3D&reserved=0 >> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.cluecon.com&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636415940392006303&sdata=J2p7ztQidG%2FLf4c6prqao3hhOnrmHHl2pxlwod3LNKM%3D&reserved=0 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists.freeswitch.org%2Fmailman%2Flistinfo%2Ffreeswitch-users&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636415940392006303&sdata=kpqcuibvTvN3zMcS54w5CemHeLiKgj0OrMMTq1xiMUo%3D&reserved=0 >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636415940392006303&sdata=wFbEKkGrG49JjArYiD7D1mo6icJFqaxinnS3%2F5xq6hs%3D&reserved=0 >> >> ________________________________ >> >> This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. >> >> >> Thank you for your compliance. >> >> ________________________________ >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Sincerely, Brandon Armstead CTO / CRYY.com From ssinyagin at gmail.com Thu Sep 21 15:25:38 2017 From: ssinyagin at gmail.com (Stanislav Sinyagin) Date: Thu, 21 Sep 2017 17:25:38 +0200 Subject: [Freeswitch-users] Session recording performance In-Reply-To: References: Message-ID: but the other box is doing the same TX job, and it's doing fine. I'm trying different timers now. On Thu, Sep 21, 2017 at 5:02 PM, Brandon Armstead wrote: > Any leads maybe under ethtool -S [interface] , possible some kind of > checksum / network "interference" , possibly the tx side is doing more > work? > > https://www.kernel.org/doc/Documentation/networking/checksum-offloads.txt > > On Thu, Sep 21, 2017 at 7:49 AM, Stanislav Sinyagin wrote: >> I actually mentioned it in the thread. Yes, after switching the roles >> of the boxes, the problem switches too. It's only the call origination >> box that is producing the jitter on outbound RTP. I also verified it >> with packet capture on both sides. The RTP leaves the box with jitter >> already. >> >> I'm still analyzing it, trying to find the reason for the jitter. >> >> >> >> On Thu, Sep 21, 2017 at 4:08 PM, Mundkowsky, Robert wrote: >>> Just curious, have you tried switching the boxes and seeing if you get same situation? >>> >>> Robert >>> >>> -----Original Message----- >>> From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Stanislav Sinyagin >>> Sent: Thursday, September 21, 2017 8:31 AM >>> To: FreeSWITCH Users Help >>> Subject: Re: [Freeswitch-users] Session recording performance >>> >>> but the interesting thing is, that it's the device that originates the SIP call is producing a significantly higher jitter than the one receiving the call. >>> >>> The originating command looks like this here, with a SIP address as destination, and a number in local XML dialplan as transfer >>> application: >>> https://na01.safelinks.protection.outlook.com/?url=https%3A%2F%2Ffreeswitch.org%2Fconfluence%2Fdisplay%2FFREESWITCH%2FOriginate%2BExample&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636415940392006303&sdata=M5OEPvRrB9JkM5fodeKscJakpsvZn42LhZs2jQle9Ro%3D&reserved=0 >>> >>> >>> >>> On Thu, Sep 21, 2017 at 7:46 AM, Stanislav Sinyagin wrote: >>>> with jitterbuffer, it now works as expected. So, the sending party was >>>> generating extra jitter. >>>> >>>> I am playing the same audio file in both directions, and then compare >>>> the recorded audio with the original by Sevana AQuA software. This >>>> allows me to detect packet loss in a heterogeneous environment where >>>> the voice goes through RTP proxies and TDM. >>>> >>>> >>>> >>>> >>>> On Thu, Sep 21, 2017 at 3:43 AM, jungle Boogie wrote: >>>>> What occurs if you send identical audio for the same duration? IOW, >>>>> play the sound file for each call. >>>>> >>>>> >>>>> _____________________________________________________________________ >>>>> ____ Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.f >>>>> reeswitchsolutions.com&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846 >>>>> ba37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0% >>>>> 7C636415940392006303&sdata=nGh9In%2FvTyg37f7nJph6TXJFaz2nMMjZ0NXzIZr3 >>>>> hWg%3D&reserved=0 >>>>> >>>>> Official FreeSWITCH Sites >>>>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.f >>>>> reeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba >>>>> 852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C6364159 >>>>> 40392006303&sdata=wFbEKkGrG49JjArYiD7D1mo6icJFqaxinnS3%2F5xq6hs%3D&re >>>>> served=0 >>>>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fconfl >>>>> uence.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba >>>>> 37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C >>>>> 636415940392006303&sdata=ehMZuD9vMm325yuExibRBY6G9ysSDmvqoBQaXcHXFww% >>>>> 3D&reserved=0 >>>>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.c >>>>> luecon.com&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852 >>>>> 608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C6364159403 >>>>> 92006303&sdata=J2p7ztQidG%2FLf4c6prqao3hhOnrmHHl2pxlwod3LNKM%3D&reser >>>>> ved=0 >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists >>>>> .freeswitch.org%2Fmailman%2Flistinfo%2Ffreeswitch-users&data=02%7C01% >>>>> 7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852608d500ed0775%7C0ba6e9b7 >>>>> 60b34fae92f37e6ddd9e9b65%7C0%7C0%7C636415940392006303&sdata=kpqcuibvT >>>>> vN3zMcS54w5CemHeLiKgj0OrMMTq1xiMUo%3D&reserved=0 >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us >>>>> ers >>>>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.f >>>>> reeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba >>>>> 852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C6364159 >>>>> 40392006303&sdata=wFbEKkGrG49JjArYiD7D1mo6icJFqaxinnS3%2F5xq6hs%3D&re >>>>> served=0 >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitchsolutions.com&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636415940392006303&sdata=nGh9In%2FvTyg37f7nJph6TXJFaz2nMMjZ0NXzIZr3hWg%3D&reserved=0 >>> >>> Official FreeSWITCH Sites >>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636415940392006303&sdata=wFbEKkGrG49JjArYiD7D1mo6icJFqaxinnS3%2F5xq6hs%3D&reserved=0 >>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fconfluence.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636415940392006303&sdata=ehMZuD9vMm325yuExibRBY6G9ysSDmvqoBQaXcHXFww%3D&reserved=0 >>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.cluecon.com&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636415940392006303&sdata=J2p7ztQidG%2FLf4c6prqao3hhOnrmHHl2pxlwod3LNKM%3D&reserved=0 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists.freeswitch.org%2Fmailman%2Flistinfo%2Ffreeswitch-users&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636415940392006303&sdata=kpqcuibvTvN3zMcS54w5CemHeLiKgj0OrMMTq1xiMUo%3D&reserved=0 >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636415940392006303&sdata=wFbEKkGrG49JjArYiD7D1mo6icJFqaxinnS3%2F5xq6hs%3D&reserved=0 >>> >>> ________________________________ >>> >>> This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. >>> >>> >>> Thank you for your compliance. >>> >>> ________________________________ >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Sincerely, > Brandon Armstead > CTO / CRYY.com > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brandon at cryy.com Thu Sep 21 15:28:32 2017 From: brandon at cryy.com (Brandon Armstead) Date: Thu, 21 Sep 2017 08:28:32 -0700 Subject: [Freeswitch-users] Session recording performance In-Reply-To: References: Message-ID: Your positive GSO / TSO / all are the same? Same Kernel? Same hardware? replica? What happens if you docker your test on 1 same physical host? On Thu, Sep 21, 2017 at 8:25 AM, Stanislav Sinyagin wrote: > but the other box is doing the same TX job, and it's doing fine. > I'm trying different timers now. > > > > On Thu, Sep 21, 2017 at 5:02 PM, Brandon Armstead wrote: >> Any leads maybe under ethtool -S [interface] , possible some kind of >> checksum / network "interference" , possibly the tx side is doing more >> work? >> >> https://www.kernel.org/doc/Documentation/networking/checksum-offloads.txt >> >> On Thu, Sep 21, 2017 at 7:49 AM, Stanislav Sinyagin wrote: >>> I actually mentioned it in the thread. Yes, after switching the roles >>> of the boxes, the problem switches too. It's only the call origination >>> box that is producing the jitter on outbound RTP. I also verified it >>> with packet capture on both sides. The RTP leaves the box with jitter >>> already. >>> >>> I'm still analyzing it, trying to find the reason for the jitter. >>> >>> >>> >>> On Thu, Sep 21, 2017 at 4:08 PM, Mundkowsky, Robert wrote: >>>> Just curious, have you tried switching the boxes and seeing if you get same situation? >>>> >>>> Robert >>>> >>>> -----Original Message----- >>>> From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Stanislav Sinyagin >>>> Sent: Thursday, September 21, 2017 8:31 AM >>>> To: FreeSWITCH Users Help >>>> Subject: Re: [Freeswitch-users] Session recording performance >>>> >>>> but the interesting thing is, that it's the device that originates the SIP call is producing a significantly higher jitter than the one receiving the call. >>>> >>>> The originating command looks like this here, with a SIP address as destination, and a number in local XML dialplan as transfer >>>> application: >>>> https://na01.safelinks.protection.outlook.com/?url=https%3A%2F%2Ffreeswitch.org%2Fconfluence%2Fdisplay%2FFREESWITCH%2FOriginate%2BExample&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636415940392006303&sdata=M5OEPvRrB9JkM5fodeKscJakpsvZn42LhZs2jQle9Ro%3D&reserved=0 >>>> >>>> >>>> >>>> On Thu, Sep 21, 2017 at 7:46 AM, Stanislav Sinyagin wrote: >>>>> with jitterbuffer, it now works as expected. So, the sending party was >>>>> generating extra jitter. >>>>> >>>>> I am playing the same audio file in both directions, and then compare >>>>> the recorded audio with the original by Sevana AQuA software. This >>>>> allows me to detect packet loss in a heterogeneous environment where >>>>> the voice goes through RTP proxies and TDM. >>>>> >>>>> >>>>> >>>>> >>>>> On Thu, Sep 21, 2017 at 3:43 AM, jungle Boogie wrote: >>>>>> What occurs if you send identical audio for the same duration? IOW, >>>>>> play the sound file for each call. >>>>>> >>>>>> >>>>>> _____________________________________________________________________ >>>>>> ____ Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.f >>>>>> reeswitchsolutions.com&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846 >>>>>> ba37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0% >>>>>> 7C636415940392006303&sdata=nGh9In%2FvTyg37f7nJph6TXJFaz2nMMjZ0NXzIZr3 >>>>>> hWg%3D&reserved=0 >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.f >>>>>> reeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba >>>>>> 852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C6364159 >>>>>> 40392006303&sdata=wFbEKkGrG49JjArYiD7D1mo6icJFqaxinnS3%2F5xq6hs%3D&re >>>>>> served=0 >>>>>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fconfl >>>>>> uence.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba >>>>>> 37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C >>>>>> 636415940392006303&sdata=ehMZuD9vMm325yuExibRBY6G9ysSDmvqoBQaXcHXFww% >>>>>> 3D&reserved=0 >>>>>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.c >>>>>> luecon.com&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852 >>>>>> 608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C6364159403 >>>>>> 92006303&sdata=J2p7ztQidG%2FLf4c6prqao3hhOnrmHHl2pxlwod3LNKM%3D&reser >>>>>> ved=0 >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists >>>>>> .freeswitch.org%2Fmailman%2Flistinfo%2Ffreeswitch-users&data=02%7C01% >>>>>> 7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852608d500ed0775%7C0ba6e9b7 >>>>>> 60b34fae92f37e6ddd9e9b65%7C0%7C0%7C636415940392006303&sdata=kpqcuibvT >>>>>> vN3zMcS54w5CemHeLiKgj0OrMMTq1xiMUo%3D&reserved=0 >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us >>>>>> ers >>>>>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.f >>>>>> reeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba >>>>>> 852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C6364159 >>>>>> 40392006303&sdata=wFbEKkGrG49JjArYiD7D1mo6icJFqaxinnS3%2F5xq6hs%3D&re >>>>>> served=0 >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitchsolutions.com&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636415940392006303&sdata=nGh9In%2FvTyg37f7nJph6TXJFaz2nMMjZ0NXzIZr3hWg%3D&reserved=0 >>>> >>>> Official FreeSWITCH Sites >>>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636415940392006303&sdata=wFbEKkGrG49JjArYiD7D1mo6icJFqaxinnS3%2F5xq6hs%3D&reserved=0 >>>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fconfluence.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636415940392006303&sdata=ehMZuD9vMm325yuExibRBY6G9ysSDmvqoBQaXcHXFww%3D&reserved=0 >>>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.cluecon.com&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636415940392006303&sdata=J2p7ztQidG%2FLf4c6prqao3hhOnrmHHl2pxlwod3LNKM%3D&reserved=0 >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists.freeswitch.org%2Fmailman%2Flistinfo%2Ffreeswitch-users&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636415940392006303&sdata=kpqcuibvTvN3zMcS54w5CemHeLiKgj0OrMMTq1xiMUo%3D&reserved=0 >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636415940392006303&sdata=wFbEKkGrG49JjArYiD7D1mo6icJFqaxinnS3%2F5xq6hs%3D&reserved=0 >>>> >>>> ________________________________ >>>> >>>> This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. >>>> >>>> >>>> Thank you for your compliance. >>>> >>>> ________________________________ >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> >> -- >> Sincerely, >> Brandon Armstead >> CTO / CRYY.com >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Sincerely, Brandon Armstead CTO / CRYY.com From ssinyagin at gmail.com Thu Sep 21 16:06:20 2017 From: ssinyagin at gmail.com (Stanislav Sinyagin) Date: Thu, 21 Sep 2017 18:06:20 +0200 Subject: [Freeswitch-users] Session recording performance In-Reply-To: References: Message-ID: it is two identical boxes, set up identically by myself :) I added mod_fdtimer, and set fdtimer as Sofia RTP timer, and the results are impressively better. I'll do some additional tests before a final commit. On Thu, Sep 21, 2017 at 5:28 PM, Brandon Armstead wrote: > Your positive GSO / TSO / all are the same? Same Kernel? Same > hardware? replica? What happens if you docker your test on 1 same > physical host? > > On Thu, Sep 21, 2017 at 8:25 AM, Stanislav Sinyagin wrote: >> but the other box is doing the same TX job, and it's doing fine. >> I'm trying different timers now. >> >> >> >> On Thu, Sep 21, 2017 at 5:02 PM, Brandon Armstead wrote: >>> Any leads maybe under ethtool -S [interface] , possible some kind of >>> checksum / network "interference" , possibly the tx side is doing more >>> work? >>> >>> https://www.kernel.org/doc/Documentation/networking/checksum-offloads.txt >>> >>> On Thu, Sep 21, 2017 at 7:49 AM, Stanislav Sinyagin wrote: >>>> I actually mentioned it in the thread. Yes, after switching the roles >>>> of the boxes, the problem switches too. It's only the call origination >>>> box that is producing the jitter on outbound RTP. I also verified it >>>> with packet capture on both sides. The RTP leaves the box with jitter >>>> already. >>>> >>>> I'm still analyzing it, trying to find the reason for the jitter. >>>> >>>> >>>> >>>> On Thu, Sep 21, 2017 at 4:08 PM, Mundkowsky, Robert wrote: >>>>> Just curious, have you tried switching the boxes and seeing if you get same situation? >>>>> >>>>> Robert >>>>> >>>>> -----Original Message----- >>>>> From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Stanislav Sinyagin >>>>> Sent: Thursday, September 21, 2017 8:31 AM >>>>> To: FreeSWITCH Users Help >>>>> Subject: Re: [Freeswitch-users] Session recording performance >>>>> >>>>> but the interesting thing is, that it's the device that originates the SIP call is producing a significantly higher jitter than the one receiving the call. >>>>> >>>>> The originating command looks like this here, with a SIP address as destination, and a number in local XML dialplan as transfer >>>>> application: >>>>> https://na01.safelinks.protection.outlook.com/?url=https%3A%2F%2Ffreeswitch.org%2Fconfluence%2Fdisplay%2FFREESWITCH%2FOriginate%2BExample&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636415940392006303&sdata=M5OEPvRrB9JkM5fodeKscJakpsvZn42LhZs2jQle9Ro%3D&reserved=0 >>>>> >>>>> >>>>> >>>>> On Thu, Sep 21, 2017 at 7:46 AM, Stanislav Sinyagin wrote: >>>>>> with jitterbuffer, it now works as expected. So, the sending party was >>>>>> generating extra jitter. >>>>>> >>>>>> I am playing the same audio file in both directions, and then compare >>>>>> the recorded audio with the original by Sevana AQuA software. This >>>>>> allows me to detect packet loss in a heterogeneous environment where >>>>>> the voice goes through RTP proxies and TDM. >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> On Thu, Sep 21, 2017 at 3:43 AM, jungle Boogie wrote: >>>>>>> What occurs if you send identical audio for the same duration? IOW, >>>>>>> play the sound file for each call. >>>>>>> >>>>>>> >>>>>>> _____________________________________________________________________ >>>>>>> ____ Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.f >>>>>>> reeswitchsolutions.com&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846 >>>>>>> ba37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0% >>>>>>> 7C636415940392006303&sdata=nGh9In%2FvTyg37f7nJph6TXJFaz2nMMjZ0NXzIZr3 >>>>>>> hWg%3D&reserved=0 >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.f >>>>>>> reeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba >>>>>>> 852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C6364159 >>>>>>> 40392006303&sdata=wFbEKkGrG49JjArYiD7D1mo6icJFqaxinnS3%2F5xq6hs%3D&re >>>>>>> served=0 >>>>>>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fconfl >>>>>>> uence.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba >>>>>>> 37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C >>>>>>> 636415940392006303&sdata=ehMZuD9vMm325yuExibRBY6G9ysSDmvqoBQaXcHXFww% >>>>>>> 3D&reserved=0 >>>>>>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.c >>>>>>> luecon.com&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852 >>>>>>> 608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C6364159403 >>>>>>> 92006303&sdata=J2p7ztQidG%2FLf4c6prqao3hhOnrmHHl2pxlwod3LNKM%3D&reser >>>>>>> ved=0 >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists >>>>>>> .freeswitch.org%2Fmailman%2Flistinfo%2Ffreeswitch-users&data=02%7C01% >>>>>>> 7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852608d500ed0775%7C0ba6e9b7 >>>>>>> 60b34fae92f37e6ddd9e9b65%7C0%7C0%7C636415940392006303&sdata=kpqcuibvT >>>>>>> vN3zMcS54w5CemHeLiKgj0OrMMTq1xiMUo%3D&reserved=0 >>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us >>>>>>> ers >>>>>>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.f >>>>>>> reeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba >>>>>>> 852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C6364159 >>>>>>> 40392006303&sdata=wFbEKkGrG49JjArYiD7D1mo6icJFqaxinnS3%2F5xq6hs%3D&re >>>>>>> served=0 >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitchsolutions.com&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636415940392006303&sdata=nGh9In%2FvTyg37f7nJph6TXJFaz2nMMjZ0NXzIZr3hWg%3D&reserved=0 >>>>> >>>>> Official FreeSWITCH Sites >>>>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636415940392006303&sdata=wFbEKkGrG49JjArYiD7D1mo6icJFqaxinnS3%2F5xq6hs%3D&reserved=0 >>>>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fconfluence.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636415940392006303&sdata=ehMZuD9vMm325yuExibRBY6G9ysSDmvqoBQaXcHXFww%3D&reserved=0 >>>>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.cluecon.com&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636415940392006303&sdata=J2p7ztQidG%2FLf4c6prqao3hhOnrmHHl2pxlwod3LNKM%3D&reserved=0 >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists.freeswitch.org%2Fmailman%2Flistinfo%2Ffreeswitch-users&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636415940392006303&sdata=kpqcuibvTvN3zMcS54w5CemHeLiKgj0OrMMTq1xiMUo%3D&reserved=0 >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636415940392006303&sdata=wFbEKkGrG49JjArYiD7D1mo6icJFqaxinnS3%2F5xq6hs%3D&reserved=0 >>>>> >>>>> ________________________________ >>>>> >>>>> This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. >>>>> >>>>> >>>>> Thank you for your compliance. >>>>> >>>>> ________________________________ >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> >>> -- >>> Sincerely, >>> Brandon Armstead >>> CTO / CRYY.com >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Sincerely, > Brandon Armstead > CTO / CRYY.com > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From luis.daniel.lucio at gmail.com Wed Sep 20 22:34:34 2017 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Wed, 20 Sep 2017 18:34:34 -0400 Subject: [Freeswitch-users] Starting modules when free Message-ID: Just a script I wrote that could be useful to you all http://inside-out.xyz/technology/reloading-a-module-in-a-heavy-used-freeswitch-server.html -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Thu Sep 21 17:51:32 2017 From: mike at jerris.com (Michael Jerris) Date: Thu, 21 Sep 2017 13:51:32 -0400 Subject: [Freeswitch-users] Routing calls to registered user. In-Reply-To: <995A631E-8C13-4E34-9735-806E20C5FA6C@tm.net.uk> References: <995A631E-8C13-4E34-9735-806E20C5FA6C@tm.net.uk> Message-ID: https://freeswitch.org/confluence/display/FREESWITCH/XML+User+Directory#XMLUserDirectory-DialString > On Sep 20, 2017, at 6:36 PM, Joseph Waite wrote: > > Hi Guys > > Knocking my head against this. I am trying to route calls to registered endpoints. But it is not working. > > The log of the whole call is as follows > > 2017-09-20 23:31:49.230782 [NOTICE] switch_channel.c:1104 New Channel sofia/external/07966677711 at 185.8.92.3 [a923b79f-e080-416c-934c-af13b570f813] > 2017-09-20 23:31:49.230782 [DEBUG] switch_core_state_machine.c:584 (sofia/external/07966677711 at 185.8.92.3 ) Running State Change CS_NEW (Cur 1 Tot 16430) > 2017-09-20 23:31:49.230782 [DEBUG] sofia.c:9873 sofia/external/07966677711 at 185.8.92.3 receiving invite from 185.8.92.3:5062 version: 1.6.19 git febfb38 2017-09-14 23:14:02Z 64bit > 2017-09-20 23:31:49.270606 [DEBUG] sofia.c:7084 Channel sofia/external/07966677711 at 185.8.92.3 entering state [received][100] > 2017-09-20 23:31:49.270606 [DEBUG] sofia.c:7094 Remote SDP: > v=0 > o=FreeSWITCH 1505928870 1505928871 IN IP4 185.8.92.3 > s=FreeSWITCH > c=IN IP4 185.8.92.3 > t=0 0 > m=audio 17854 RTP/AVP 8 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > > 2017-09-20 23:31:49.270606 [DEBUG] sofia.c:7486 (sofia/external/07966677711 at 185.8.92.3 ) State Change CS_NEW -> CS_INIT > 2017-09-20 23:31:49.270606 [DEBUG] switch_core_state_machine.c:603 (sofia/external/07966677711 at 185.8.92.3 ) State NEW > 2017-09-20 23:31:49.270606 [DEBUG] switch_core_state_machine.c:584 (sofia/external/07966677711 at 185.8.92.3 ) Running State Change CS_INIT (Cur 1 Tot 16430) > 2017-09-20 23:31:49.270606 [DEBUG] switch_core_state_machine.c:627 (sofia/external/07966677711 at 185.8.92.3 ) State INIT > 2017-09-20 23:31:49.270606 [DEBUG] mod_sofia.c:90 sofia/external/07966677711 at 185.8.92.3 SOFIA INIT > 2017-09-20 23:31:49.270606 [DEBUG] switch_core_state_machine.c:40 sofia/external/07966677711 at 185.8.92.3 Standard INIT > 2017-09-20 23:31:49.270606 [DEBUG] switch_core_state_machine.c:48 (sofia/external/07966677711 at 185.8.92.3 ) State Change CS_INIT -> CS_ROUTING > 2017-09-20 23:31:49.270606 [DEBUG] switch_core_state_machine.c:627 (sofia/external/07966677711 at 185.8.92.3 ) State INIT going to sleep > 2017-09-20 23:31:49.270606 [DEBUG] switch_core_state_machine.c:584 (sofia/external/07966677711 at 185.8.92.3 ) Running State Change CS_ROUTING (Cur 1 Tot 16430) > 2017-09-20 23:31:49.270606 [DEBUG] switch_channel.c:2249 (sofia/external/07966677711 at 185.8.92.3 ) Callstate Change DOWN -> RINGING > 2017-09-20 23:31:49.270606 [DEBUG] switch_core_state_machine.c:643 (sofia/external/07966677711 at 185.8.92.3 ) State ROUTING > 2017-09-20 23:31:49.270606 [DEBUG] mod_sofia.c:143 sofia/external/07966677711 at 185.8.92.3 SOFIA ROUTING > 2017-09-20 23:31:49.270606 [ERR] mod_xml_radius.c:933 Result of true match: 185.35.229.6 == ^185\.35\.229\.30 > 2017-09-20 23:31:49.270606 [ERR] mod_xml_radius.c:930 Didn't match: inbound == ^outbound > 2017-09-20 23:31:49.270606 [DEBUG] switch_core_state_machine.c:236 sofia/external/07966677711 at 185.8.92.3 Standard ROUTING > 2017-09-20 23:31:49.270606 [INFO] mod_dialplan_xml.c:637 Processing 07966677711 <07966677711>->443307881011 in context public > Dialplan: sofia/external/07966677711 at 185.8.92.3 parsing [public->unloop] continue=false > Dialplan: sofia/external/07966677711 at 185.8.92.3 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false > Dialplan: sofia/external/07966677711 at 185.8.92.3 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false > Dialplan: sofia/external/07966677711 at 185.8.92.3 parsing [public->outside_call] continue=true > Dialplan: sofia/external/07966677711 at 185.8.92.3 Absolute Condition [outside_call] > Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(outside_call=true) > Dialplan: sofia/external/07966677711 at 185.8.92.3 Action export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) > Dialplan: sofia/external/07966677711 at 185.8.92.3 parsing [public->call_debug] continue=true > Dialplan: sofia/external/07966677711 at 185.8.92.3 Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never > Dialplan: sofia/external/07966677711 at 185.8.92.3 parsing [public->rejections] continue=false > Dialplan: sofia/external/07966677711 at 185.8.92.3 Regex (FAIL) [rejections] ${radius_auth_result}() =~ /2/ break=on-false > Dialplan: sofia/external/07966677711 at 185.8.92.3 parsing [public->timedouts] continue=false > Dialplan: sofia/external/07966677711 at 185.8.92.3 Regex (FAIL) [timedouts] ${radius_auth_result}() =~ /1/ break=on-false > Dialplan: sofia/external/07966677711 at 185.8.92.3 parsing [public->JeraSoft VCS Routing] continue=false > Dialplan: sofia/external/07966677711 at 185.8.92.3 Regex (PASS) [JeraSoft VCS Routing] destination_number(443307881011) =~ /^(.+)$/ break=on-false > Dialplan: sofia/external/07966677711 at 185.8.92.3 Action export(nolocal:h323-call-origin=originate) > Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(sip_h_X-accountcode=${accountcode}) > Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(call_direction=outbound) > Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(hangup_after_bridge=true) > Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(continue_on_fail=true) > Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(inherit_codec=true) > Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(call_timeout=20) > Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(fail_on_single_reject=USER_BUSY) > Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(origination_caller_id_name=${sip_req_user}) > Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(origination_caller_id_number=${sip_from_user}) > Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(SRCGWIP=src-gw-ip=${network_addr}) INLINE > EXECUTE sofia/external/07966677711 at 185.8.92.3 set(SRCGWIP=src-gw-ip=185.8.92.3) > 2017-09-20 23:31:49.270606 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [SRCGWIP]=[src-gw-ip=185.8.92.3] > Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(CALLID=h323-conf-id=${uuid}) INLINE > EXECUTE sofia/external/07966677711 at 185.8.92.3 set(CALLID=h323-conf-id=a923b79f-e080-416c-934c-af13b570f813) > 2017-09-20 23:31:49.270606 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [CALLID]=[h323-conf-id=a923b79f-e080-416c-934c-af13b570f813] > Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(CALLINGNUMBER=${caller_id_number}) INLINE > EXECUTE sofia/external/07966677711 at 185.8.92.3 set(CALLINGNUMBER=07966677711) > 2017-09-20 23:31:49.270606 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [CALLINGNUMBER]=[07966677711] > Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(CALLEDNUMBER=${destination_number}) INLINE > EXECUTE sofia/external/07966677711 at 185.8.92.3 set(CALLEDNUMBER=443307881011) > 2017-09-20 23:31:49.270606 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [CALLEDNUMBER]=[443307881011] > Dialplan: sofia/external/07966677711 at 185.8.92.3 Action auth_function(in ${DIALED_NUMBER}, in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT) > Dialplan: sofia/external/07966677711 at 185.8.92.3 Action export(credit_time=${credit_time}) > Dialplan: sofia/external/07966677711 at 185.8.92.3 Action export(cisco_service_info=${cisco_service_info) > Dialplan: sofia/external/07966677711 at 185.8.92.3 Action export(sip_redirect_context=redirect) > Dialplan: sofia/external/07966677711 at 185.8.92.3 Action transfer(443307881011 XML ipauth) > Dialplan: sofia/external/07966677711 at 185.8.92.3 Action hangup(${bridge_hangup_cause}) > 2017-09-20 23:31:49.270606 [DEBUG] switch_core_state_machine.c:286 (sofia/external/07966677711 at 185.8.92.3 ) State Change CS_ROUTING -> CS_EXECUTE > 2017-09-20 23:31:49.270606 [DEBUG] switch_core_state_machine.c:643 (sofia/external/07966677711 at 185.8.92.3 ) State ROUTING going to sleep > 2017-09-20 23:31:49.270606 [DEBUG] switch_core_state_machine.c:584 (sofia/external/07966677711 at 185.8.92.3 ) Running State Change CS_EXECUTE (Cur 1 Tot 16430) > 2017-09-20 23:31:49.270606 [DEBUG] switch_core_state_machine.c:650 (sofia/external/07966677711 at 185.8.92.3 ) State EXECUTE > 2017-09-20 23:31:49.270606 [DEBUG] mod_sofia.c:198 sofia/external/07966677711 at 185.8.92.3 SOFIA EXECUTE > 2017-09-20 23:31:49.270606 [DEBUG] switch_core_state_machine.c:328 sofia/external/07966677711 at 185.8.92.3 Standard EXECUTE > EXECUTE sofia/external/07966677711 at 185.8.92.3 set(outside_call=true) > 2017-09-20 23:31:49.270606 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [outside_call]=[true] > EXECUTE sofia/external/07966677711 at 185.8.92.3 export(RFC2822_DATE=Wed, 20 Sep 2017 23:31:49 +0100) > 2017-09-20 23:31:49.270606 [DEBUG] switch_channel.c:1296 EXPORT (export_vars) [RFC2822_DATE]=[Wed, 20 Sep 2017 23:31:49 +0100] > EXECUTE sofia/external/07966677711 at 185.8.92.3 export(nolocal:h323-call-origin=originate) > 2017-09-20 23:31:49.270606 [DEBUG] switch_channel.c:1296 EXPORT (export_vars) (REMOTE ONLY) [h323-call-origin]=[originate] > EXECUTE sofia/external/07966677711 at 185.8.92.3 set(sip_h_X-accountcode=) > 2017-09-20 23:31:49.270606 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [sip_h_X-accountcode]=[UNDEF] > EXECUTE sofia/external/07966677711 at 185.8.92.3 set(call_direction=outbound) > 2017-09-20 23:31:49.270606 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [call_direction]=[outbound] > EXECUTE sofia/external/07966677711 at 185.8.92.3 set(hangup_after_bridge=true) > 2017-09-20 23:31:49.270606 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [hangup_after_bridge]=[true] > EXECUTE sofia/external/07966677711 at 185.8.92.3 set(continue_on_fail=true) > 2017-09-20 23:31:49.270606 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [continue_on_fail]=[true] > EXECUTE sofia/external/07966677711 at 185.8.92.3 set(inherit_codec=true) > 2017-09-20 23:31:49.270606 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [inherit_codec]=[true] > EXECUTE sofia/external/07966677711 at 185.8.92.3 set(call_timeout=20) > 2017-09-20 23:31:49.270606 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [call_timeout]=[20] > EXECUTE sofia/external/07966677711 at 185.8.92.3 set(fail_on_single_reject=USER_BUSY) > 2017-09-20 23:31:49.270606 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [fail_on_single_reject]=[USER_BUSY] > EXECUTE sofia/external/07966677711 at 185.8.92.3 set(origination_caller_id_name=443307881011) > 2017-09-20 23:31:49.270606 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [origination_caller_id_name]=[443307881011] > EXECUTE sofia/external/07966677711 at 185.8.92.3 set(origination_caller_id_number=07966677711) > 2017-09-20 23:31:49.270606 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [origination_caller_id_number]=[07966677711] > EXECUTE sofia/external/07966677711 at 185.8.92.3 auth_function(in , in 07966677711, in , out AUTH_RESULT) > 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:301 allocate initial structure. > 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:313 initialzed configuration. > 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:318 set authserver := 185.35.229.30:1812:h4nn4h. > 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:318 set dictionary := /usr/local/freeswitch/conf/dictionaries/dictionary. > 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:318 set seqfile := /var/run/radius.seq. > 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:318 set mapfile := /usr/local/etc/radiusclient/port-id-map. > 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:318 set default_realm := . > 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:318 set radius_timeout := 3. > 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:318 set radius_retries := 2. > 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:318 set radius_deadtime := 0. > 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:318 set bindaddr := 185.35.229.6. > 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:371 ... radius: User-Name: 07966677711 > 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:380 ... radius: User-Password: > 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:391 ... radius: Called-station-Id is empty, ignoring... > 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:413 Handle attribute: h323-conf-id > 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:439 ... dictionary data: id:24, vendor id:9, attr type:0, attr name:h323-conf-id (589848) > 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:444 ... radius: key:h323-conf-id, value:CALLID (h323-conf-id=a923b79f-e080-416c-934c-af13b570f813) as string > 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:413 Handle attribute: Cisco-AVPair > 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:439 ... dictionary data: id:1, vendor id:9, attr type:0, attr name:Cisco-AVPair (589825) > 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:444 ... radius: key:Cisco-AVPair, value:SRCGWIP (src-gw-ip=185.8.92.3) as string > 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:413 Handle attribute: Cisco-AVPair > 2017-09-20 23:31:49.270606 [ERR] mod_rad_auth.c:178 Undefined channel variable: SRCGWNAME. > 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:439 ... dictionary data: id:1, vendor id:9, attr type:0, attr name:Cisco-AVPair (589825) > 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:444 ... radius: key:Cisco-AVPair, value:SRCGWNAME () as string > 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:413 Handle attribute: Cisco-AVPair > 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:439 ... dictionary data: id:1, vendor id:9, attr type:0, attr name:Cisco-AVPair (589825) > 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:444 ... radius: key:Cisco-AVPair, value:request-type=number (request-type=number) as string > 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:413 Handle attribute: Calling-Station-Id > 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:439 ... dictionary data: id:31, vendor id:0, attr type:0, attr name:Calling-Station-Id (31) > 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:444 ... radius: key:Calling-Station-Id, value:CALLINGNUMBER (07966677711) as string > 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:413 Handle attribute: Called-Station-Id > 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:439 ... dictionary data: id:30, vendor id:0, attr type:0, attr name:Called-Station-Id (30) > 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:444 ... radius: key:Called-Station-Id, value:CALLEDNUMBER (443307881011) as string > 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:491 sending radius packet ... > 2017-09-20 23:31:49.290658 [DEBUG] mod_rad_auth.c:497 RADIUS Authentication OK > 2017-09-20 23:31:49.290658 [DEBUG] mod_rad_auth.c:519 Handle attribute: CREDIT_TIME > 2017-09-20 23:31:49.290658 [DEBUG] mod_rad_auth.c:522 attribute (CREDIT_TIME) found in radius packet > 2017-09-20 23:31:49.290658 [DEBUG] mod_rad_auth.c:523 set variable credit_time := 7199 > 2017-09-20 23:31:49.290658 [DEBUG] mod_rad_auth.c:519 Handle attribute: CISCO_SERVICE_INFO > 2017-09-20 23:31:49.290658 [DEBUG] mod_rad_auth.c:522 attribute (CISCO_SERVICE_INFO) found in radius packet > 2017-09-20 23:31:49.290658 [DEBUG] mod_rad_auth.c:523 set variable cisco_service_info := 200 > 2017-09-20 23:31:49.290658 [DEBUG] mod_rad_auth.c:519 Handle attribute: RADIUS_RETURN_CODE > 2017-09-20 23:31:49.290658 [DEBUG] mod_rad_auth.c:522 attribute (RADIUS_RETURN_CODE) found in radius packet > 2017-09-20 23:31:49.290658 [DEBUG] mod_rad_auth.c:523 set variable return_code := 16 > EXECUTE sofia/external/07966677711 at 185.8.92.3 export(credit_time=7199) > 2017-09-20 23:31:49.290658 [DEBUG] switch_channel.c:1296 EXPORT (export_vars) [credit_time]=[7199] > EXECUTE sofia/external/07966677711 at 185.8.92.3 export(cisco_service_info=200) > 2017-09-20 23:31:49.290658 [DEBUG] switch_channel.c:1296 EXPORT (export_vars) [cisco_service_info]=[200] > EXECUTE sofia/external/07966677711 at 185.8.92.3 export(sip_redirect_context=redirect) > 2017-09-20 23:31:49.290658 [DEBUG] switch_channel.c:1296 EXPORT (export_vars) [sip_redirect_context]=[redirect] > EXECUTE sofia/external/07966677711 at 185.8.92.3 transfer(443307881011 XML ipauth) > 2017-09-20 23:31:49.290658 [DEBUG] switch_ivr.c:2165 (sofia/external/07966677711 at 185.8.92.3 ) State Change CS_EXECUTE -> CS_ROUTING > 2017-09-20 23:31:49.290658 [NOTICE] switch_ivr.c:2172 Transfer sofia/external/07966677711 at 185.8.92.3 to XML[443307881011 at ipauth] > 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:650 (sofia/external/07966677711 at 185.8.92.3 ) State EXECUTE going to sleep > 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:584 (sofia/external/07966677711 at 185.8.92.3 ) Running State Change CS_ROUTING (Cur 1 Tot 16430) > 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:643 (sofia/external/07966677711 at 185.8.92.3 ) State ROUTING > 2017-09-20 23:31:49.290658 [DEBUG] mod_sofia.c:143 sofia/external/07966677711 at 185.8.92.3 SOFIA ROUTING > 2017-09-20 23:31:49.290658 [ERR] mod_xml_radius.c:933 Result of true match: 185.35.229.6 == ^185\.35\.229\.30 > 2017-09-20 23:31:49.290658 [ERR] mod_xml_radius.c:930 Didn't match: inbound == ^outbound > 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:236 sofia/external/07966677711 at 185.8.92.3 Standard ROUTING > 2017-09-20 23:31:49.290658 [INFO] mod_dialplan_xml.c:637 Processing 07966677711 <07966677711>->443307881011 in context ipauth > Dialplan: sofia/external/07966677711 at 185.8.92.3 parsing [ipauth->Auth Calls] continue=false > Dialplan: sofia/external/07966677711 at 185.8.92.3 Regex (PASS) [Auth Calls] ${cisco_service_info}(200) =~ /^200$/ break=never > Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(execute_on_answer=sched_hangup +${credit_time} alloted_timeout) > Dialplan: sofia/external/07966677711 at 185.8.92.3 Action bridge({sip_invite_from_uri=sip:${sip_from_user}@${sip_network_ip}}sofia/external/${destination_number}@185.35.229.30:5060) > Dialplan: sofia/external/07966677711 at 185.8.92.3 Action hangup(${bridge_hangup_cause}) > Dialplan: sofia/external/07966677711 at 185.8.92.3 Regex (FAIL) [Auth Calls] ${cisco_service_info}(200) =~ /^101$/ break=on-true > Dialplan: sofia/external/07966677711 at 185.8.92.3 Regex (FAIL) [Auth Calls] ${cisco_service_info}(200) =~ /^221$/ break=on-true > Dialplan: sofia/external/07966677711 at 185.8.92.3 Regex (FAIL) [Auth Calls] ${cisco_service_info}(200) =~ /^201$/ break=on-true > Dialplan: sofia/external/07966677711 at 185.8.92.3 Absolute Condition [Auth Calls] > Dialplan: sofia/external/07966677711 at 185.8.92.3 Action log(INFO Were in the IPauth Context!!) > Dialplan: sofia/external/07966677711 at 185.8.92.3 Action log(INFO credit_time=${credit_time}) > Dialplan: sofia/external/07966677711 at 185.8.92.3 Action log(INFO cisco_service_info=${cisco_service_info}) > 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:286 (sofia/external/07966677711 at 185.8.92.3 ) State Change CS_ROUTING -> CS_EXECUTE > 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:643 (sofia/external/07966677711 at 185.8.92.3 ) State ROUTING going to sleep > 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:584 (sofia/external/07966677711 at 185.8.92.3 ) Running State Change CS_EXECUTE (Cur 1 Tot 16430) > 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:650 (sofia/external/07966677711 at 185.8.92.3 ) State EXECUTE > 2017-09-20 23:31:49.290658 [DEBUG] mod_sofia.c:198 sofia/external/07966677711 at 185.8.92.3 SOFIA EXECUTE > 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:328 sofia/external/07966677711 at 185.8.92.3 Standard EXECUTE > EXECUTE sofia/external/07966677711 at 185.8.92.3 set(execute_on_answer=sched_hangup +7199 alloted_timeout) > 2017-09-20 23:31:49.290658 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [execute_on_answer]=[sched_hangup +7199 alloted_timeout] > EXECUTE sofia/external/07966677711 at 185.8.92.3 bridge({sip_invite_from_uri=sip:07966677711 at 185.8.92.3 }sofia/external/443307881011 at 185.35.229.30 :5060) > 2017-09-20 23:31:49.290658 [DEBUG] switch_channel.c:1250 sofia/external/07966677711 at 185.8.92.3 EXPORTING[export_vars] [RFC2822_DATE]=[Wed, 20 Sep 2017 23:31:49 +0100] to event > 2017-09-20 23:31:49.290658 [DEBUG] switch_channel.c:1250 sofia/external/07966677711 at 185.8.92.3 EXPORTING[export_vars] [h323-call-origin]=[originate] to event > 2017-09-20 23:31:49.290658 [DEBUG] switch_channel.c:1250 sofia/external/07966677711 at 185.8.92.3 EXPORTING[export_vars] [credit_time]=[7199] to event > 2017-09-20 23:31:49.290658 [DEBUG] switch_channel.c:1250 sofia/external/07966677711 at 185.8.92.3 EXPORTING[export_vars] [cisco_service_info]=[200] to event > 2017-09-20 23:31:49.290658 [DEBUG] switch_channel.c:1250 sofia/external/07966677711 at 185.8.92.3 EXPORTING[export_vars] [sip_redirect_context]=[redirect] to event > 2017-09-20 23:31:49.290658 [DEBUG] switch_ivr_originate.c:2142 Parsing global variables > 2017-09-20 23:31:49.290658 [NOTICE] switch_channel.c:1104 New Channel sofia/external/443307881011 at 185.35.229.30 :5060 [848843db-cad9-48a8-a4bd-2ac6b6b77dff] > 2017-09-20 23:31:49.290658 [DEBUG] mod_sofia.c:4819 (sofia/external/443307881011 at 185.35.229.30 :5060) State Change CS_NEW -> CS_INIT > 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:584 (sofia/external/443307881011 at 185.35.229.30 :5060) Running State Change CS_INIT (Cur 2 Tot 16431) > 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:627 (sofia/external/443307881011 at 185.35.229.30 :5060) State INIT > 2017-09-20 23:31:49.290658 [DEBUG] mod_sofia.c:90 sofia/external/443307881011 at 185.35.229.30 :5060 SOFIA INIT > 2017-09-20 23:31:49.290658 [DEBUG] sofia_glue.c:1295 sofia/external/443307881011 at 185.35.229.30 :5060 sending invite version: 1.6.19 git febfb38 2017-09-14 23:14:02Z 64bit > Local SDP: > v=0 > o=FreeSWITCH 1505929783 1505929784 IN IP4 185.35.229.6 > s=FreeSWITCH > c=IN IP4 185.35.229.6 > t=0 0 > m=audio 16926 RTP/AVP 8 101 13 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > a=sendrecv > > 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:40 sofia/external/443307881011 at 185.35.229.30 :5060 Standard INIT > 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:48 (sofia/external/443307881011 at 185.35.229.30 :5060) State Change CS_INIT -> CS_ROUTING > 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:627 (sofia/external/443307881011 at 185.35.229.30 :5060) State INIT going to sleep > 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:584 (sofia/external/443307881011 at 185.35.229.30 :5060) Running State Change CS_ROUTING (Cur 2 Tot 16431) > 2017-09-20 23:31:49.290658 [DEBUG] sofia.c:7084 Channel sofia/external/443307881011 at 185.35.229.30 :5060 entering state [calling][0] > 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:643 (sofia/external/443307881011 at 185.35.229.30 :5060) State ROUTING > 2017-09-20 23:31:49.290658 [DEBUG] mod_sofia.c:143 sofia/external/443307881011 at 185.35.229.30 :5060 SOFIA ROUTING > 2017-09-20 23:31:49.290658 [DEBUG] switch_ivr_originate.c:67 (sofia/external/443307881011 at 185.35.229.30 :5060) State Change CS_ROUTING -> CS_CONSUME_MEDIA > 2017-09-20 23:31:49.290658 [ERR] mod_xml_radius.c:930 Didn't match: 185.35.229.30:5060 == ^185\.35\.229\.30 > 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:643 (sofia/external/443307881011 at 185.35.229.30 :5060) State ROUTING going to sleep > 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:584 (sofia/external/443307881011 at 185.35.229.30 :5060) Running State Change CS_CONSUME_MEDIA (Cur 2 Tot 16431) > 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:662 (sofia/external/443307881011 at 185.35.229.30 :5060) State CONSUME_MEDIA > 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:662 (sofia/external/443307881011 at 185.35.229.30 :5060) State CONSUME_MEDIA going to sleep > 2017-09-20 23:31:49.350696 [DEBUG] sofia.c:6540 Redirect: Transfering to joehouse XML redirect > 2017-09-20 23:31:49.350696 [DEBUG] switch_ivr.c:2165 (sofia/external/07966677711 at 185.8.92.3 ) State Change CS_EXECUTE -> CS_ROUTING > 2017-09-20 23:31:49.350696 [NOTICE] switch_ivr.c:2172 Transfer sofia/external/07966677711 at 185.8.92.3 to XML[joehouse at redirect] > 2017-09-20 23:31:49.350696 [NOTICE] sofia.c:6549 Hangup sofia/external/443307881011 at 185.35.229.30 :5060 [CS_CONSUME_MEDIA] [REDIRECTION_TO_NEW_DESTINATION] > 2017-09-20 23:31:49.350696 [DEBUG] sofia.c:1453 Channel is already hungup. > 2017-09-20 23:31:49.350696 [DEBUG] sofia.c:1453 Channel is already hungup. > 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:584 (sofia/external/443307881011 at 185.35.229.30 :5060) Running State Change CS_HANGUP (Cur 2 Tot 16431) > 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:850 (sofia/external/443307881011 at 185.35.229.30 :5060) Callstate Change DOWN -> HANGUP > 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:852 (sofia/external/443307881011 at 185.35.229.30 :5060) State HANGUP > 2017-09-20 23:31:49.350696 [DEBUG] mod_sofia.c:438 Channel sofia/external/443307881011 at 185.35.229.30 :5060 hanging up, cause: REDIRECTION_TO_NEW_DESTINATION > 2017-09-20 23:31:49.350696 [DEBUG] mod_sofia.c:502 Sending CANCEL to sofia/external/443307881011 at 185.35.229.30 :5060 > 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:60 sofia/external/443307881011 at 185.35.229.30 :5060 Standard HANGUP, cause: REDIRECTION_TO_NEW_DESTINATION > 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:852 (sofia/external/443307881011 at 185.35.229.30 :5060) State HANGUP going to sleep > 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:619 (sofia/external/443307881011 at 185.35.229.30 :5060) State Change CS_HANGUP -> CS_REPORTING > 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:584 (sofia/external/443307881011 at 185.35.229.30 :5060) Running State Change CS_REPORTING (Cur 2 Tot 16431) > 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:938 (sofia/external/443307881011 at 185.35.229.30 :5060) State REPORTING > 2017-09-20 23:31:49.350696 [ERR] mod_xml_radius.c:930 Didn't match: 185.35.229.30:5060 == ^185\.35\.229\.30 > 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:174 sofia/external/443307881011 at 185.35.229.30 :5060 Standard REPORTING, cause: REDIRECTION_TO_NEW_DESTINATION > 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:938 (sofia/external/443307881011 at 185.35.229.30 :5060) State REPORTING going to sleep > 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:610 (sofia/external/443307881011 at 185.35.229.30 :5060) State Change CS_REPORTING -> CS_DESTROY > 2017-09-20 23:31:49.350696 [DEBUG] switch_core_session.c:1665 Session 16431 (sofia/external/443307881011 at 185.35.229.30 :5060) Locked, Waiting on external entities > 2017-09-20 23:31:49.350696 [DEBUG] switch_ivr_originate.c:3837 Originate Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL] > 2017-09-20 23:31:49.350696 [NOTICE] switch_core_session.c:1683 Session 16431 (sofia/external/443307881011 at 185.35.229.30 :5060) Ended > 2017-09-20 23:31:49.350696 [NOTICE] switch_core_session.c:1687 Close Channel sofia/external/443307881011 at 185.35.229.30 :5060 [CS_DESTROY] > 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:741 (sofia/external/443307881011 at 185.35.229.30 :5060) Running State Change CS_DESTROY (Cur 1 Tot 16431) > 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:751 (sofia/external/443307881011 at 185.35.229.30 :5060) State DESTROY > 2017-09-20 23:31:49.350696 [DEBUG] mod_sofia.c:343 sofia/external/443307881011 at 185.35.229.30 :5060 SOFIA DESTROY > 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:181 sofia/external/443307881011 at 185.35.229.30 :5060 Standard DESTROY > 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:751 (sofia/external/443307881011 at 185.35.229.30 :5060) State DESTROY going to sleep > 2017-09-20 23:31:49.350696 [INFO] mod_dptools.c:3436 Originate Failed. Cause: ORIGINATOR_CANCEL > 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:650 (sofia/external/07966677711 at 185.8.92.3 ) State EXECUTE going to sleep > 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:584 (sofia/external/07966677711 at 185.8.92.3 ) Running State Change CS_ROUTING (Cur 1 Tot 16431) > 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:643 (sofia/external/07966677711 at 185.8.92.3 ) State ROUTING > 2017-09-20 23:31:49.350696 [DEBUG] mod_sofia.c:143 sofia/external/07966677711 at 185.8.92.3 SOFIA ROUTING > 2017-09-20 23:31:49.350696 [ERR] mod_xml_radius.c:933 Result of true match: 185.35.229.6 == ^185\.35\.229\.30 > 2017-09-20 23:31:49.350696 [ERR] mod_xml_radius.c:930 Didn't match: inbound == ^outbound > 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:236 sofia/external/07966677711 at 185.8.92.3 Standard ROUTING > 2017-09-20 23:31:49.350696 [INFO] mod_dialplan_xml.c:637 Processing 07966677711 <07966677711>->joehouse in context redirect > Dialplan: sofia/external/07966677711 at 185.8.92.3 parsing [redirect->Redirect Calls] continue=false > Dialplan: sofia/external/07966677711 at 185.8.92.3 Regex (PASS) [Redirect Calls] ${sip_redirect_contact_0}(>;src_number=07966677711;q=1.00) =~ /^ Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(process_cdr=true) > Dialplan: sofia/external/07966677711 at 185.8.92.3 Action log(ALERT Local User) > Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(accountcode=jerasoftoutbound) > Dialplan: sofia/external/07966677711 at 185.8.92.3 Action log(ALERT joehouse) > Dialplan: sofia/external/07966677711 at 185.8.92.3 Action bridge(user/joehouse at sip.biznetuk.com ) > 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:286 (sofia/external/07966677711 at 185.8.92.3 ) State Change CS_ROUTING -> CS_EXECUTE > 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:643 (sofia/external/07966677711 at 185.8.92.3 ) State ROUTING going to sleep > 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:584 (sofia/external/07966677711 at 185.8.92.3 ) Running State Change CS_EXECUTE (Cur 1 Tot 16431) > 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:650 (sofia/external/07966677711 at 185.8.92.3 ) State EXECUTE > 2017-09-20 23:31:49.350696 [DEBUG] mod_sofia.c:198 sofia/external/07966677711 at 185.8.92.3 SOFIA EXECUTE > 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:328 sofia/external/07966677711 at 185.8.92.3 Standard EXECUTE > EXECUTE sofia/external/07966677711 at 185.8.92.3 set(process_cdr=true) > 2017-09-20 23:31:49.350696 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [process_cdr]=[true] > EXECUTE sofia/external/07966677711 at 185.8.92.3 log(ALERT Local User) > 2017-09-20 23:31:49.350696 [ALERT] mod_dptools.c:1742 Local User > EXECUTE sofia/external/07966677711 at 185.8.92.3 set(accountcode=jerasoftoutbound) > 2017-09-20 23:31:49.350696 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [accountcode]=[jerasoftoutbound] > EXECUTE sofia/external/07966677711 at 185.8.92.3 log(ALERT joehouse) > 2017-09-20 23:31:49.350696 [ALERT] mod_dptools.c:1742 joehouse > EXECUTE sofia/external/07966677711 at 185.8.92.3 bridge(user/joehouse at sip.biznetuk.com ) > 2017-09-20 23:31:49.350696 [DEBUG] switch_channel.c:1250 sofia/external/07966677711 at 185.8.92.3 EXPORTING[export_vars] [RFC2822_DATE]=[Wed, 20 Sep 2017 23:31:49 +0100] to event > 2017-09-20 23:31:49.350696 [DEBUG] switch_channel.c:1250 sofia/external/07966677711 at 185.8.92.3 EXPORTING[export_vars] [h323-call-origin]=[originate] to event > 2017-09-20 23:31:49.350696 [DEBUG] switch_channel.c:1250 sofia/external/07966677711 at 185.8.92.3 EXPORTING[export_vars] [credit_time]=[7199] to event > 2017-09-20 23:31:49.350696 [DEBUG] switch_channel.c:1250 sofia/external/07966677711 at 185.8.92.3 EXPORTING[export_vars] [cisco_service_info]=[200] to event > 2017-09-20 23:31:49.350696 [DEBUG] switch_channel.c:1250 sofia/external/07966677711 at 185.8.92.3 EXPORTING[export_vars] [sip_redirect_context]=[redirect] to event > 2017-09-20 23:31:49.350696 [DEBUG] switch_ivr_originate.c:2142 Parsing global variables > 2017-09-20 23:31:49.350696 [WARNING] mod_dptools.c:4184 Can't find user [joehouse at sip.biznetuk.com ] > 2017-09-20 23:31:49.350696 [NOTICE] switch_ivr_originate.c:2851 Cannot create outgoing channel of type [user] cause: [SUBSCRIBER_ABSENT] > 2017-09-20 23:31:49.350696 [DEBUG] switch_ivr_originate.c:3848 Originate Resulted in Error Cause: 20 [SUBSCRIBER_ABSENT] > 2017-09-20 23:31:49.350696 [INFO] mod_dptools.c:3436 Originate Failed. Cause: SUBSCRIBER_ABSENT > 2017-09-20 23:31:49.350696 [NOTICE] switch_core_state_machine.c:385 sofia/external/07966677711 at 185.8.92.3 has executed the last dialplan instruction, hanging up. > 2017-09-20 23:31:49.350696 [NOTICE] switch_core_state_machine.c:387 Hangup sofia/external/07966677711 at 185.8.92.3 [CS_EXECUTE] [NORMAL_CLEARING] > 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:650 (sofia/external/07966677711 at 185.8.92.3 ) State EXECUTE going to sleep > 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:584 (sofia/external/07966677711 at 185.8.92.3 ) Running State Change CS_HANGUP (Cur 1 Tot 16431) > 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:850 (sofia/external/07966677711 at 185.8.92.3 ) Callstate Change RINGING -> HANGUP > 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:852 (sofia/external/07966677711 at 185.8.92.3 ) State HANGUP > 2017-09-20 23:31:49.350696 [DEBUG] mod_sofia.c:438 Channel sofia/external/07966677711 at 185.8.92.3 hanging up, cause: NORMAL_CLEARING > 2017-09-20 23:31:49.350696 [DEBUG] mod_sofia.c:577 Responding to INVITE with: 480 > 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:60 sofia/external/07966677711 at 185.8.92.3 Standard HANGUP, cause: NORMAL_CLEARING > 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:852 (sofia/external/07966677711 at 185.8.92.3 ) State HANGUP going to sleep > 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:619 (sofia/external/07966677711 at 185.8.92.3 ) State Change CS_HANGUP -> CS_REPORTING > 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:584 (sofia/external/07966677711 at 185.8.92.3 ) Running State Change CS_REPORTING (Cur 1 Tot 16431) > 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:938 (sofia/external/07966677711 at 185.8.92.3 ) State REPORTING > 2017-09-20 23:31:49.350696 [ERR] mod_xml_radius.c:933 Result of true match: 185.35.229.6 == ^185\.35\.229\.30 > 2017-09-20 23:31:49.350696 [ERR] mod_xml_radius.c:930 Didn't match: inbound == ^outbound > 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:174 sofia/external/07966677711 at 185.8.92.3 Standard REPORTING, cause: NORMAL_CLEARING > 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:938 (sofia/external/07966677711 at 185.8.92.3 ) State REPORTING going to sleep > 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:610 (sofia/external/07966677711 at 185.8.92.3 ) State Change CS_REPORTING -> CS_DESTROY > 2017-09-20 23:31:49.350696 [DEBUG] switch_core_session.c:1665 Session 16430 (sofia/external/07966677711 at 185.8.92.3 ) Locked, Waiting on external entities > 2017-09-20 23:31:49.350696 [NOTICE] switch_core_session.c:1683 Session 16430 (sofia/external/07966677711 at 185.8.92.3 ) Ended > 2017-09-20 23:31:49.350696 [NOTICE] switch_core_session.c:1687 Close Channel sofia/external/07966677711 at 185.8.92.3 [CS_DESTROY] > 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:741 (sofia/external/07966677711 at 185.8.92.3 ) Running State Change CS_DESTROY (Cur 0 Tot 16431) > 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:751 (sofia/external/07966677711 at 185.8.92.3 ) State DESTROY > 2017-09-20 23:31:49.350696 [DEBUG] mod_sofia.c:343 sofia/external/07966677711 at 185.8.92.3 SOFIA DESTROY > 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:181 sofia/external/07966677711 at 185.8.92.3 Standard DESTROY > 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:751 (sofia/external/07966677711 at 185.8.92.3 ) State DESTROY going to sleep > > its saying there is no endpoint registered yet the output of show registrations shows there is > > freeswitch at BiznetukSBC1> show registrations > reg_user,realm,token,url,expires,network_ip,network_port,network_proto,hostname,metadata,registration_uuid > joehouse,sip.biznetuk.com ,1030250472 at 192_168_1_101,sofia/internal/sip:joehouse at 185.35.230.2:5382,1505947110,185.35.230.2,5382,udp,BiznetukSBC1,,60484000-aa2d-4b21-9d58-11e7d4155f1e > > 1 total. > > > Any ideas? > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Thu Sep 21 17:54:59 2017 From: mike at jerris.com (Michael Jerris) Date: Thu, 21 Sep 2017 13:54:59 -0400 Subject: [Freeswitch-users] ICE/DTLS handshake In-Reply-To: References: Message-ID: its not going to negotiate until we get the stun responses. If we are not, you should look if the client is sending them and something is blocking, or why the client is waiting to send them. Sounds broken on client side from the description. > On Sep 21, 2017, at 7:53 AM, Mirko Brankovic wrote: > > HI, > Has anyone experienced DTLS handshake takes 5s to get to SETUP state: > 2017-09-21 11:40:40.021178 [INFO] switch_rtp.c:3515 Changing audio DTLS state from OFF to HANDSHAKE > 2017-09-21 11:40:45.319457 [INFO] switch_rtp.c:3172 Changing audio DTLS state from HANDSHAKE to SETUP > > In network dump i see that answering side is not sending STUN for this 5s and then suddenly answers last 5 STUNs from A side. > > Has anyone encountered this kind of problem ? > > I have a pcap if necessary... -------------- next part -------------- An HTML attachment was scrubbed... URL: From joelists at tm.net.uk Thu Sep 21 19:18:18 2017 From: joelists at tm.net.uk (Joseph Waite) Date: Thu, 21 Sep 2017 20:18:18 +0100 Subject: [Freeswitch-users] Routing calls to registered user. In-Reply-To: References: <995A631E-8C13-4E34-9735-806E20C5FA6C@tm.net.uk> Message-ID: <3C175D07-A686-4B45-A099-D6FCB111F395@tm.net.uk> Thanks for that Michael The specific point referred to wouldn’t help in my scenario as I have no xml directory to refer to. However reading that doc further I realised that in my use with users authenticated via Radius and not xml directory I needed to use So I now have calls routing to registered endpoints! > On 21 Sep 2017, at 18:51, Michael Jerris wrote: > > https://freeswitch.org/confluence/display/FREESWITCH/XML+User+Directory#XMLUserDirectory-DialString > >> On Sep 20, 2017, at 6:36 PM, Joseph Waite > wrote: >> >> Hi Guys >> >> Knocking my head against this. I am trying to route calls to registered endpoints. But it is not working. >> >> The log of the whole call is as follows >> >> 2017-09-20 23:31:49.230782 [NOTICE] switch_channel.c:1104 New Channel sofia/external/07966677711 at 185.8.92.3 [a923b79f-e080-416c-934c-af13b570f813] >> 2017-09-20 23:31:49.230782 [DEBUG] switch_core_state_machine.c:584 (sofia/external/07966677711 at 185.8.92.3 ) Running State Change CS_NEW (Cur 1 Tot 16430) >> 2017-09-20 23:31:49.230782 [DEBUG] sofia.c:9873 sofia/external/07966677711 at 185.8.92.3 receiving invite from 185.8.92.3:5062 version: 1.6.19 git febfb38 2017-09-14 23:14:02Z 64bit >> 2017-09-20 23:31:49.270606 [DEBUG] sofia.c:7084 Channel sofia/external/07966677711 at 185.8.92.3 entering state [received][100] >> 2017-09-20 23:31:49.270606 [DEBUG] sofia.c:7094 Remote SDP: >> v=0 >> o=FreeSWITCH 1505928870 1505928871 IN IP4 185.8.92.3 >> s=FreeSWITCH >> c=IN IP4 185.8.92.3 >> t=0 0 >> m=audio 17854 RTP/AVP 8 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=silenceSupp:off - - - - >> a=ptime:20 >> >> 2017-09-20 23:31:49.270606 [DEBUG] sofia.c:7486 (sofia/external/07966677711 at 185.8.92.3 ) State Change CS_NEW -> CS_INIT >> 2017-09-20 23:31:49.270606 [DEBUG] switch_core_state_machine.c:603 (sofia/external/07966677711 at 185.8.92.3 ) State NEW >> 2017-09-20 23:31:49.270606 [DEBUG] switch_core_state_machine.c:584 (sofia/external/07966677711 at 185.8.92.3 ) Running State Change CS_INIT (Cur 1 Tot 16430) >> 2017-09-20 23:31:49.270606 [DEBUG] switch_core_state_machine.c:627 (sofia/external/07966677711 at 185.8.92.3 ) State INIT >> 2017-09-20 23:31:49.270606 [DEBUG] mod_sofia.c:90 sofia/external/07966677711 at 185.8.92.3 SOFIA INIT >> 2017-09-20 23:31:49.270606 [DEBUG] switch_core_state_machine.c:40 sofia/external/07966677711 at 185.8.92.3 Standard INIT >> 2017-09-20 23:31:49.270606 [DEBUG] switch_core_state_machine.c:48 (sofia/external/07966677711 at 185.8.92.3 ) State Change CS_INIT -> CS_ROUTING >> 2017-09-20 23:31:49.270606 [DEBUG] switch_core_state_machine.c:627 (sofia/external/07966677711 at 185.8.92.3 ) State INIT going to sleep >> 2017-09-20 23:31:49.270606 [DEBUG] switch_core_state_machine.c:584 (sofia/external/07966677711 at 185.8.92.3 ) Running State Change CS_ROUTING (Cur 1 Tot 16430) >> 2017-09-20 23:31:49.270606 [DEBUG] switch_channel.c:2249 (sofia/external/07966677711 at 185.8.92.3 ) Callstate Change DOWN -> RINGING >> 2017-09-20 23:31:49.270606 [DEBUG] switch_core_state_machine.c:643 (sofia/external/07966677711 at 185.8.92.3 ) State ROUTING >> 2017-09-20 23:31:49.270606 [DEBUG] mod_sofia.c:143 sofia/external/07966677711 at 185.8.92.3 SOFIA ROUTING >> 2017-09-20 23:31:49.270606 [ERR] mod_xml_radius.c:933 Result of true match: 185.35.229.6 == ^185\.35\.229\.30 >> 2017-09-20 23:31:49.270606 [ERR] mod_xml_radius.c:930 Didn't match: inbound == ^outbound >> 2017-09-20 23:31:49.270606 [DEBUG] switch_core_state_machine.c:236 sofia/external/07966677711 at 185.8.92.3 Standard ROUTING >> 2017-09-20 23:31:49.270606 [INFO] mod_dialplan_xml.c:637 Processing 07966677711 <07966677711>->443307881011 in context public >> Dialplan: sofia/external/07966677711 at 185.8.92.3 parsing [public->unloop] continue=false >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false >> Dialplan: sofia/external/07966677711 at 185.8.92.3 parsing [public->outside_call] continue=true >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Absolute Condition [outside_call] >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(outside_call=true) >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) >> Dialplan: sofia/external/07966677711 at 185.8.92.3 parsing [public->call_debug] continue=true >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never >> Dialplan: sofia/external/07966677711 at 185.8.92.3 parsing [public->rejections] continue=false >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Regex (FAIL) [rejections] ${radius_auth_result}() =~ /2/ break=on-false >> Dialplan: sofia/external/07966677711 at 185.8.92.3 parsing [public->timedouts] continue=false >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Regex (FAIL) [timedouts] ${radius_auth_result}() =~ /1/ break=on-false >> Dialplan: sofia/external/07966677711 at 185.8.92.3 parsing [public->JeraSoft VCS Routing] continue=false >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Regex (PASS) [JeraSoft VCS Routing] destination_number(443307881011) =~ /^(.+)$/ break=on-false >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action export(nolocal:h323-call-origin=originate) >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(sip_h_X-accountcode=${accountcode}) >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(call_direction=outbound) >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(hangup_after_bridge=true) >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(continue_on_fail=true) >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(inherit_codec=true) >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(call_timeout=20) >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(fail_on_single_reject=USER_BUSY) >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(origination_caller_id_name=${sip_req_user}) >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(origination_caller_id_number=${sip_from_user}) >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(SRCGWIP=src-gw-ip=${network_addr}) INLINE >> EXECUTE sofia/external/07966677711 at 185.8.92.3 set(SRCGWIP=src-gw-ip=185.8.92.3) >> 2017-09-20 23:31:49.270606 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [SRCGWIP]=[src-gw-ip=185.8.92.3] >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(CALLID=h323-conf-id=${uuid}) INLINE >> EXECUTE sofia/external/07966677711 at 185.8.92.3 set(CALLID=h323-conf-id=a923b79f-e080-416c-934c-af13b570f813) >> 2017-09-20 23:31:49.270606 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [CALLID]=[h323-conf-id=a923b79f-e080-416c-934c-af13b570f813] >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(CALLINGNUMBER=${caller_id_number}) INLINE >> EXECUTE sofia/external/07966677711 at 185.8.92.3 set(CALLINGNUMBER=07966677711) >> 2017-09-20 23:31:49.270606 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [CALLINGNUMBER]=[07966677711] >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(CALLEDNUMBER=${destination_number}) INLINE >> EXECUTE sofia/external/07966677711 at 185.8.92.3 set(CALLEDNUMBER=443307881011) >> 2017-09-20 23:31:49.270606 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [CALLEDNUMBER]=[443307881011] >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action auth_function(in ${DIALED_NUMBER}, in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT) >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action export(credit_time=${credit_time}) >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action export(cisco_service_info=${cisco_service_info) >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action export(sip_redirect_context=redirect) >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action transfer(443307881011 XML ipauth) >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action hangup(${bridge_hangup_cause}) >> 2017-09-20 23:31:49.270606 [DEBUG] switch_core_state_machine.c:286 (sofia/external/07966677711 at 185.8.92.3 ) State Change CS_ROUTING -> CS_EXECUTE >> 2017-09-20 23:31:49.270606 [DEBUG] switch_core_state_machine.c:643 (sofia/external/07966677711 at 185.8.92.3 ) State ROUTING going to sleep >> 2017-09-20 23:31:49.270606 [DEBUG] switch_core_state_machine.c:584 (sofia/external/07966677711 at 185.8.92.3 ) Running State Change CS_EXECUTE (Cur 1 Tot 16430) >> 2017-09-20 23:31:49.270606 [DEBUG] switch_core_state_machine.c:650 (sofia/external/07966677711 at 185.8.92.3 ) State EXECUTE >> 2017-09-20 23:31:49.270606 [DEBUG] mod_sofia.c:198 sofia/external/07966677711 at 185.8.92.3 SOFIA EXECUTE >> 2017-09-20 23:31:49.270606 [DEBUG] switch_core_state_machine.c:328 sofia/external/07966677711 at 185.8.92.3 Standard EXECUTE >> EXECUTE sofia/external/07966677711 at 185.8.92.3 set(outside_call=true) >> 2017-09-20 23:31:49.270606 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [outside_call]=[true] >> EXECUTE sofia/external/07966677711 at 185.8.92.3 export(RFC2822_DATE=Wed, 20 Sep 2017 23:31:49 +0100) >> 2017-09-20 23:31:49.270606 [DEBUG] switch_channel.c:1296 EXPORT (export_vars) [RFC2822_DATE]=[Wed, 20 Sep 2017 23:31:49 +0100] >> EXECUTE sofia/external/07966677711 at 185.8.92.3 export(nolocal:h323-call-origin=originate) >> 2017-09-20 23:31:49.270606 [DEBUG] switch_channel.c:1296 EXPORT (export_vars) (REMOTE ONLY) [h323-call-origin]=[originate] >> EXECUTE sofia/external/07966677711 at 185.8.92.3 set(sip_h_X-accountcode=) >> 2017-09-20 23:31:49.270606 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [sip_h_X-accountcode]=[UNDEF] >> EXECUTE sofia/external/07966677711 at 185.8.92.3 set(call_direction=outbound) >> 2017-09-20 23:31:49.270606 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [call_direction]=[outbound] >> EXECUTE sofia/external/07966677711 at 185.8.92.3 set(hangup_after_bridge=true) >> 2017-09-20 23:31:49.270606 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [hangup_after_bridge]=[true] >> EXECUTE sofia/external/07966677711 at 185.8.92.3 set(continue_on_fail=true) >> 2017-09-20 23:31:49.270606 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [continue_on_fail]=[true] >> EXECUTE sofia/external/07966677711 at 185.8.92.3 set(inherit_codec=true) >> 2017-09-20 23:31:49.270606 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [inherit_codec]=[true] >> EXECUTE sofia/external/07966677711 at 185.8.92.3 set(call_timeout=20) >> 2017-09-20 23:31:49.270606 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [call_timeout]=[20] >> EXECUTE sofia/external/07966677711 at 185.8.92.3 set(fail_on_single_reject=USER_BUSY) >> 2017-09-20 23:31:49.270606 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [fail_on_single_reject]=[USER_BUSY] >> EXECUTE sofia/external/07966677711 at 185.8.92.3 set(origination_caller_id_name=443307881011) >> 2017-09-20 23:31:49.270606 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [origination_caller_id_name]=[443307881011] >> EXECUTE sofia/external/07966677711 at 185.8.92.3 set(origination_caller_id_number=07966677711) >> 2017-09-20 23:31:49.270606 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [origination_caller_id_number]=[07966677711] >> EXECUTE sofia/external/07966677711 at 185.8.92.3 auth_function(in , in 07966677711, in , out AUTH_RESULT) >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:301 allocate initial structure. >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:313 initialzed configuration. >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:318 set authserver := 185.35.229.30:1812:h4nn4h. >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:318 set dictionary := /usr/local/freeswitch/conf/dictionaries/dictionary. >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:318 set seqfile := /var/run/radius.seq. >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:318 set mapfile := /usr/local/etc/radiusclient/port-id-map. >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:318 set default_realm := . >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:318 set radius_timeout := 3. >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:318 set radius_retries := 2. >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:318 set radius_deadtime := 0. >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:318 set bindaddr := 185.35.229.6. >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:371 ... radius: User-Name: 07966677711 >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:380 ... radius: User-Password: >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:391 ... radius: Called-station-Id is empty, ignoring... >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:413 Handle attribute: h323-conf-id >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:439 ... dictionary data: id:24, vendor id:9, attr type:0, attr name:h323-conf-id (589848) >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:444 ... radius: key:h323-conf-id, value:CALLID (h323-conf-id=a923b79f-e080-416c-934c-af13b570f813) as string >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:413 Handle attribute: Cisco-AVPair >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:439 ... dictionary data: id:1, vendor id:9, attr type:0, attr name:Cisco-AVPair (589825) >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:444 ... radius: key:Cisco-AVPair, value:SRCGWIP (src-gw-ip=185.8.92.3) as string >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:413 Handle attribute: Cisco-AVPair >> 2017-09-20 23:31:49.270606 [ERR] mod_rad_auth.c:178 Undefined channel variable: SRCGWNAME. >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:439 ... dictionary data: id:1, vendor id:9, attr type:0, attr name:Cisco-AVPair (589825) >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:444 ... radius: key:Cisco-AVPair, value:SRCGWNAME () as string >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:413 Handle attribute: Cisco-AVPair >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:439 ... dictionary data: id:1, vendor id:9, attr type:0, attr name:Cisco-AVPair (589825) >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:444 ... radius: key:Cisco-AVPair, value:request-type=number (request-type=number) as string >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:413 Handle attribute: Calling-Station-Id >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:439 ... dictionary data: id:31, vendor id:0, attr type:0, attr name:Calling-Station-Id (31) >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:444 ... radius: key:Calling-Station-Id, value:CALLINGNUMBER (07966677711) as string >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:413 Handle attribute: Called-Station-Id >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:439 ... dictionary data: id:30, vendor id:0, attr type:0, attr name:Called-Station-Id (30) >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:444 ... radius: key:Called-Station-Id, value:CALLEDNUMBER (443307881011) as string >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:491 sending radius packet ... >> 2017-09-20 23:31:49.290658 [DEBUG] mod_rad_auth.c:497 RADIUS Authentication OK >> 2017-09-20 23:31:49.290658 [DEBUG] mod_rad_auth.c:519 Handle attribute: CREDIT_TIME >> 2017-09-20 23:31:49.290658 [DEBUG] mod_rad_auth.c:522 attribute (CREDIT_TIME) found in radius packet >> 2017-09-20 23:31:49.290658 [DEBUG] mod_rad_auth.c:523 set variable credit_time := 7199 >> 2017-09-20 23:31:49.290658 [DEBUG] mod_rad_auth.c:519 Handle attribute: CISCO_SERVICE_INFO >> 2017-09-20 23:31:49.290658 [DEBUG] mod_rad_auth.c:522 attribute (CISCO_SERVICE_INFO) found in radius packet >> 2017-09-20 23:31:49.290658 [DEBUG] mod_rad_auth.c:523 set variable cisco_service_info := 200 >> 2017-09-20 23:31:49.290658 [DEBUG] mod_rad_auth.c:519 Handle attribute: RADIUS_RETURN_CODE >> 2017-09-20 23:31:49.290658 [DEBUG] mod_rad_auth.c:522 attribute (RADIUS_RETURN_CODE) found in radius packet >> 2017-09-20 23:31:49.290658 [DEBUG] mod_rad_auth.c:523 set variable return_code := 16 >> EXECUTE sofia/external/07966677711 at 185.8.92.3 export(credit_time=7199) >> 2017-09-20 23:31:49.290658 [DEBUG] switch_channel.c:1296 EXPORT (export_vars) [credit_time]=[7199] >> EXECUTE sofia/external/07966677711 at 185.8.92.3 export(cisco_service_info=200) >> 2017-09-20 23:31:49.290658 [DEBUG] switch_channel.c:1296 EXPORT (export_vars) [cisco_service_info]=[200] >> EXECUTE sofia/external/07966677711 at 185.8.92.3 export(sip_redirect_context=redirect) >> 2017-09-20 23:31:49.290658 [DEBUG] switch_channel.c:1296 EXPORT (export_vars) [sip_redirect_context]=[redirect] >> EXECUTE sofia/external/07966677711 at 185.8.92.3 transfer(443307881011 XML ipauth) >> 2017-09-20 23:31:49.290658 [DEBUG] switch_ivr.c:2165 (sofia/external/07966677711 at 185.8.92.3 ) State Change CS_EXECUTE -> CS_ROUTING >> 2017-09-20 23:31:49.290658 [NOTICE] switch_ivr.c:2172 Transfer sofia/external/07966677711 at 185.8.92.3 to XML[443307881011 at ipauth] >> 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:650 (sofia/external/07966677711 at 185.8.92.3 ) State EXECUTE going to sleep >> 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:584 (sofia/external/07966677711 at 185.8.92.3 ) Running State Change CS_ROUTING (Cur 1 Tot 16430) >> 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:643 (sofia/external/07966677711 at 185.8.92.3 ) State ROUTING >> 2017-09-20 23:31:49.290658 [DEBUG] mod_sofia.c:143 sofia/external/07966677711 at 185.8.92.3 SOFIA ROUTING >> 2017-09-20 23:31:49.290658 [ERR] mod_xml_radius.c:933 Result of true match: 185.35.229.6 == ^185\.35\.229\.30 >> 2017-09-20 23:31:49.290658 [ERR] mod_xml_radius.c:930 Didn't match: inbound == ^outbound >> 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:236 sofia/external/07966677711 at 185.8.92.3 Standard ROUTING >> 2017-09-20 23:31:49.290658 [INFO] mod_dialplan_xml.c:637 Processing 07966677711 <07966677711>->443307881011 in context ipauth >> Dialplan: sofia/external/07966677711 at 185.8.92.3 parsing [ipauth->Auth Calls] continue=false >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Regex (PASS) [Auth Calls] ${cisco_service_info}(200) =~ /^200$/ break=never >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(execute_on_answer=sched_hangup +${credit_time} alloted_timeout) >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action bridge({sip_invite_from_uri=sip:${sip_from_user}@${sip_network_ip}}sofia/external/${destination_number}@185.35.229.30:5060 ) >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action hangup(${bridge_hangup_cause}) >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Regex (FAIL) [Auth Calls] ${cisco_service_info}(200) =~ /^101$/ break=on-true >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Regex (FAIL) [Auth Calls] ${cisco_service_info}(200) =~ /^221$/ break=on-true >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Regex (FAIL) [Auth Calls] ${cisco_service_info}(200) =~ /^201$/ break=on-true >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Absolute Condition [Auth Calls] >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action log(INFO Were in the IPauth Context!!) >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action log(INFO credit_time=${credit_time}) >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action log(INFO cisco_service_info=${cisco_service_info}) >> 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:286 (sofia/external/07966677711 at 185.8.92.3 ) State Change CS_ROUTING -> CS_EXECUTE >> 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:643 (sofia/external/07966677711 at 185.8.92.3 ) State ROUTING going to sleep >> 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:584 (sofia/external/07966677711 at 185.8.92.3 ) Running State Change CS_EXECUTE (Cur 1 Tot 16430) >> 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:650 (sofia/external/07966677711 at 185.8.92.3 ) State EXECUTE >> 2017-09-20 23:31:49.290658 [DEBUG] mod_sofia.c:198 sofia/external/07966677711 at 185.8.92.3 SOFIA EXECUTE >> 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:328 sofia/external/07966677711 at 185.8.92.3 Standard EXECUTE >> EXECUTE sofia/external/07966677711 at 185.8.92.3 set(execute_on_answer=sched_hangup +7199 alloted_timeout) >> 2017-09-20 23:31:49.290658 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [execute_on_answer]=[sched_hangup +7199 alloted_timeout] >> EXECUTE sofia/external/07966677711 at 185.8.92.3 bridge({sip_invite_from_uri=sip:07966677711 at 185.8.92.3 }sofia/external/443307881011 at 185.35.229.30 :5060) >> 2017-09-20 23:31:49.290658 [DEBUG] switch_channel.c:1250 sofia/external/07966677711 at 185.8.92.3 EXPORTING[export_vars] [RFC2822_DATE]=[Wed, 20 Sep 2017 23:31:49 +0100] to event >> 2017-09-20 23:31:49.290658 [DEBUG] switch_channel.c:1250 sofia/external/07966677711 at 185.8.92.3 EXPORTING[export_vars] [h323-call-origin]=[originate] to event >> 2017-09-20 23:31:49.290658 [DEBUG] switch_channel.c:1250 sofia/external/07966677711 at 185.8.92.3 EXPORTING[export_vars] [credit_time]=[7199] to event >> 2017-09-20 23:31:49.290658 [DEBUG] switch_channel.c:1250 sofia/external/07966677711 at 185.8.92.3 EXPORTING[export_vars] [cisco_service_info]=[200] to event >> 2017-09-20 23:31:49.290658 [DEBUG] switch_channel.c:1250 sofia/external/07966677711 at 185.8.92.3 EXPORTING[export_vars] [sip_redirect_context]=[redirect] to event >> 2017-09-20 23:31:49.290658 [DEBUG] switch_ivr_originate.c:2142 Parsing global variables >> 2017-09-20 23:31:49.290658 [NOTICE] switch_channel.c:1104 New Channel sofia/external/443307881011 at 185.35.229.30 :5060 [848843db-cad9-48a8-a4bd-2ac6b6b77dff] >> 2017-09-20 23:31:49.290658 [DEBUG] mod_sofia.c:4819 (sofia/external/443307881011 at 185.35.229.30 :5060) State Change CS_NEW -> CS_INIT >> 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:584 (sofia/external/443307881011 at 185.35.229.30 :5060) Running State Change CS_INIT (Cur 2 Tot 16431) >> 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:627 (sofia/external/443307881011 at 185.35.229.30 :5060) State INIT >> 2017-09-20 23:31:49.290658 [DEBUG] mod_sofia.c:90 sofia/external/443307881011 at 185.35.229.30 :5060 SOFIA INIT >> 2017-09-20 23:31:49.290658 [DEBUG] sofia_glue.c:1295 sofia/external/443307881011 at 185.35.229.30 :5060 sending invite version: 1.6.19 git febfb38 2017-09-14 23:14:02Z 64bit >> Local SDP: >> v=0 >> o=FreeSWITCH 1505929783 1505929784 IN IP4 185.35.229.6 >> s=FreeSWITCH >> c=IN IP4 185.35.229.6 >> t=0 0 >> m=audio 16926 RTP/AVP 8 101 13 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=rtpmap:13 CN/8000 >> a=ptime:20 >> a=sendrecv >> >> 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:40 sofia/external/443307881011 at 185.35.229.30 :5060 Standard INIT >> 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:48 (sofia/external/443307881011 at 185.35.229.30 :5060) State Change CS_INIT -> CS_ROUTING >> 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:627 (sofia/external/443307881011 at 185.35.229.30 :5060) State INIT going to sleep >> 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:584 (sofia/external/443307881011 at 185.35.229.30 :5060) Running State Change CS_ROUTING (Cur 2 Tot 16431) >> 2017-09-20 23:31:49.290658 [DEBUG] sofia.c:7084 Channel sofia/external/443307881011 at 185.35.229.30 :5060 entering state [calling][0] >> 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:643 (sofia/external/443307881011 at 185.35.229.30 :5060) State ROUTING >> 2017-09-20 23:31:49.290658 [DEBUG] mod_sofia.c:143 sofia/external/443307881011 at 185.35.229.30 :5060 SOFIA ROUTING >> 2017-09-20 23:31:49.290658 [DEBUG] switch_ivr_originate.c:67 (sofia/external/443307881011 at 185.35.229.30 :5060) State Change CS_ROUTING -> CS_CONSUME_MEDIA >> 2017-09-20 23:31:49.290658 [ERR] mod_xml_radius.c:930 Didn't match: 185.35.229.30:5060 == ^185\.35\.229\.30 >> 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:643 (sofia/external/443307881011 at 185.35.229.30 :5060) State ROUTING going to sleep >> 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:584 (sofia/external/443307881011 at 185.35.229.30 :5060) Running State Change CS_CONSUME_MEDIA (Cur 2 Tot 16431) >> 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:662 (sofia/external/443307881011 at 185.35.229.30 :5060) State CONSUME_MEDIA >> 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:662 (sofia/external/443307881011 at 185.35.229.30 :5060) State CONSUME_MEDIA going to sleep >> 2017-09-20 23:31:49.350696 [DEBUG] sofia.c:6540 Redirect: Transfering to joehouse XML redirect >> 2017-09-20 23:31:49.350696 [DEBUG] switch_ivr.c:2165 (sofia/external/07966677711 at 185.8.92.3 ) State Change CS_EXECUTE -> CS_ROUTING >> 2017-09-20 23:31:49.350696 [NOTICE] switch_ivr.c:2172 Transfer sofia/external/07966677711 at 185.8.92.3 to XML[joehouse at redirect] >> 2017-09-20 23:31:49.350696 [NOTICE] sofia.c:6549 Hangup sofia/external/443307881011 at 185.35.229.30 :5060 [CS_CONSUME_MEDIA] [REDIRECTION_TO_NEW_DESTINATION] >> 2017-09-20 23:31:49.350696 [DEBUG] sofia.c:1453 Channel is already hungup. >> 2017-09-20 23:31:49.350696 [DEBUG] sofia.c:1453 Channel is already hungup. >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:584 (sofia/external/443307881011 at 185.35.229.30 :5060) Running State Change CS_HANGUP (Cur 2 Tot 16431) >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:850 (sofia/external/443307881011 at 185.35.229.30 :5060) Callstate Change DOWN -> HANGUP >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:852 (sofia/external/443307881011 at 185.35.229.30 :5060) State HANGUP >> 2017-09-20 23:31:49.350696 [DEBUG] mod_sofia.c:438 Channel sofia/external/443307881011 at 185.35.229.30 :5060 hanging up, cause: REDIRECTION_TO_NEW_DESTINATION >> 2017-09-20 23:31:49.350696 [DEBUG] mod_sofia.c:502 Sending CANCEL to sofia/external/443307881011 at 185.35.229.30 :5060 >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:60 sofia/external/443307881011 at 185.35.229.30 :5060 Standard HANGUP, cause: REDIRECTION_TO_NEW_DESTINATION >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:852 (sofia/external/443307881011 at 185.35.229.30 :5060) State HANGUP going to sleep >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:619 (sofia/external/443307881011 at 185.35.229.30 :5060) State Change CS_HANGUP -> CS_REPORTING >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:584 (sofia/external/443307881011 at 185.35.229.30 :5060) Running State Change CS_REPORTING (Cur 2 Tot 16431) >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:938 (sofia/external/443307881011 at 185.35.229.30 :5060) State REPORTING >> 2017-09-20 23:31:49.350696 [ERR] mod_xml_radius.c:930 Didn't match: 185.35.229.30:5060 == ^185\.35\.229\.30 >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:174 sofia/external/443307881011 at 185.35.229.30 :5060 Standard REPORTING, cause: REDIRECTION_TO_NEW_DESTINATION >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:938 (sofia/external/443307881011 at 185.35.229.30 :5060) State REPORTING going to sleep >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:610 (sofia/external/443307881011 at 185.35.229.30 :5060) State Change CS_REPORTING -> CS_DESTROY >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_session.c:1665 Session 16431 (sofia/external/443307881011 at 185.35.229.30 :5060) Locked, Waiting on external entities >> 2017-09-20 23:31:49.350696 [DEBUG] switch_ivr_originate.c:3837 Originate Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL] >> 2017-09-20 23:31:49.350696 [NOTICE] switch_core_session.c:1683 Session 16431 (sofia/external/443307881011 at 185.35.229.30 :5060) Ended >> 2017-09-20 23:31:49.350696 [NOTICE] switch_core_session.c:1687 Close Channel sofia/external/443307881011 at 185.35.229.30 :5060 [CS_DESTROY] >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:741 (sofia/external/443307881011 at 185.35.229.30 :5060) Running State Change CS_DESTROY (Cur 1 Tot 16431) >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:751 (sofia/external/443307881011 at 185.35.229.30 :5060) State DESTROY >> 2017-09-20 23:31:49.350696 [DEBUG] mod_sofia.c:343 sofia/external/443307881011 at 185.35.229.30 :5060 SOFIA DESTROY >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:181 sofia/external/443307881011 at 185.35.229.30 :5060 Standard DESTROY >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:751 (sofia/external/443307881011 at 185.35.229.30 :5060) State DESTROY going to sleep >> 2017-09-20 23:31:49.350696 [INFO] mod_dptools.c:3436 Originate Failed. Cause: ORIGINATOR_CANCEL >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:650 (sofia/external/07966677711 at 185.8.92.3 ) State EXECUTE going to sleep >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:584 (sofia/external/07966677711 at 185.8.92.3 ) Running State Change CS_ROUTING (Cur 1 Tot 16431) >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:643 (sofia/external/07966677711 at 185.8.92.3 ) State ROUTING >> 2017-09-20 23:31:49.350696 [DEBUG] mod_sofia.c:143 sofia/external/07966677711 at 185.8.92.3 SOFIA ROUTING >> 2017-09-20 23:31:49.350696 [ERR] mod_xml_radius.c:933 Result of true match: 185.35.229.6 == ^185\.35\.229\.30 >> 2017-09-20 23:31:49.350696 [ERR] mod_xml_radius.c:930 Didn't match: inbound == ^outbound >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:236 sofia/external/07966677711 at 185.8.92.3 Standard ROUTING >> 2017-09-20 23:31:49.350696 [INFO] mod_dialplan_xml.c:637 Processing 07966677711 <07966677711>->joehouse in context redirect >> Dialplan: sofia/external/07966677711 at 185.8.92.3 parsing [redirect->Redirect Calls] continue=false >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Regex (PASS) [Redirect Calls] ${sip_redirect_contact_0}(>;src_number=07966677711;q=1.00) =~ /^> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(process_cdr=true) >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action log(ALERT Local User) >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(accountcode=jerasoftoutbound) >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action log(ALERT joehouse) >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action bridge(user/joehouse at sip.biznetuk.com ) >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:286 (sofia/external/07966677711 at 185.8.92.3 ) State Change CS_ROUTING -> CS_EXECUTE >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:643 (sofia/external/07966677711 at 185.8.92.3 ) State ROUTING going to sleep >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:584 (sofia/external/07966677711 at 185.8.92.3 ) Running State Change CS_EXECUTE (Cur 1 Tot 16431) >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:650 (sofia/external/07966677711 at 185.8.92.3 ) State EXECUTE >> 2017-09-20 23:31:49.350696 [DEBUG] mod_sofia.c:198 sofia/external/07966677711 at 185.8.92.3 SOFIA EXECUTE >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:328 sofia/external/07966677711 at 185.8.92.3 Standard EXECUTE >> EXECUTE sofia/external/07966677711 at 185.8.92.3 set(process_cdr=true) >> 2017-09-20 23:31:49.350696 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [process_cdr]=[true] >> EXECUTE sofia/external/07966677711 at 185.8.92.3 log(ALERT Local User) >> 2017-09-20 23:31:49.350696 [ALERT] mod_dptools.c:1742 Local User >> EXECUTE sofia/external/07966677711 at 185.8.92.3 set(accountcode=jerasoftoutbound) >> 2017-09-20 23:31:49.350696 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [accountcode]=[jerasoftoutbound] >> EXECUTE sofia/external/07966677711 at 185.8.92.3 log(ALERT joehouse) >> 2017-09-20 23:31:49.350696 [ALERT] mod_dptools.c:1742 joehouse >> EXECUTE sofia/external/07966677711 at 185.8.92.3 bridge(user/joehouse at sip.biznetuk.com ) >> 2017-09-20 23:31:49.350696 [DEBUG] switch_channel.c:1250 sofia/external/07966677711 at 185.8.92.3 EXPORTING[export_vars] [RFC2822_DATE]=[Wed, 20 Sep 2017 23:31:49 +0100] to event >> 2017-09-20 23:31:49.350696 [DEBUG] switch_channel.c:1250 sofia/external/07966677711 at 185.8.92.3 EXPORTING[export_vars] [h323-call-origin]=[originate] to event >> 2017-09-20 23:31:49.350696 [DEBUG] switch_channel.c:1250 sofia/external/07966677711 at 185.8.92.3 EXPORTING[export_vars] [credit_time]=[7199] to event >> 2017-09-20 23:31:49.350696 [DEBUG] switch_channel.c:1250 sofia/external/07966677711 at 185.8.92.3 EXPORTING[export_vars] [cisco_service_info]=[200] to event >> 2017-09-20 23:31:49.350696 [DEBUG] switch_channel.c:1250 sofia/external/07966677711 at 185.8.92.3 EXPORTING[export_vars] [sip_redirect_context]=[redirect] to event >> 2017-09-20 23:31:49.350696 [DEBUG] switch_ivr_originate.c:2142 Parsing global variables >> 2017-09-20 23:31:49.350696 [WARNING] mod_dptools.c:4184 Can't find user [joehouse at sip.biznetuk.com ] >> 2017-09-20 23:31:49.350696 [NOTICE] switch_ivr_originate.c:2851 Cannot create outgoing channel of type [user] cause: [SUBSCRIBER_ABSENT] >> 2017-09-20 23:31:49.350696 [DEBUG] switch_ivr_originate.c:3848 Originate Resulted in Error Cause: 20 [SUBSCRIBER_ABSENT] >> 2017-09-20 23:31:49.350696 [INFO] mod_dptools.c:3436 Originate Failed. Cause: SUBSCRIBER_ABSENT >> 2017-09-20 23:31:49.350696 [NOTICE] switch_core_state_machine.c:385 sofia/external/07966677711 at 185.8.92.3 has executed the last dialplan instruction, hanging up. >> 2017-09-20 23:31:49.350696 [NOTICE] switch_core_state_machine.c:387 Hangup sofia/external/07966677711 at 185.8.92.3 [CS_EXECUTE] [NORMAL_CLEARING] >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:650 (sofia/external/07966677711 at 185.8.92.3 ) State EXECUTE going to sleep >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:584 (sofia/external/07966677711 at 185.8.92.3 ) Running State Change CS_HANGUP (Cur 1 Tot 16431) >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:850 (sofia/external/07966677711 at 185.8.92.3 ) Callstate Change RINGING -> HANGUP >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:852 (sofia/external/07966677711 at 185.8.92.3 ) State HANGUP >> 2017-09-20 23:31:49.350696 [DEBUG] mod_sofia.c:438 Channel sofia/external/07966677711 at 185.8.92.3 hanging up, cause: NORMAL_CLEARING >> 2017-09-20 23:31:49.350696 [DEBUG] mod_sofia.c:577 Responding to INVITE with: 480 >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:60 sofia/external/07966677711 at 185.8.92.3 Standard HANGUP, cause: NORMAL_CLEARING >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:852 (sofia/external/07966677711 at 185.8.92.3 ) State HANGUP going to sleep >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:619 (sofia/external/07966677711 at 185.8.92.3 ) State Change CS_HANGUP -> CS_REPORTING >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:584 (sofia/external/07966677711 at 185.8.92.3 ) Running State Change CS_REPORTING (Cur 1 Tot 16431) >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:938 (sofia/external/07966677711 at 185.8.92.3 ) State REPORTING >> 2017-09-20 23:31:49.350696 [ERR] mod_xml_radius.c:933 Result of true match: 185.35.229.6 == ^185\.35\.229\.30 >> 2017-09-20 23:31:49.350696 [ERR] mod_xml_radius.c:930 Didn't match: inbound == ^outbound >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:174 sofia/external/07966677711 at 185.8.92.3 Standard REPORTING, cause: NORMAL_CLEARING >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:938 (sofia/external/07966677711 at 185.8.92.3 ) State REPORTING going to sleep >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:610 (sofia/external/07966677711 at 185.8.92.3 ) State Change CS_REPORTING -> CS_DESTROY >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_session.c:1665 Session 16430 (sofia/external/07966677711 at 185.8.92.3 ) Locked, Waiting on external entities >> 2017-09-20 23:31:49.350696 [NOTICE] switch_core_session.c:1683 Session 16430 (sofia/external/07966677711 at 185.8.92.3 ) Ended >> 2017-09-20 23:31:49.350696 [NOTICE] switch_core_session.c:1687 Close Channel sofia/external/07966677711 at 185.8.92.3 [CS_DESTROY] >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:741 (sofia/external/07966677711 at 185.8.92.3 ) Running State Change CS_DESTROY (Cur 0 Tot 16431) >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:751 (sofia/external/07966677711 at 185.8.92.3 ) State DESTROY >> 2017-09-20 23:31:49.350696 [DEBUG] mod_sofia.c:343 sofia/external/07966677711 at 185.8.92.3 SOFIA DESTROY >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:181 sofia/external/07966677711 at 185.8.92.3 Standard DESTROY >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:751 (sofia/external/07966677711 at 185.8.92.3 ) State DESTROY going to sleep >> >> its saying there is no endpoint registered yet the output of show registrations shows there is >> >> freeswitch at BiznetukSBC1> show registrations >> reg_user,realm,token,url,expires,network_ip,network_port,network_proto,hostname,metadata,registration_uuid >> joehouse,sip.biznetuk.com ,1030250472 at 192_168_1_101,sofia/internal/sip:joehouse at 185.35.230.2:5382,1505947110,185.35.230.2,5382,udp,BiznetukSBC1,,60484000-aa2d-4b21-9d58-11e7d4155f1e >> >> 1 total. >> >> >> Any ideas? >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From joelists at tm.net.uk Thu Sep 21 19:46:36 2017 From: joelists at tm.net.uk (Joseph Waite) Date: Thu, 21 Sep 2017 20:46:36 +0100 Subject: [Freeswitch-users] Routing calls to registered user. In-Reply-To: References: <995A631E-8C13-4E34-9735-806E20C5FA6C@tm.net.uk> Message-ID: <2480B0A5-CC7A-433E-AC26-02B15829575C@tm.net.uk> On the same line, can anyone shed some light on the correct way to pass the number called to a registered user. The scenario is were a DID provider, call comes in on a DID routed to a registered user, who effectively are registering to FS as a trunk from their own PBX. What would be the correct method to send the DID called in the invite to registered endpoint? We would normally be in the To: in the invite, but not sure this would work. > On 21 Sep 2017, at 18:51, Michael Jerris wrote: > > https://freeswitch.org/confluence/display/FREESWITCH/XML+User+Directory#XMLUserDirectory-DialString > >> On Sep 20, 2017, at 6:36 PM, Joseph Waite > wrote: >> >> Hi Guys >> >> Knocking my head against this. I am trying to route calls to registered endpoints. But it is not working. >> >> The log of the whole call is as follows >> >> 2017-09-20 23:31:49.230782 [NOTICE] switch_channel.c:1104 New Channel sofia/external/07966677711 at 185.8.92.3 [a923b79f-e080-416c-934c-af13b570f813] >> 2017-09-20 23:31:49.230782 [DEBUG] switch_core_state_machine.c:584 (sofia/external/07966677711 at 185.8.92.3 ) Running State Change CS_NEW (Cur 1 Tot 16430) >> 2017-09-20 23:31:49.230782 [DEBUG] sofia.c:9873 sofia/external/07966677711 at 185.8.92.3 receiving invite from 185.8.92.3:5062 version: 1.6.19 git febfb38 2017-09-14 23:14:02Z 64bit >> 2017-09-20 23:31:49.270606 [DEBUG] sofia.c:7084 Channel sofia/external/07966677711 at 185.8.92.3 entering state [received][100] >> 2017-09-20 23:31:49.270606 [DEBUG] sofia.c:7094 Remote SDP: >> v=0 >> o=FreeSWITCH 1505928870 1505928871 IN IP4 185.8.92.3 >> s=FreeSWITCH >> c=IN IP4 185.8.92.3 >> t=0 0 >> m=audio 17854 RTP/AVP 8 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=silenceSupp:off - - - - >> a=ptime:20 >> >> 2017-09-20 23:31:49.270606 [DEBUG] sofia.c:7486 (sofia/external/07966677711 at 185.8.92.3 ) State Change CS_NEW -> CS_INIT >> 2017-09-20 23:31:49.270606 [DEBUG] switch_core_state_machine.c:603 (sofia/external/07966677711 at 185.8.92.3 ) State NEW >> 2017-09-20 23:31:49.270606 [DEBUG] switch_core_state_machine.c:584 (sofia/external/07966677711 at 185.8.92.3 ) Running State Change CS_INIT (Cur 1 Tot 16430) >> 2017-09-20 23:31:49.270606 [DEBUG] switch_core_state_machine.c:627 (sofia/external/07966677711 at 185.8.92.3 ) State INIT >> 2017-09-20 23:31:49.270606 [DEBUG] mod_sofia.c:90 sofia/external/07966677711 at 185.8.92.3 SOFIA INIT >> 2017-09-20 23:31:49.270606 [DEBUG] switch_core_state_machine.c:40 sofia/external/07966677711 at 185.8.92.3 Standard INIT >> 2017-09-20 23:31:49.270606 [DEBUG] switch_core_state_machine.c:48 (sofia/external/07966677711 at 185.8.92.3 ) State Change CS_INIT -> CS_ROUTING >> 2017-09-20 23:31:49.270606 [DEBUG] switch_core_state_machine.c:627 (sofia/external/07966677711 at 185.8.92.3 ) State INIT going to sleep >> 2017-09-20 23:31:49.270606 [DEBUG] switch_core_state_machine.c:584 (sofia/external/07966677711 at 185.8.92.3 ) Running State Change CS_ROUTING (Cur 1 Tot 16430) >> 2017-09-20 23:31:49.270606 [DEBUG] switch_channel.c:2249 (sofia/external/07966677711 at 185.8.92.3 ) Callstate Change DOWN -> RINGING >> 2017-09-20 23:31:49.270606 [DEBUG] switch_core_state_machine.c:643 (sofia/external/07966677711 at 185.8.92.3 ) State ROUTING >> 2017-09-20 23:31:49.270606 [DEBUG] mod_sofia.c:143 sofia/external/07966677711 at 185.8.92.3 SOFIA ROUTING >> 2017-09-20 23:31:49.270606 [ERR] mod_xml_radius.c:933 Result of true match: 185.35.229.6 == ^185\.35\.229\.30 >> 2017-09-20 23:31:49.270606 [ERR] mod_xml_radius.c:930 Didn't match: inbound == ^outbound >> 2017-09-20 23:31:49.270606 [DEBUG] switch_core_state_machine.c:236 sofia/external/07966677711 at 185.8.92.3 Standard ROUTING >> 2017-09-20 23:31:49.270606 [INFO] mod_dialplan_xml.c:637 Processing 07966677711 <07966677711>->443307881011 in context public >> Dialplan: sofia/external/07966677711 at 185.8.92.3 parsing [public->unloop] continue=false >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false >> Dialplan: sofia/external/07966677711 at 185.8.92.3 parsing [public->outside_call] continue=true >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Absolute Condition [outside_call] >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(outside_call=true) >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) >> Dialplan: sofia/external/07966677711 at 185.8.92.3 parsing [public->call_debug] continue=true >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never >> Dialplan: sofia/external/07966677711 at 185.8.92.3 parsing [public->rejections] continue=false >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Regex (FAIL) [rejections] ${radius_auth_result}() =~ /2/ break=on-false >> Dialplan: sofia/external/07966677711 at 185.8.92.3 parsing [public->timedouts] continue=false >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Regex (FAIL) [timedouts] ${radius_auth_result}() =~ /1/ break=on-false >> Dialplan: sofia/external/07966677711 at 185.8.92.3 parsing [public->JeraSoft VCS Routing] continue=false >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Regex (PASS) [JeraSoft VCS Routing] destination_number(443307881011) =~ /^(.+)$/ break=on-false >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action export(nolocal:h323-call-origin=originate) >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(sip_h_X-accountcode=${accountcode}) >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(call_direction=outbound) >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(hangup_after_bridge=true) >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(continue_on_fail=true) >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(inherit_codec=true) >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(call_timeout=20) >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(fail_on_single_reject=USER_BUSY) >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(origination_caller_id_name=${sip_req_user}) >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(origination_caller_id_number=${sip_from_user}) >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(SRCGWIP=src-gw-ip=${network_addr}) INLINE >> EXECUTE sofia/external/07966677711 at 185.8.92.3 set(SRCGWIP=src-gw-ip=185.8.92.3) >> 2017-09-20 23:31:49.270606 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [SRCGWIP]=[src-gw-ip=185.8.92.3] >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(CALLID=h323-conf-id=${uuid}) INLINE >> EXECUTE sofia/external/07966677711 at 185.8.92.3 set(CALLID=h323-conf-id=a923b79f-e080-416c-934c-af13b570f813) >> 2017-09-20 23:31:49.270606 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [CALLID]=[h323-conf-id=a923b79f-e080-416c-934c-af13b570f813] >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(CALLINGNUMBER=${caller_id_number}) INLINE >> EXECUTE sofia/external/07966677711 at 185.8.92.3 set(CALLINGNUMBER=07966677711) >> 2017-09-20 23:31:49.270606 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [CALLINGNUMBER]=[07966677711] >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(CALLEDNUMBER=${destination_number}) INLINE >> EXECUTE sofia/external/07966677711 at 185.8.92.3 set(CALLEDNUMBER=443307881011) >> 2017-09-20 23:31:49.270606 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [CALLEDNUMBER]=[443307881011] >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action auth_function(in ${DIALED_NUMBER}, in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT) >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action export(credit_time=${credit_time}) >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action export(cisco_service_info=${cisco_service_info) >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action export(sip_redirect_context=redirect) >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action transfer(443307881011 XML ipauth) >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action hangup(${bridge_hangup_cause}) >> 2017-09-20 23:31:49.270606 [DEBUG] switch_core_state_machine.c:286 (sofia/external/07966677711 at 185.8.92.3 ) State Change CS_ROUTING -> CS_EXECUTE >> 2017-09-20 23:31:49.270606 [DEBUG] switch_core_state_machine.c:643 (sofia/external/07966677711 at 185.8.92.3 ) State ROUTING going to sleep >> 2017-09-20 23:31:49.270606 [DEBUG] switch_core_state_machine.c:584 (sofia/external/07966677711 at 185.8.92.3 ) Running State Change CS_EXECUTE (Cur 1 Tot 16430) >> 2017-09-20 23:31:49.270606 [DEBUG] switch_core_state_machine.c:650 (sofia/external/07966677711 at 185.8.92.3 ) State EXECUTE >> 2017-09-20 23:31:49.270606 [DEBUG] mod_sofia.c:198 sofia/external/07966677711 at 185.8.92.3 SOFIA EXECUTE >> 2017-09-20 23:31:49.270606 [DEBUG] switch_core_state_machine.c:328 sofia/external/07966677711 at 185.8.92.3 Standard EXECUTE >> EXECUTE sofia/external/07966677711 at 185.8.92.3 set(outside_call=true) >> 2017-09-20 23:31:49.270606 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [outside_call]=[true] >> EXECUTE sofia/external/07966677711 at 185.8.92.3 export(RFC2822_DATE=Wed, 20 Sep 2017 23:31:49 +0100) >> 2017-09-20 23:31:49.270606 [DEBUG] switch_channel.c:1296 EXPORT (export_vars) [RFC2822_DATE]=[Wed, 20 Sep 2017 23:31:49 +0100] >> EXECUTE sofia/external/07966677711 at 185.8.92.3 export(nolocal:h323-call-origin=originate) >> 2017-09-20 23:31:49.270606 [DEBUG] switch_channel.c:1296 EXPORT (export_vars) (REMOTE ONLY) [h323-call-origin]=[originate] >> EXECUTE sofia/external/07966677711 at 185.8.92.3 set(sip_h_X-accountcode=) >> 2017-09-20 23:31:49.270606 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [sip_h_X-accountcode]=[UNDEF] >> EXECUTE sofia/external/07966677711 at 185.8.92.3 set(call_direction=outbound) >> 2017-09-20 23:31:49.270606 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [call_direction]=[outbound] >> EXECUTE sofia/external/07966677711 at 185.8.92.3 set(hangup_after_bridge=true) >> 2017-09-20 23:31:49.270606 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [hangup_after_bridge]=[true] >> EXECUTE sofia/external/07966677711 at 185.8.92.3 set(continue_on_fail=true) >> 2017-09-20 23:31:49.270606 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [continue_on_fail]=[true] >> EXECUTE sofia/external/07966677711 at 185.8.92.3 set(inherit_codec=true) >> 2017-09-20 23:31:49.270606 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [inherit_codec]=[true] >> EXECUTE sofia/external/07966677711 at 185.8.92.3 set(call_timeout=20) >> 2017-09-20 23:31:49.270606 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [call_timeout]=[20] >> EXECUTE sofia/external/07966677711 at 185.8.92.3 set(fail_on_single_reject=USER_BUSY) >> 2017-09-20 23:31:49.270606 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [fail_on_single_reject]=[USER_BUSY] >> EXECUTE sofia/external/07966677711 at 185.8.92.3 set(origination_caller_id_name=443307881011) >> 2017-09-20 23:31:49.270606 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [origination_caller_id_name]=[443307881011] >> EXECUTE sofia/external/07966677711 at 185.8.92.3 set(origination_caller_id_number=07966677711) >> 2017-09-20 23:31:49.270606 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [origination_caller_id_number]=[07966677711] >> EXECUTE sofia/external/07966677711 at 185.8.92.3 auth_function(in , in 07966677711, in , out AUTH_RESULT) >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:301 allocate initial structure. >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:313 initialzed configuration. >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:318 set authserver := 185.35.229.30:1812:h4nn4h. >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:318 set dictionary := /usr/local/freeswitch/conf/dictionaries/dictionary. >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:318 set seqfile := /var/run/radius.seq. >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:318 set mapfile := /usr/local/etc/radiusclient/port-id-map. >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:318 set default_realm := . >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:318 set radius_timeout := 3. >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:318 set radius_retries := 2. >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:318 set radius_deadtime := 0. >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:318 set bindaddr := 185.35.229.6. >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:371 ... radius: User-Name: 07966677711 >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:380 ... radius: User-Password: >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:391 ... radius: Called-station-Id is empty, ignoring... >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:413 Handle attribute: h323-conf-id >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:439 ... dictionary data: id:24, vendor id:9, attr type:0, attr name:h323-conf-id (589848) >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:444 ... radius: key:h323-conf-id, value:CALLID (h323-conf-id=a923b79f-e080-416c-934c-af13b570f813) as string >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:413 Handle attribute: Cisco-AVPair >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:439 ... dictionary data: id:1, vendor id:9, attr type:0, attr name:Cisco-AVPair (589825) >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:444 ... radius: key:Cisco-AVPair, value:SRCGWIP (src-gw-ip=185.8.92.3) as string >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:413 Handle attribute: Cisco-AVPair >> 2017-09-20 23:31:49.270606 [ERR] mod_rad_auth.c:178 Undefined channel variable: SRCGWNAME. >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:439 ... dictionary data: id:1, vendor id:9, attr type:0, attr name:Cisco-AVPair (589825) >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:444 ... radius: key:Cisco-AVPair, value:SRCGWNAME () as string >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:413 Handle attribute: Cisco-AVPair >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:439 ... dictionary data: id:1, vendor id:9, attr type:0, attr name:Cisco-AVPair (589825) >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:444 ... radius: key:Cisco-AVPair, value:request-type=number (request-type=number) as string >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:413 Handle attribute: Calling-Station-Id >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:439 ... dictionary data: id:31, vendor id:0, attr type:0, attr name:Calling-Station-Id (31) >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:444 ... radius: key:Calling-Station-Id, value:CALLINGNUMBER (07966677711) as string >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:413 Handle attribute: Called-Station-Id >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:439 ... dictionary data: id:30, vendor id:0, attr type:0, attr name:Called-Station-Id (30) >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:444 ... radius: key:Called-Station-Id, value:CALLEDNUMBER (443307881011) as string >> 2017-09-20 23:31:49.270606 [DEBUG] mod_rad_auth.c:491 sending radius packet ... >> 2017-09-20 23:31:49.290658 [DEBUG] mod_rad_auth.c:497 RADIUS Authentication OK >> 2017-09-20 23:31:49.290658 [DEBUG] mod_rad_auth.c:519 Handle attribute: CREDIT_TIME >> 2017-09-20 23:31:49.290658 [DEBUG] mod_rad_auth.c:522 attribute (CREDIT_TIME) found in radius packet >> 2017-09-20 23:31:49.290658 [DEBUG] mod_rad_auth.c:523 set variable credit_time := 7199 >> 2017-09-20 23:31:49.290658 [DEBUG] mod_rad_auth.c:519 Handle attribute: CISCO_SERVICE_INFO >> 2017-09-20 23:31:49.290658 [DEBUG] mod_rad_auth.c:522 attribute (CISCO_SERVICE_INFO) found in radius packet >> 2017-09-20 23:31:49.290658 [DEBUG] mod_rad_auth.c:523 set variable cisco_service_info := 200 >> 2017-09-20 23:31:49.290658 [DEBUG] mod_rad_auth.c:519 Handle attribute: RADIUS_RETURN_CODE >> 2017-09-20 23:31:49.290658 [DEBUG] mod_rad_auth.c:522 attribute (RADIUS_RETURN_CODE) found in radius packet >> 2017-09-20 23:31:49.290658 [DEBUG] mod_rad_auth.c:523 set variable return_code := 16 >> EXECUTE sofia/external/07966677711 at 185.8.92.3 export(credit_time=7199) >> 2017-09-20 23:31:49.290658 [DEBUG] switch_channel.c:1296 EXPORT (export_vars) [credit_time]=[7199] >> EXECUTE sofia/external/07966677711 at 185.8.92.3 export(cisco_service_info=200) >> 2017-09-20 23:31:49.290658 [DEBUG] switch_channel.c:1296 EXPORT (export_vars) [cisco_service_info]=[200] >> EXECUTE sofia/external/07966677711 at 185.8.92.3 export(sip_redirect_context=redirect) >> 2017-09-20 23:31:49.290658 [DEBUG] switch_channel.c:1296 EXPORT (export_vars) [sip_redirect_context]=[redirect] >> EXECUTE sofia/external/07966677711 at 185.8.92.3 transfer(443307881011 XML ipauth) >> 2017-09-20 23:31:49.290658 [DEBUG] switch_ivr.c:2165 (sofia/external/07966677711 at 185.8.92.3 ) State Change CS_EXECUTE -> CS_ROUTING >> 2017-09-20 23:31:49.290658 [NOTICE] switch_ivr.c:2172 Transfer sofia/external/07966677711 at 185.8.92.3 to XML[443307881011 at ipauth] >> 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:650 (sofia/external/07966677711 at 185.8.92.3 ) State EXECUTE going to sleep >> 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:584 (sofia/external/07966677711 at 185.8.92.3 ) Running State Change CS_ROUTING (Cur 1 Tot 16430) >> 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:643 (sofia/external/07966677711 at 185.8.92.3 ) State ROUTING >> 2017-09-20 23:31:49.290658 [DEBUG] mod_sofia.c:143 sofia/external/07966677711 at 185.8.92.3 SOFIA ROUTING >> 2017-09-20 23:31:49.290658 [ERR] mod_xml_radius.c:933 Result of true match: 185.35.229.6 == ^185\.35\.229\.30 >> 2017-09-20 23:31:49.290658 [ERR] mod_xml_radius.c:930 Didn't match: inbound == ^outbound >> 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:236 sofia/external/07966677711 at 185.8.92.3 Standard ROUTING >> 2017-09-20 23:31:49.290658 [INFO] mod_dialplan_xml.c:637 Processing 07966677711 <07966677711>->443307881011 in context ipauth >> Dialplan: sofia/external/07966677711 at 185.8.92.3 parsing [ipauth->Auth Calls] continue=false >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Regex (PASS) [Auth Calls] ${cisco_service_info}(200) =~ /^200$/ break=never >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(execute_on_answer=sched_hangup +${credit_time} alloted_timeout) >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action bridge({sip_invite_from_uri=sip:${sip_from_user}@${sip_network_ip}}sofia/external/${destination_number}@185.35.229.30:5060 ) >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action hangup(${bridge_hangup_cause}) >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Regex (FAIL) [Auth Calls] ${cisco_service_info}(200) =~ /^101$/ break=on-true >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Regex (FAIL) [Auth Calls] ${cisco_service_info}(200) =~ /^221$/ break=on-true >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Regex (FAIL) [Auth Calls] ${cisco_service_info}(200) =~ /^201$/ break=on-true >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Absolute Condition [Auth Calls] >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action log(INFO Were in the IPauth Context!!) >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action log(INFO credit_time=${credit_time}) >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action log(INFO cisco_service_info=${cisco_service_info}) >> 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:286 (sofia/external/07966677711 at 185.8.92.3 ) State Change CS_ROUTING -> CS_EXECUTE >> 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:643 (sofia/external/07966677711 at 185.8.92.3 ) State ROUTING going to sleep >> 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:584 (sofia/external/07966677711 at 185.8.92.3 ) Running State Change CS_EXECUTE (Cur 1 Tot 16430) >> 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:650 (sofia/external/07966677711 at 185.8.92.3 ) State EXECUTE >> 2017-09-20 23:31:49.290658 [DEBUG] mod_sofia.c:198 sofia/external/07966677711 at 185.8.92.3 SOFIA EXECUTE >> 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:328 sofia/external/07966677711 at 185.8.92.3 Standard EXECUTE >> EXECUTE sofia/external/07966677711 at 185.8.92.3 set(execute_on_answer=sched_hangup +7199 alloted_timeout) >> 2017-09-20 23:31:49.290658 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [execute_on_answer]=[sched_hangup +7199 alloted_timeout] >> EXECUTE sofia/external/07966677711 at 185.8.92.3 bridge({sip_invite_from_uri=sip:07966677711 at 185.8.92.3 }sofia/external/443307881011 at 185.35.229.30 :5060) >> 2017-09-20 23:31:49.290658 [DEBUG] switch_channel.c:1250 sofia/external/07966677711 at 185.8.92.3 EXPORTING[export_vars] [RFC2822_DATE]=[Wed, 20 Sep 2017 23:31:49 +0100] to event >> 2017-09-20 23:31:49.290658 [DEBUG] switch_channel.c:1250 sofia/external/07966677711 at 185.8.92.3 EXPORTING[export_vars] [h323-call-origin]=[originate] to event >> 2017-09-20 23:31:49.290658 [DEBUG] switch_channel.c:1250 sofia/external/07966677711 at 185.8.92.3 EXPORTING[export_vars] [credit_time]=[7199] to event >> 2017-09-20 23:31:49.290658 [DEBUG] switch_channel.c:1250 sofia/external/07966677711 at 185.8.92.3 EXPORTING[export_vars] [cisco_service_info]=[200] to event >> 2017-09-20 23:31:49.290658 [DEBUG] switch_channel.c:1250 sofia/external/07966677711 at 185.8.92.3 EXPORTING[export_vars] [sip_redirect_context]=[redirect] to event >> 2017-09-20 23:31:49.290658 [DEBUG] switch_ivr_originate.c:2142 Parsing global variables >> 2017-09-20 23:31:49.290658 [NOTICE] switch_channel.c:1104 New Channel sofia/external/443307881011 at 185.35.229.30 :5060 [848843db-cad9-48a8-a4bd-2ac6b6b77dff] >> 2017-09-20 23:31:49.290658 [DEBUG] mod_sofia.c:4819 (sofia/external/443307881011 at 185.35.229.30 :5060) State Change CS_NEW -> CS_INIT >> 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:584 (sofia/external/443307881011 at 185.35.229.30 :5060) Running State Change CS_INIT (Cur 2 Tot 16431) >> 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:627 (sofia/external/443307881011 at 185.35.229.30 :5060) State INIT >> 2017-09-20 23:31:49.290658 [DEBUG] mod_sofia.c:90 sofia/external/443307881011 at 185.35.229.30 :5060 SOFIA INIT >> 2017-09-20 23:31:49.290658 [DEBUG] sofia_glue.c:1295 sofia/external/443307881011 at 185.35.229.30 :5060 sending invite version: 1.6.19 git febfb38 2017-09-14 23:14:02Z 64bit >> Local SDP: >> v=0 >> o=FreeSWITCH 1505929783 1505929784 IN IP4 185.35.229.6 >> s=FreeSWITCH >> c=IN IP4 185.35.229.6 >> t=0 0 >> m=audio 16926 RTP/AVP 8 101 13 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=rtpmap:13 CN/8000 >> a=ptime:20 >> a=sendrecv >> >> 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:40 sofia/external/443307881011 at 185.35.229.30 :5060 Standard INIT >> 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:48 (sofia/external/443307881011 at 185.35.229.30 :5060) State Change CS_INIT -> CS_ROUTING >> 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:627 (sofia/external/443307881011 at 185.35.229.30 :5060) State INIT going to sleep >> 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:584 (sofia/external/443307881011 at 185.35.229.30 :5060) Running State Change CS_ROUTING (Cur 2 Tot 16431) >> 2017-09-20 23:31:49.290658 [DEBUG] sofia.c:7084 Channel sofia/external/443307881011 at 185.35.229.30 :5060 entering state [calling][0] >> 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:643 (sofia/external/443307881011 at 185.35.229.30 :5060) State ROUTING >> 2017-09-20 23:31:49.290658 [DEBUG] mod_sofia.c:143 sofia/external/443307881011 at 185.35.229.30 :5060 SOFIA ROUTING >> 2017-09-20 23:31:49.290658 [DEBUG] switch_ivr_originate.c:67 (sofia/external/443307881011 at 185.35.229.30 :5060) State Change CS_ROUTING -> CS_CONSUME_MEDIA >> 2017-09-20 23:31:49.290658 [ERR] mod_xml_radius.c:930 Didn't match: 185.35.229.30:5060 == ^185\.35\.229\.30 >> 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:643 (sofia/external/443307881011 at 185.35.229.30 :5060) State ROUTING going to sleep >> 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:584 (sofia/external/443307881011 at 185.35.229.30 :5060) Running State Change CS_CONSUME_MEDIA (Cur 2 Tot 16431) >> 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:662 (sofia/external/443307881011 at 185.35.229.30 :5060) State CONSUME_MEDIA >> 2017-09-20 23:31:49.290658 [DEBUG] switch_core_state_machine.c:662 (sofia/external/443307881011 at 185.35.229.30 :5060) State CONSUME_MEDIA going to sleep >> 2017-09-20 23:31:49.350696 [DEBUG] sofia.c:6540 Redirect: Transfering to joehouse XML redirect >> 2017-09-20 23:31:49.350696 [DEBUG] switch_ivr.c:2165 (sofia/external/07966677711 at 185.8.92.3 ) State Change CS_EXECUTE -> CS_ROUTING >> 2017-09-20 23:31:49.350696 [NOTICE] switch_ivr.c:2172 Transfer sofia/external/07966677711 at 185.8.92.3 to XML[joehouse at redirect] >> 2017-09-20 23:31:49.350696 [NOTICE] sofia.c:6549 Hangup sofia/external/443307881011 at 185.35.229.30 :5060 [CS_CONSUME_MEDIA] [REDIRECTION_TO_NEW_DESTINATION] >> 2017-09-20 23:31:49.350696 [DEBUG] sofia.c:1453 Channel is already hungup. >> 2017-09-20 23:31:49.350696 [DEBUG] sofia.c:1453 Channel is already hungup. >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:584 (sofia/external/443307881011 at 185.35.229.30 :5060) Running State Change CS_HANGUP (Cur 2 Tot 16431) >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:850 (sofia/external/443307881011 at 185.35.229.30 :5060) Callstate Change DOWN -> HANGUP >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:852 (sofia/external/443307881011 at 185.35.229.30 :5060) State HANGUP >> 2017-09-20 23:31:49.350696 [DEBUG] mod_sofia.c:438 Channel sofia/external/443307881011 at 185.35.229.30 :5060 hanging up, cause: REDIRECTION_TO_NEW_DESTINATION >> 2017-09-20 23:31:49.350696 [DEBUG] mod_sofia.c:502 Sending CANCEL to sofia/external/443307881011 at 185.35.229.30 :5060 >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:60 sofia/external/443307881011 at 185.35.229.30 :5060 Standard HANGUP, cause: REDIRECTION_TO_NEW_DESTINATION >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:852 (sofia/external/443307881011 at 185.35.229.30 :5060) State HANGUP going to sleep >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:619 (sofia/external/443307881011 at 185.35.229.30 :5060) State Change CS_HANGUP -> CS_REPORTING >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:584 (sofia/external/443307881011 at 185.35.229.30 :5060) Running State Change CS_REPORTING (Cur 2 Tot 16431) >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:938 (sofia/external/443307881011 at 185.35.229.30 :5060) State REPORTING >> 2017-09-20 23:31:49.350696 [ERR] mod_xml_radius.c:930 Didn't match: 185.35.229.30:5060 == ^185\.35\.229\.30 >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:174 sofia/external/443307881011 at 185.35.229.30 :5060 Standard REPORTING, cause: REDIRECTION_TO_NEW_DESTINATION >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:938 (sofia/external/443307881011 at 185.35.229.30 :5060) State REPORTING going to sleep >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:610 (sofia/external/443307881011 at 185.35.229.30 :5060) State Change CS_REPORTING -> CS_DESTROY >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_session.c:1665 Session 16431 (sofia/external/443307881011 at 185.35.229.30 :5060) Locked, Waiting on external entities >> 2017-09-20 23:31:49.350696 [DEBUG] switch_ivr_originate.c:3837 Originate Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL] >> 2017-09-20 23:31:49.350696 [NOTICE] switch_core_session.c:1683 Session 16431 (sofia/external/443307881011 at 185.35.229.30 :5060) Ended >> 2017-09-20 23:31:49.350696 [NOTICE] switch_core_session.c:1687 Close Channel sofia/external/443307881011 at 185.35.229.30 :5060 [CS_DESTROY] >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:741 (sofia/external/443307881011 at 185.35.229.30 :5060) Running State Change CS_DESTROY (Cur 1 Tot 16431) >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:751 (sofia/external/443307881011 at 185.35.229.30 :5060) State DESTROY >> 2017-09-20 23:31:49.350696 [DEBUG] mod_sofia.c:343 sofia/external/443307881011 at 185.35.229.30 :5060 SOFIA DESTROY >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:181 sofia/external/443307881011 at 185.35.229.30 :5060 Standard DESTROY >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:751 (sofia/external/443307881011 at 185.35.229.30 :5060) State DESTROY going to sleep >> 2017-09-20 23:31:49.350696 [INFO] mod_dptools.c:3436 Originate Failed. Cause: ORIGINATOR_CANCEL >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:650 (sofia/external/07966677711 at 185.8.92.3 ) State EXECUTE going to sleep >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:584 (sofia/external/07966677711 at 185.8.92.3 ) Running State Change CS_ROUTING (Cur 1 Tot 16431) >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:643 (sofia/external/07966677711 at 185.8.92.3 ) State ROUTING >> 2017-09-20 23:31:49.350696 [DEBUG] mod_sofia.c:143 sofia/external/07966677711 at 185.8.92.3 SOFIA ROUTING >> 2017-09-20 23:31:49.350696 [ERR] mod_xml_radius.c:933 Result of true match: 185.35.229.6 == ^185\.35\.229\.30 >> 2017-09-20 23:31:49.350696 [ERR] mod_xml_radius.c:930 Didn't match: inbound == ^outbound >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:236 sofia/external/07966677711 at 185.8.92.3 Standard ROUTING >> 2017-09-20 23:31:49.350696 [INFO] mod_dialplan_xml.c:637 Processing 07966677711 <07966677711>->joehouse in context redirect >> Dialplan: sofia/external/07966677711 at 185.8.92.3 parsing [redirect->Redirect Calls] continue=false >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Regex (PASS) [Redirect Calls] ${sip_redirect_contact_0}(>;src_number=07966677711;q=1.00) =~ /^> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(process_cdr=true) >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action log(ALERT Local User) >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action set(accountcode=jerasoftoutbound) >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action log(ALERT joehouse) >> Dialplan: sofia/external/07966677711 at 185.8.92.3 Action bridge(user/joehouse at sip.biznetuk.com ) >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:286 (sofia/external/07966677711 at 185.8.92.3 ) State Change CS_ROUTING -> CS_EXECUTE >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:643 (sofia/external/07966677711 at 185.8.92.3 ) State ROUTING going to sleep >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:584 (sofia/external/07966677711 at 185.8.92.3 ) Running State Change CS_EXECUTE (Cur 1 Tot 16431) >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:650 (sofia/external/07966677711 at 185.8.92.3 ) State EXECUTE >> 2017-09-20 23:31:49.350696 [DEBUG] mod_sofia.c:198 sofia/external/07966677711 at 185.8.92.3 SOFIA EXECUTE >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:328 sofia/external/07966677711 at 185.8.92.3 Standard EXECUTE >> EXECUTE sofia/external/07966677711 at 185.8.92.3 set(process_cdr=true) >> 2017-09-20 23:31:49.350696 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [process_cdr]=[true] >> EXECUTE sofia/external/07966677711 at 185.8.92.3 log(ALERT Local User) >> 2017-09-20 23:31:49.350696 [ALERT] mod_dptools.c:1742 Local User >> EXECUTE sofia/external/07966677711 at 185.8.92.3 set(accountcode=jerasoftoutbound) >> 2017-09-20 23:31:49.350696 [DEBUG] mod_dptools.c:1548 SET sofia/external/07966677711 at 185.8.92.3 [accountcode]=[jerasoftoutbound] >> EXECUTE sofia/external/07966677711 at 185.8.92.3 log(ALERT joehouse) >> 2017-09-20 23:31:49.350696 [ALERT] mod_dptools.c:1742 joehouse >> EXECUTE sofia/external/07966677711 at 185.8.92.3 bridge(user/joehouse at sip.biznetuk.com ) >> 2017-09-20 23:31:49.350696 [DEBUG] switch_channel.c:1250 sofia/external/07966677711 at 185.8.92.3 EXPORTING[export_vars] [RFC2822_DATE]=[Wed, 20 Sep 2017 23:31:49 +0100] to event >> 2017-09-20 23:31:49.350696 [DEBUG] switch_channel.c:1250 sofia/external/07966677711 at 185.8.92.3 EXPORTING[export_vars] [h323-call-origin]=[originate] to event >> 2017-09-20 23:31:49.350696 [DEBUG] switch_channel.c:1250 sofia/external/07966677711 at 185.8.92.3 EXPORTING[export_vars] [credit_time]=[7199] to event >> 2017-09-20 23:31:49.350696 [DEBUG] switch_channel.c:1250 sofia/external/07966677711 at 185.8.92.3 EXPORTING[export_vars] [cisco_service_info]=[200] to event >> 2017-09-20 23:31:49.350696 [DEBUG] switch_channel.c:1250 sofia/external/07966677711 at 185.8.92.3 EXPORTING[export_vars] [sip_redirect_context]=[redirect] to event >> 2017-09-20 23:31:49.350696 [DEBUG] switch_ivr_originate.c:2142 Parsing global variables >> 2017-09-20 23:31:49.350696 [WARNING] mod_dptools.c:4184 Can't find user [joehouse at sip.biznetuk.com ] >> 2017-09-20 23:31:49.350696 [NOTICE] switch_ivr_originate.c:2851 Cannot create outgoing channel of type [user] cause: [SUBSCRIBER_ABSENT] >> 2017-09-20 23:31:49.350696 [DEBUG] switch_ivr_originate.c:3848 Originate Resulted in Error Cause: 20 [SUBSCRIBER_ABSENT] >> 2017-09-20 23:31:49.350696 [INFO] mod_dptools.c:3436 Originate Failed. Cause: SUBSCRIBER_ABSENT >> 2017-09-20 23:31:49.350696 [NOTICE] switch_core_state_machine.c:385 sofia/external/07966677711 at 185.8.92.3 has executed the last dialplan instruction, hanging up. >> 2017-09-20 23:31:49.350696 [NOTICE] switch_core_state_machine.c:387 Hangup sofia/external/07966677711 at 185.8.92.3 [CS_EXECUTE] [NORMAL_CLEARING] >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:650 (sofia/external/07966677711 at 185.8.92.3 ) State EXECUTE going to sleep >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:584 (sofia/external/07966677711 at 185.8.92.3 ) Running State Change CS_HANGUP (Cur 1 Tot 16431) >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:850 (sofia/external/07966677711 at 185.8.92.3 ) Callstate Change RINGING -> HANGUP >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:852 (sofia/external/07966677711 at 185.8.92.3 ) State HANGUP >> 2017-09-20 23:31:49.350696 [DEBUG] mod_sofia.c:438 Channel sofia/external/07966677711 at 185.8.92.3 hanging up, cause: NORMAL_CLEARING >> 2017-09-20 23:31:49.350696 [DEBUG] mod_sofia.c:577 Responding to INVITE with: 480 >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:60 sofia/external/07966677711 at 185.8.92.3 Standard HANGUP, cause: NORMAL_CLEARING >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:852 (sofia/external/07966677711 at 185.8.92.3 ) State HANGUP going to sleep >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:619 (sofia/external/07966677711 at 185.8.92.3 ) State Change CS_HANGUP -> CS_REPORTING >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:584 (sofia/external/07966677711 at 185.8.92.3 ) Running State Change CS_REPORTING (Cur 1 Tot 16431) >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:938 (sofia/external/07966677711 at 185.8.92.3 ) State REPORTING >> 2017-09-20 23:31:49.350696 [ERR] mod_xml_radius.c:933 Result of true match: 185.35.229.6 == ^185\.35\.229\.30 >> 2017-09-20 23:31:49.350696 [ERR] mod_xml_radius.c:930 Didn't match: inbound == ^outbound >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:174 sofia/external/07966677711 at 185.8.92.3 Standard REPORTING, cause: NORMAL_CLEARING >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:938 (sofia/external/07966677711 at 185.8.92.3 ) State REPORTING going to sleep >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:610 (sofia/external/07966677711 at 185.8.92.3 ) State Change CS_REPORTING -> CS_DESTROY >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_session.c:1665 Session 16430 (sofia/external/07966677711 at 185.8.92.3 ) Locked, Waiting on external entities >> 2017-09-20 23:31:49.350696 [NOTICE] switch_core_session.c:1683 Session 16430 (sofia/external/07966677711 at 185.8.92.3 ) Ended >> 2017-09-20 23:31:49.350696 [NOTICE] switch_core_session.c:1687 Close Channel sofia/external/07966677711 at 185.8.92.3 [CS_DESTROY] >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:741 (sofia/external/07966677711 at 185.8.92.3 ) Running State Change CS_DESTROY (Cur 0 Tot 16431) >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:751 (sofia/external/07966677711 at 185.8.92.3 ) State DESTROY >> 2017-09-20 23:31:49.350696 [DEBUG] mod_sofia.c:343 sofia/external/07966677711 at 185.8.92.3 SOFIA DESTROY >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:181 sofia/external/07966677711 at 185.8.92.3 Standard DESTROY >> 2017-09-20 23:31:49.350696 [DEBUG] switch_core_state_machine.c:751 (sofia/external/07966677711 at 185.8.92.3 ) State DESTROY going to sleep >> >> its saying there is no endpoint registered yet the output of show registrations shows there is >> >> freeswitch at BiznetukSBC1> show registrations >> reg_user,realm,token,url,expires,network_ip,network_port,network_proto,hostname,metadata,registration_uuid >> joehouse,sip.biznetuk.com ,1030250472 at 192_168_1_101,sofia/internal/sip:joehouse at 185.35.230.2:5382,1505947110,185.35.230.2,5382,udp,BiznetukSBC1,,60484000-aa2d-4b21-9d58-11e7d4155f1e >> >> 1 total. >> >> >> Any ideas? >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From ssinyagin at gmail.com Thu Sep 21 19:55:22 2017 From: ssinyagin at gmail.com (Stanislav Sinyagin) Date: Thu, 21 Sep 2017 21:55:22 +0200 Subject: [Freeswitch-users] Session recording performance In-Reply-To: References: Message-ID: that's right, loading mod_timerfd, and setting in the SIP profile has eliminated the problem, and now all outbound RTP has jitter below 1ms. On Thu, Sep 21, 2017 at 6:06 PM, Stanislav Sinyagin wrote: > it is two identical boxes, set up identically by myself :) > > I added mod_fdtimer, and set fdtimer as Sofia RTP timer, and the > results are impressively better. I'll do some additional tests before > a final commit. > > > > On Thu, Sep 21, 2017 at 5:28 PM, Brandon Armstead wrote: >> Your positive GSO / TSO / all are the same? Same Kernel? Same >> hardware? replica? What happens if you docker your test on 1 same >> physical host? >> >> On Thu, Sep 21, 2017 at 8:25 AM, Stanislav Sinyagin wrote: >>> but the other box is doing the same TX job, and it's doing fine. >>> I'm trying different timers now. >>> >>> >>> >>> On Thu, Sep 21, 2017 at 5:02 PM, Brandon Armstead wrote: >>>> Any leads maybe under ethtool -S [interface] , possible some kind of >>>> checksum / network "interference" , possibly the tx side is doing more >>>> work? >>>> >>>> https://www.kernel.org/doc/Documentation/networking/checksum-offloads.txt >>>> >>>> On Thu, Sep 21, 2017 at 7:49 AM, Stanislav Sinyagin wrote: >>>>> I actually mentioned it in the thread. Yes, after switching the roles >>>>> of the boxes, the problem switches too. It's only the call origination >>>>> box that is producing the jitter on outbound RTP. I also verified it >>>>> with packet capture on both sides. The RTP leaves the box with jitter >>>>> already. >>>>> >>>>> I'm still analyzing it, trying to find the reason for the jitter. >>>>> >>>>> >>>>> >>>>> On Thu, Sep 21, 2017 at 4:08 PM, Mundkowsky, Robert wrote: >>>>>> Just curious, have you tried switching the boxes and seeing if you get same situation? >>>>>> >>>>>> Robert >>>>>> >>>>>> -----Original Message----- >>>>>> From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Stanislav Sinyagin >>>>>> Sent: Thursday, September 21, 2017 8:31 AM >>>>>> To: FreeSWITCH Users Help >>>>>> Subject: Re: [Freeswitch-users] Session recording performance >>>>>> >>>>>> but the interesting thing is, that it's the device that originates the SIP call is producing a significantly higher jitter than the one receiving the call. >>>>>> >>>>>> The originating command looks like this here, with a SIP address as destination, and a number in local XML dialplan as transfer >>>>>> application: >>>>>> https://na01.safelinks.protection.outlook.com/?url=https%3A%2F%2Ffreeswitch.org%2Fconfluence%2Fdisplay%2FFREESWITCH%2FOriginate%2BExample&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636415940392006303&sdata=M5OEPvRrB9JkM5fodeKscJakpsvZn42LhZs2jQle9Ro%3D&reserved=0 >>>>>> >>>>>> >>>>>> >>>>>> On Thu, Sep 21, 2017 at 7:46 AM, Stanislav Sinyagin wrote: >>>>>>> with jitterbuffer, it now works as expected. So, the sending party was >>>>>>> generating extra jitter. >>>>>>> >>>>>>> I am playing the same audio file in both directions, and then compare >>>>>>> the recorded audio with the original by Sevana AQuA software. This >>>>>>> allows me to detect packet loss in a heterogeneous environment where >>>>>>> the voice goes through RTP proxies and TDM. >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> On Thu, Sep 21, 2017 at 3:43 AM, jungle Boogie wrote: >>>>>>>> What occurs if you send identical audio for the same duration? IOW, >>>>>>>> play the sound file for each call. >>>>>>>> >>>>>>>> >>>>>>>> _____________________________________________________________________ >>>>>>>> ____ Professional FreeSWITCH Consulting Services: >>>>>>>> consulting at freeswitch.org >>>>>>>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.f >>>>>>>> reeswitchsolutions.com&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846 >>>>>>>> ba37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0% >>>>>>>> 7C636415940392006303&sdata=nGh9In%2FvTyg37f7nJph6TXJFaz2nMMjZ0NXzIZr3 >>>>>>>> hWg%3D&reserved=0 >>>>>>>> >>>>>>>> Official FreeSWITCH Sites >>>>>>>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.f >>>>>>>> reeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba >>>>>>>> 852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C6364159 >>>>>>>> 40392006303&sdata=wFbEKkGrG49JjArYiD7D1mo6icJFqaxinnS3%2F5xq6hs%3D&re >>>>>>>> served=0 >>>>>>>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fconfl >>>>>>>> uence.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba >>>>>>>> 37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C >>>>>>>> 636415940392006303&sdata=ehMZuD9vMm325yuExibRBY6G9ysSDmvqoBQaXcHXFww% >>>>>>>> 3D&reserved=0 >>>>>>>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.c >>>>>>>> luecon.com&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852 >>>>>>>> 608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C6364159403 >>>>>>>> 92006303&sdata=J2p7ztQidG%2FLf4c6prqao3hhOnrmHHl2pxlwod3LNKM%3D&reser >>>>>>>> ved=0 >>>>>>>> >>>>>>>> FreeSWITCH-users mailing list >>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists >>>>>>>> .freeswitch.org%2Fmailman%2Flistinfo%2Ffreeswitch-users&data=02%7C01% >>>>>>>> 7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852608d500ed0775%7C0ba6e9b7 >>>>>>>> 60b34fae92f37e6ddd9e9b65%7C0%7C0%7C636415940392006303&sdata=kpqcuibvT >>>>>>>> vN3zMcS54w5CemHeLiKgj0OrMMTq1xiMUo%3D&reserved=0 >>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us >>>>>>>> ers >>>>>>>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.f >>>>>>>> reeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba >>>>>>>> 852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C6364159 >>>>>>>> 40392006303&sdata=wFbEKkGrG49JjArYiD7D1mo6icJFqaxinnS3%2F5xq6hs%3D&re >>>>>>>> served=0 >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitchsolutions.com&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636415940392006303&sdata=nGh9In%2FvTyg37f7nJph6TXJFaz2nMMjZ0NXzIZr3hWg%3D&reserved=0 >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636415940392006303&sdata=wFbEKkGrG49JjArYiD7D1mo6icJFqaxinnS3%2F5xq6hs%3D&reserved=0 >>>>>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fconfluence.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636415940392006303&sdata=ehMZuD9vMm325yuExibRBY6G9ysSDmvqoBQaXcHXFww%3D&reserved=0 >>>>>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.cluecon.com&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636415940392006303&sdata=J2p7ztQidG%2FLf4c6prqao3hhOnrmHHl2pxlwod3LNKM%3D&reserved=0 >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists.freeswitch.org%2Fmailman%2Flistinfo%2Ffreeswitch-users&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636415940392006303&sdata=kpqcuibvTvN3zMcS54w5CemHeLiKgj0OrMMTq1xiMUo%3D&reserved=0 >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7Ca687846ba37f40ba852608d500ed0775%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636415940392006303&sdata=wFbEKkGrG49JjArYiD7D1mo6icJFqaxinnS3%2F5xq6hs%3D&reserved=0 >>>>>> >>>>>> ________________________________ >>>>>> >>>>>> This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. >>>>>> >>>>>> >>>>>> Thank you for your compliance. >>>>>> >>>>>> ________________________________ >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> -- >>>> Sincerely, >>>> Brandon Armstead >>>> CTO / CRYY.com >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> >> -- >> Sincerely, >> Brandon Armstead >> CTO / CRYY.com >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org From gregor at infomedia.si Thu Sep 21 20:31:52 2017 From: gregor at infomedia.si (Gregor Nanger) Date: Thu, 21 Sep 2017 22:31:52 +0200 Subject: [Freeswitch-users] Bridge - user not registered - verto In-Reply-To: References: Message-ID: I am returning back to this thread for some more information. This is text from wiki about dialstring and bridge command: "When you use the user channel the user is pulled from your domain in your directory then the main domain is searched for a tag. If that is not found it searches in the for a tag. If a tag is located, it looks for the param "dial-string" as the originate string to use for that user." So, if I understand correctly, if dial-string is not found in domain, it searches for dial-string in user. And this happens for each bridge to user/xxxxx at xxxxxx. Does anyone know what happens in case of xml_curl, when domain and user directory is pulled from http request? Does FS make http request for each bridge to user/xxxxx at xxxxxx or does it cache internally? 2017-08-11 9:47 GMT+02:00 Michael Jerris : > your logs show you that's not what it's getting.... do some debugging for > what all is returned, check user and domain level, grep for sofia_contact > to see where it's set > > On Thu, Aug 10, 2017 at 7:23 PM Gregor Nanger wrote: > >> This is what I thought and played with dial-string, but it looks I am >> missing something. I set this dialstring: >> >> >> >> in User/Params section of XML (I am using xml_curl). So returning this >> for each verto extension at og in. Is this right place or something >> override this dial-string in user section? >> >> 2017-08-11 1:58 GMT+02:00 Michael Jerris : >> >>> it sounds like your user dial string calls functions to resolve sip and >>> verto endpoints and calls them both, sip returns not registered >>> >>> On Thu, Aug 10, 2017 at 4:51 PM Gregor Nanger >>> wrote: >>> >>>> Anyone has similar issue? >>>> >>>> If call is transfered to user/extension and this extension is verto >>>> than call is transfered, but also get USER NOT REGISTERED in log, because >>>> FS tries to transfer to SIP and Verto extension. Regardles of what I set in >>>> dial-string in user params. >>>> >>>> Best regards, Gregor >>>> >>>> 2017-07-12 19:40 GMT+02:00 Gregor Nanger : >>>> >>>>> When I make bridge to user/extension to verto user, I get user is not >>>>> registered. Call is transfered, but it looks like freeswitch tries to >>>>> transfer to verto and sip endpoint. And since sip endpoint is not >>>>> registered, I get User_not-registered. Everything works, but I get 2 >>>>> calllogs because of this. >>>>> >>>>> I also try to modify dial-string in user params, but same behavior >>>>> >>>>> Is this by design? >>>>> >>>>> Best regards, Gregor >>>>> -- >>>>> Gregor Nanger >>>>> >>>>> *CTO* >>>>> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 >>>>> • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia >>>>> • www.infomedia.si >>>>> >>>> >>>> >>>> >>>> -- >>>> Gregor Nanger >>>> >>>> *CTO* >>>> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 >>>> • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia >>>> • www.infomedia.si >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/ >>>> options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Gregor Nanger >> >> *CTO* >> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 >> • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia >> • www.infomedia.si >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Gregor Nanger *CTO* t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia • www.infomedia.si -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Thu Sep 21 20:58:04 2017 From: mike at jerris.com (Michael Jerris) Date: Thu, 21 Sep 2017 16:58:04 -0400 Subject: [Freeswitch-users] Bridge - user not registered - verto In-Reply-To: References: Message-ID: <38FE9B43-4BA3-4F17-A004-4D3B96C0B357@jerris.com> https://freeswitch.org/confluence/display/FREESWITCH/mod_xml_curl#mod_xml_curl-Caching > On Sep 21, 2017, at 4:31 PM, Gregor Nanger wrote: > > I am returning back to this thread for some more information. > > This is text from wiki about dialstring and bridge command: > > "When you use the user channel the user is pulled from your domain in your directory then the main domain is searched for a tag. If that is not found it searches in the for a tag. If a tag is located, it looks for the param "dial-string" as the originate string to use for that user." > > So, if I understand correctly, if dial-string is not found in domain, it searches for dial-string in user. And this happens for each bridge to user/xxxxx at xxxxxx. > > Does anyone know what happens in case of xml_curl, when domain and user directory is pulled from http request? Does FS make http request for each bridge to user/xxxxx at xxxxxx or does it cache internally? > > 2017-08-11 9:47 GMT+02:00 Michael Jerris >: > your logs show you that's not what it's getting.... do some debugging for what all is returned, check user and domain level, grep for sofia_contact to see where it's set > > On Thu, Aug 10, 2017 at 7:23 PM Gregor Nanger > wrote: > This is what I thought and played with dial-string, but it looks I am missing something. I set this dialstring: > > > > in User/Params section of XML (I am using xml_curl). So returning this for each verto extension at og in. Is this right place or something override this dial-string in user section? > > 2017-08-11 1:58 GMT+02:00 Michael Jerris >: > it sounds like your user dial string calls functions to resolve sip and verto endpoints and calls them both, sip returns not registered > > On Thu, Aug 10, 2017 at 4:51 PM Gregor Nanger > wrote: > Anyone has similar issue? > > If call is transfered to user/extension and this extension is verto than call is transfered, but also get USER NOT REGISTERED in log, because FS tries to transfer to SIP and Verto extension. Regardles of what I set in dial-string in user params. > > Best regards, Gregor > > 2017-07-12 19:40 GMT+02:00 Gregor Nanger >: > When I make bridge to user/extension to verto user, I get user is not registered. Call is transfered, but it looks like freeswitch tries to transfer to verto and sip endpoint. And since sip endpoint is not registered, I get User_not-registered. Everything works, but I get 2 calllogs because of this. > > I also try to modify dial-string in user params, but same behavior > > Is this by design? > > Best regards, Gregor > -- > Gregor Nanger > > CTO > t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 > • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia > • www.infomedia.si > > > -- > Gregor Nanger > > CTO > t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 > • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia > • www.infomedia.si _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > Gregor Nanger > > CTO > t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 > • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia > • www.infomedia.si _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > Gregor Nanger > > CTO > t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 > • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia > • www.infomedia.si _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From gregor at infomedia.si Fri Sep 22 04:08:25 2017 From: gregor at infomedia.si (Gregor Nanger) Date: Fri, 22 Sep 2017 06:08:25 +0200 Subject: [Freeswitch-users] Bridge - user not registered - verto In-Reply-To: <38FE9B43-4BA3-4F17-A004-4D3B96C0B357@jerris.com> References: <38FE9B43-4BA3-4F17-A004-4D3B96C0B357@jerris.com> Message-ID: Your hints are always usefull :-) Thank you. 2017-09-21 22:58 GMT+02:00 Michael Jerris : > https://freeswitch.org/confluence/display/FREESWITCH/ > mod_xml_curl#mod_xml_curl-Caching > > On Sep 21, 2017, at 4:31 PM, Gregor Nanger wrote: > > I am returning back to this thread for some more information. > > This is text from wiki about dialstring and bridge command: > > "When you use the user channel the user is pulled from your domain in > your directory then the main domain is searched for a tag. If that > is not found it searches in the for a tag. If a > tag is located, it looks for the param "dial-string" as the originate > string to use for that user." > > So, if I understand correctly, if dial-string is not found in domain, it > searches for dial-string in user. And this happens for each bridge to > user/xxxxx at xxxxxx. > > Does anyone know what happens in case of xml_curl, when domain and user > directory is pulled from http request? Does FS make http request for each > bridge to user/xxxxx at xxxxxx or does it cache internally? > > 2017-08-11 9:47 GMT+02:00 Michael Jerris : > >> your logs show you that's not what it's getting.... do some debugging for >> what all is returned, check user and domain level, grep for sofia_contact >> to see where it's set >> >> On Thu, Aug 10, 2017 at 7:23 PM Gregor Nanger >> wrote: >> >>> This is what I thought and played with dial-string, but it looks I am >>> missing something. I set this dialstring: >>> >>> >>> >>> in User/Params section of XML (I am using xml_curl). So returning this >>> for each verto extension at og in. Is this right place or something >>> override this dial-string in user section? >>> >>> 2017-08-11 1:58 GMT+02:00 Michael Jerris : >>> >>>> it sounds like your user dial string calls functions to resolve sip and >>>> verto endpoints and calls them both, sip returns not registered >>>> >>>> On Thu, Aug 10, 2017 at 4:51 PM Gregor Nanger >>>> wrote: >>>> >>>>> Anyone has similar issue? >>>>> >>>>> If call is transfered to user/extension and this extension is verto >>>>> than call is transfered, but also get USER NOT REGISTERED in log, because >>>>> FS tries to transfer to SIP and Verto extension. Regardles of what I set in >>>>> dial-string in user params. >>>>> >>>>> Best regards, Gregor >>>>> >>>>> 2017-07-12 19:40 GMT+02:00 Gregor Nanger : >>>>> >>>>>> When I make bridge to user/extension to verto user, I get user is not >>>>>> registered. Call is transfered, but it looks like freeswitch tries to >>>>>> transfer to verto and sip endpoint. And since sip endpoint is not >>>>>> registered, I get User_not-registered. Everything works, but I get 2 >>>>>> calllogs because of this. >>>>>> >>>>>> I also try to modify dial-string in user params, but same behavior >>>>>> >>>>>> Is this by design? >>>>>> >>>>>> Best regards, Gregor >>>>>> -- >>>>>> Gregor Nanger >>>>>> >>>>>> *CTO* >>>>>> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 >>>>>> • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia >>>>>> • www.infomedia.si >>>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> Gregor Nanger >>>>> >>>>> *CTO* >>>>> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 >>>>> • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia >>>>> • www.infomedia.si >>>>> ____________________________________________________________ >>>>> _____________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ >>>>> freeswitch-users >>>>> http://www.freeswitch.org >>>> >>>> >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ >>>> freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> -- >>> Gregor Nanger >>> >>> *CTO* >>> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 >>> • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia >>> • www.infomedia.si >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Gregor Nanger > > *CTO* > t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 > • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia > • www.infomedia.si > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Gregor Nanger *CTO* t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia • www.infomedia.si -------------- next part -------------- An HTML attachment was scrubbed... URL: From tculjaga at gmail.com Fri Sep 22 06:12:06 2017 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Fri, 22 Sep 2017 08:12:06 +0200 Subject: [Freeswitch-users] Bridge - user not registered - verto In-Reply-To: <38FE9B43-4BA3-4F17-A004-4D3B96C0B357@jerris.com> References: <38FE9B43-4BA3-4F17-A004-4D3B96C0B357@jerris.com> Message-ID: i set caching to 10 seconds and found this value optimal. also, since you know already the user in question is a verto user, you can change your dial-string you use in originate from user/xxxx at domain to something like this: bgapi expand originate ${verto_contact 1001 at 192.168.5.150} &echo() or bgapi originate verto.rtc/1001 at 192.168.5.150 &echo() T. On 21 September 2017 at 22:58, Michael Jerris wrote: > https://freeswitch.org/confluence/display/FREESWITCH/ > mod_xml_curl#mod_xml_curl-Caching > > On Sep 21, 2017, at 4:31 PM, Gregor Nanger wrote: > > I am returning back to this thread for some more information. > > This is text from wiki about dialstring and bridge command: > > "When you use the user channel the user is pulled from your domain in > your directory then the main domain is searched for a tag. If that > is not found it searches in the for a tag. If a > tag is located, it looks for the param "dial-string" as the originate > string to use for that user." > > So, if I understand correctly, if dial-string is not found in domain, it > searches for dial-string in user. And this happens for each bridge to > user/xxxxx at xxxxxx. > > Does anyone know what happens in case of xml_curl, when domain and user > directory is pulled from http request? Does FS make http request for each > bridge to user/xxxxx at xxxxxx or does it cache internally? > > 2017-08-11 9:47 GMT+02:00 Michael Jerris : > >> your logs show you that's not what it's getting.... do some debugging for >> what all is returned, check user and domain level, grep for sofia_contact >> to see where it's set >> >> On Thu, Aug 10, 2017 at 7:23 PM Gregor Nanger >> wrote: >> >>> This is what I thought and played with dial-string, but it looks I am >>> missing something. I set this dialstring: >>> >>> >>> >>> in User/Params section of XML (I am using xml_curl). So returning this >>> for each verto extension at og in. Is this right place or something >>> override this dial-string in user section? >>> >>> 2017-08-11 1:58 GMT+02:00 Michael Jerris : >>> >>>> it sounds like your user dial string calls functions to resolve sip and >>>> verto endpoints and calls them both, sip returns not registered >>>> >>>> On Thu, Aug 10, 2017 at 4:51 PM Gregor Nanger >>>> wrote: >>>> >>>>> Anyone has similar issue? >>>>> >>>>> If call is transfered to user/extension and this extension is verto >>>>> than call is transfered, but also get USER NOT REGISTERED in log, because >>>>> FS tries to transfer to SIP and Verto extension. Regardles of what I set in >>>>> dial-string in user params. >>>>> >>>>> Best regards, Gregor >>>>> >>>>> 2017-07-12 19:40 GMT+02:00 Gregor Nanger : >>>>> >>>>>> When I make bridge to user/extension to verto user, I get user is not >>>>>> registered. Call is transfered, but it looks like freeswitch tries to >>>>>> transfer to verto and sip endpoint. And since sip endpoint is not >>>>>> registered, I get User_not-registered. Everything works, but I get 2 >>>>>> calllogs because of this. >>>>>> >>>>>> I also try to modify dial-string in user params, but same behavior >>>>>> >>>>>> Is this by design? >>>>>> >>>>>> Best regards, Gregor >>>>>> -- >>>>>> Gregor Nanger >>>>>> >>>>>> *CTO* >>>>>> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 >>>>>> • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia >>>>>> • www.infomedia.si >>>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> Gregor Nanger >>>>> >>>>> *CTO* >>>>> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 >>>>> • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia >>>>> • www.infomedia.si >>>>> ____________________________________________________________ >>>>> _____________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ >>>>> freeswitch-users >>>>> http://www.freeswitch.org >>>> >>>> >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ >>>> freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> -- >>> Gregor Nanger >>> >>> *CTO* >>> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 >>> • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia >>> • www.infomedia.si >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Gregor Nanger > > *CTO* > t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 > • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia > • www.infomedia.si > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From gregor at infomedia.si Fri Sep 22 06:20:54 2017 From: gregor at infomedia.si (Gregor Nanger) Date: Fri, 22 Sep 2017 08:20:54 +0200 Subject: [Freeswitch-users] Bridge - user not registered - verto In-Reply-To: References: <38FE9B43-4BA3-4F17-A004-4D3B96C0B357@jerris.com> Message-ID: Since I feed dialplan from my logic, I can set bridge command with dialstring like bridge ${verto_contact 1001 at 192.168.5.150} | ${sofia_contact 1002 at 192.168.5.150 <1001 at 192.168.5.150>}? This way I will avoid xml_curl dial plan hit? 2017-09-22 8:12 GMT+02:00 Tihomir Culjaga : > i set caching to 10 seconds and found this value optimal. > > also, since you know already the user in question is a verto user, you can > change your dial-string you use in originate from user/xxxx at domain to > something like this: > > bgapi expand originate ${verto_contact 1001 at 192.168.5.150} &echo() > > or > > bgapi originate verto.rtc/1001 at 192.168.5.150 &echo() > > > > T. > > > On 21 September 2017 at 22:58, Michael Jerris wrote: > >> https://freeswitch.org/confluence/display/FREESWITCH/mod_ >> xml_curl#mod_xml_curl-Caching >> >> On Sep 21, 2017, at 4:31 PM, Gregor Nanger wrote: >> >> I am returning back to this thread for some more information. >> >> This is text from wiki about dialstring and bridge command: >> >> "When you use the user channel the user is pulled from your domain in >> your directory then the main domain is searched for a tag. If that >> is not found it searches in the for a tag. If a >> tag is located, it looks for the param "dial-string" as the originate >> string to use for that user." >> >> So, if I understand correctly, if dial-string is not found in domain, it >> searches for dial-string in user. And this happens for each bridge to >> user/xxxxx at xxxxxx. >> >> Does anyone know what happens in case of xml_curl, when domain and user >> directory is pulled from http request? Does FS make http request for each >> bridge to user/xxxxx at xxxxxx or does it cache internally? >> >> 2017-08-11 9:47 GMT+02:00 Michael Jerris : >> >>> your logs show you that's not what it's getting.... do some debugging >>> for what all is returned, check user and domain level, grep for >>> sofia_contact to see where it's set >>> >>> On Thu, Aug 10, 2017 at 7:23 PM Gregor Nanger >>> wrote: >>> >>>> This is what I thought and played with dial-string, but it looks I am >>>> missing something. I set this dialstring: >>>> >>>> >>>> >>>> in User/Params section of XML (I am using xml_curl). So returning this >>>> for each verto extension at og in. Is this right place or something >>>> override this dial-string in user section? >>>> >>>> 2017-08-11 1:58 GMT+02:00 Michael Jerris : >>>> >>>>> it sounds like your user dial string calls functions to resolve sip >>>>> and verto endpoints and calls them both, sip returns not registered >>>>> >>>>> On Thu, Aug 10, 2017 at 4:51 PM Gregor Nanger >>>>> wrote: >>>>> >>>>>> Anyone has similar issue? >>>>>> >>>>>> If call is transfered to user/extension and this extension is verto >>>>>> than call is transfered, but also get USER NOT REGISTERED in log, because >>>>>> FS tries to transfer to SIP and Verto extension. Regardles of what I set in >>>>>> dial-string in user params. >>>>>> >>>>>> Best regards, Gregor >>>>>> >>>>>> 2017-07-12 19:40 GMT+02:00 Gregor Nanger : >>>>>> >>>>>>> When I make bridge to user/extension to verto user, I get user is >>>>>>> not registered. Call is transfered, but it looks like freeswitch tries to >>>>>>> transfer to verto and sip endpoint. And since sip endpoint is not >>>>>>> registered, I get User_not-registered. Everything works, but I get 2 >>>>>>> calllogs because of this. >>>>>>> >>>>>>> I also try to modify dial-string in user params, but same behavior >>>>>>> >>>>>>> Is this by design? >>>>>>> >>>>>>> Best regards, Gregor >>>>>>> -- >>>>>>> Gregor Nanger >>>>>>> >>>>>>> *CTO* >>>>>>> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 >>>>>>> • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia >>>>>>> • www.infomedia.si >>>>>>> >>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> Gregor Nanger >>>>>> >>>>>> *CTO* >>>>>> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 >>>>>> • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia >>>>>> • www.infomedia.si >>>>>> ____________________________________________________________ >>>>>> _____________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>>>> switch-users >>>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> ____________________________________________________________ >>>>> _____________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>>> switch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> -- >>>> Gregor Nanger >>>> >>>> *CTO* >>>> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 >>>> • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia >>>> • www.infomedia.si >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>> switch-users >>>> http://www.freeswitch.org >>> >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Gregor Nanger >> >> *CTO* >> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 >> • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia >> • www.infomedia.si >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Gregor Nanger *CTO* t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia • www.infomedia.si -------------- next part -------------- An HTML attachment was scrubbed... URL: From ssinyagin at gmail.com Fri Sep 22 07:12:52 2017 From: ssinyagin at gmail.com (Stanislav Sinyagin) Date: Fri, 22 Sep 2017 09:12:52 +0200 Subject: [Freeswitch-users] debian-unstable on Stretch In-Reply-To: References: Message-ID: here's one more incentive to move to Stretch: The DHCP client is listening on a random UDP port from a range that overlaps with RTP: https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=712503 I just bumped into it while making bulk performance tests. I checked, in Stretch the dhclient is only listening on port 68, as expected. On Fri, Sep 15, 2017 at 2:23 PM, Ken Rice wrote: > there are no available packages for stretch at this time. > > Sent from my iPhone > >> On Sep 15, 2017, at 05:01, Stanislav Sinyagin wrote: >> >> I'm trying to install the unstable FreeSWITCH on debian Stretch, and I >> get the following error: >> >> # apt-get update >> Hit:1 http://security.debian.org stretch/updates InRelease >> Get:3 http://files.freeswitch.org/repo/deb/debian-unstable stretch >> InRelease [4,595 B] >> Ign:2 http://cdn-fastly.deb.debian.org/debian stretch InRelease >> Hit:4 http://cdn-fastly.deb.debian.org/debian stretch Release >> Ign:3 http://files.freeswitch.org/repo/deb/debian-unstable stretch InRelease >> Fetched 4,595 B in 1s (3,266 B/s) >> Reading package lists... Done >> W: GPG error: http://files.freeswitch.org/repo/deb/debian-unstable >> stretch InRelease: The following signatures were invalid: >> 20B06EE621AB150D40F6079FD76EDC7725E010CF >> W: The repository >> 'http://files.freeswitch.org/repo/deb/debian-unstable stretch >> InRelease' is not signed. >> N: Data from such a repository can't be authenticated and is therefore >> potentially dangerous to use. >> N: See apt-secure(8) manpage for repository creation and user >> configuration details. >> >> I imported all public keys that I could find, but it didn't help. >> Which key was used for signing this repo? >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From covici at ccs.covici.com Fri Sep 22 08:19:10 2017 From: covici at ccs.covici.com (John Covici) Date: Fri, 22 Sep 2017 04:19:10 -0400 Subject: [Freeswitch-users] strange problem with mod_portaudio Message-ID: Hi. I am using 1.6.19 and using mod_portaudio I cannot dial 9888 (the conference). It says no route and gives up. Now what is strange is that mod_portaudio can dial all other normal numbers and my other phones can dial 9888, so what is confusing mod_portaudio? The configs are the default except I eliminated other streams which were in there anc cause errors on module load. Thanks in advance for any suggestions. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici at ccs.covici.com From tculjaga at gmail.com Fri Sep 22 08:39:42 2017 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Fri, 22 Sep 2017 10:39:42 +0200 Subject: [Freeswitch-users] Bridge - user not registered - verto In-Reply-To: References: <38FE9B43-4BA3-4F17-A004-4D3B96C0B357@jerris.com> Message-ID: yes this way you will avoid 1 more xml_curl resolve on originate. On 22 September 2017 at 08:20, Gregor Nanger wrote: > Since I feed dialplan from my logic, I can set bridge command with > dialstring like bridge ${verto_contact 1001 at 192.168.5.150} | > ${sofia_contact 1002 at 192.168.5.150 <1001 at 192.168.5.150>}? This way I will > avoid xml_curl dial plan hit? > > 2017-09-22 8:12 GMT+02:00 Tihomir Culjaga : > >> i set caching to 10 seconds and found this value optimal. >> >> also, since you know already the user in question is a verto user, you >> can change your dial-string you use in originate from user/xxxx at domain >> to something like this: >> >> bgapi expand originate ${verto_contact 1001 at 192.168.5.150} &echo() >> >> or >> >> bgapi originate verto.rtc/1001 at 192.168.5.150 &echo() >> >> >> >> T. >> >> >> On 21 September 2017 at 22:58, Michael Jerris wrote: >> >>> https://freeswitch.org/confluence/display/FREESWITCH/mod_xml >>> _curl#mod_xml_curl-Caching >>> >>> On Sep 21, 2017, at 4:31 PM, Gregor Nanger wrote: >>> >>> I am returning back to this thread for some more information. >>> >>> This is text from wiki about dialstring and bridge command: >>> >>> "When you use the user channel the user is pulled from your domain in >>> your directory then the main domain is searched for a tag. If that >>> is not found it searches in the for a tag. If a >>> tag is located, it looks for the param "dial-string" as the originate >>> string to use for that user." >>> >>> So, if I understand correctly, if dial-string is not found in domain, it >>> searches for dial-string in user. And this happens for each bridge to >>> user/xxxxx at xxxxxx. >>> >>> Does anyone know what happens in case of xml_curl, when domain and user >>> directory is pulled from http request? Does FS make http request for each >>> bridge to user/xxxxx at xxxxxx or does it cache internally? >>> >>> 2017-08-11 9:47 GMT+02:00 Michael Jerris : >>> >>>> your logs show you that's not what it's getting.... do some debugging >>>> for what all is returned, check user and domain level, grep for >>>> sofia_contact to see where it's set >>>> >>>> On Thu, Aug 10, 2017 at 7:23 PM Gregor Nanger >>>> wrote: >>>> >>>>> This is what I thought and played with dial-string, but it looks I am >>>>> missing something. I set this dialstring: >>>>> >>>>> >>>>> >>>>> in User/Params section of XML (I am using xml_curl). So returning this >>>>> for each verto extension at og in. Is this right place or something >>>>> override this dial-string in user section? >>>>> >>>>> 2017-08-11 1:58 GMT+02:00 Michael Jerris : >>>>> >>>>>> it sounds like your user dial string calls functions to resolve sip >>>>>> and verto endpoints and calls them both, sip returns not registered >>>>>> >>>>>> On Thu, Aug 10, 2017 at 4:51 PM Gregor Nanger >>>>>> wrote: >>>>>> >>>>>>> Anyone has similar issue? >>>>>>> >>>>>>> If call is transfered to user/extension and this extension is verto >>>>>>> than call is transfered, but also get USER NOT REGISTERED in log, because >>>>>>> FS tries to transfer to SIP and Verto extension. Regardles of what I set in >>>>>>> dial-string in user params. >>>>>>> >>>>>>> Best regards, Gregor >>>>>>> >>>>>>> 2017-07-12 19:40 GMT+02:00 Gregor Nanger : >>>>>>> >>>>>>>> When I make bridge to user/extension to verto user, I get user is >>>>>>>> not registered. Call is transfered, but it looks like freeswitch tries to >>>>>>>> transfer to verto and sip endpoint. And since sip endpoint is not >>>>>>>> registered, I get User_not-registered. Everything works, but I get 2 >>>>>>>> calllogs because of this. >>>>>>>> >>>>>>>> I also try to modify dial-string in user params, but same behavior >>>>>>>> >>>>>>>> Is this by design? >>>>>>>> >>>>>>>> Best regards, Gregor >>>>>>>> -- >>>>>>>> Gregor Nanger >>>>>>>> >>>>>>>> *CTO* >>>>>>>> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 >>>>>>>> • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia >>>>>>>> • www.infomedia.si >>>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> Gregor Nanger >>>>>>> >>>>>>> *CTO* >>>>>>> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 >>>>>>> • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia >>>>>>> • www.infomedia.si >>>>>>> ____________________________________________________________ >>>>>>> _____________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://confluence.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>>>>> switch-users >>>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> ____________________________________________________________ >>>>>> _____________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>>>> switch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> Gregor Nanger >>>>> >>>>> *CTO* >>>>> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 >>>>> • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia >>>>> • www.infomedia.si >>>>> ____________________________________________________________ >>>>> _____________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>>> switch-users >>>>> http://www.freeswitch.org >>>> >>>> >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>> switch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> -- >>> Gregor Nanger >>> >>> *CTO* >>> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 >>> • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia >>> • www.infomedia.si >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Gregor Nanger > > *CTO* > t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 > • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia > • www.infomedia.si > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From gregor at infomedia.si Fri Sep 22 10:14:46 2017 From: gregor at infomedia.si (Gregor Nanger) Date: Fri, 22 Sep 2017 12:14:46 +0200 Subject: [Freeswitch-users] Bridge - user not registered - verto In-Reply-To: References: <38FE9B43-4BA3-4F17-A004-4D3B96C0B357@jerris.com> Message-ID: ​Tihomir, thanks +1 :-)​ 2017-09-22 10:39 GMT+02:00 Tihomir Culjaga : > yes this way you will avoid 1 more xml_curl resolve on originate. > > On 22 September 2017 at 08:20, Gregor Nanger wrote: > >> Since I feed dialplan from my logic, I can set bridge command with >> dialstring like bridge ${verto_contact 1001 at 192.168.5.150} | >> ${sofia_contact 1002 at 192.168.5.150 <1001 at 192.168.5.150>}? This way I >> will avoid xml_curl dial plan hit? >> >> 2017-09-22 8:12 GMT+02:00 Tihomir Culjaga : >> >>> i set caching to 10 seconds and found this value optimal. >>> >>> also, since you know already the user in question is a verto user, you >>> can change your dial-string you use in originate from user/xxxx at domain >>> to something like this: >>> >>> bgapi expand originate ${verto_contact 1001 at 192.168.5.150} &echo() >>> >>> or >>> >>> bgapi originate verto.rtc/1001 at 192.168.5.150 &echo() >>> >>> >>> >>> T. >>> >>> >>> On 21 September 2017 at 22:58, Michael Jerris wrote: >>> >>>> https://freeswitch.org/confluence/display/FREESWITCH/mod_xml >>>> _curl#mod_xml_curl-Caching >>>> >>>> On Sep 21, 2017, at 4:31 PM, Gregor Nanger wrote: >>>> >>>> I am returning back to this thread for some more information. >>>> >>>> This is text from wiki about dialstring and bridge command: >>>> >>>> "When you use the user channel the user is pulled from your domain in >>>> your directory then the main domain is searched for a tag. If that >>>> is not found it searches in the for a tag. If a >>>> tag is located, it looks for the param "dial-string" as the originate >>>> string to use for that user." >>>> >>>> So, if I understand correctly, if dial-string is not found in domain, >>>> it searches for dial-string in user. And this happens for each bridge to >>>> user/xxxxx at xxxxxx. >>>> >>>> Does anyone know what happens in case of xml_curl, when domain and user >>>> directory is pulled from http request? Does FS make http request for each >>>> bridge to user/xxxxx at xxxxxx or does it cache internally? >>>> >>>> 2017-08-11 9:47 GMT+02:00 Michael Jerris : >>>> >>>>> your logs show you that's not what it's getting.... do some debugging >>>>> for what all is returned, check user and domain level, grep for >>>>> sofia_contact to see where it's set >>>>> >>>>> On Thu, Aug 10, 2017 at 7:23 PM Gregor Nanger >>>>> wrote: >>>>> >>>>>> This is what I thought and played with dial-string, but it looks I am >>>>>> missing something. I set this dialstring: >>>>>> >>>>>> >>>>>> >>>>>> in User/Params section of XML (I am using xml_curl). So returning >>>>>> this for each verto extension at og in. Is this right place or something >>>>>> override this dial-string in user section? >>>>>> >>>>>> 2017-08-11 1:58 GMT+02:00 Michael Jerris : >>>>>> >>>>>>> it sounds like your user dial string calls functions to resolve sip >>>>>>> and verto endpoints and calls them both, sip returns not registered >>>>>>> >>>>>>> On Thu, Aug 10, 2017 at 4:51 PM Gregor Nanger >>>>>>> wrote: >>>>>>> >>>>>>>> Anyone has similar issue? >>>>>>>> >>>>>>>> If call is transfered to user/extension and this extension is verto >>>>>>>> than call is transfered, but also get USER NOT REGISTERED in log, because >>>>>>>> FS tries to transfer to SIP and Verto extension. Regardles of what I set in >>>>>>>> dial-string in user params. >>>>>>>> >>>>>>>> Best regards, Gregor >>>>>>>> >>>>>>>> 2017-07-12 19:40 GMT+02:00 Gregor Nanger : >>>>>>>> >>>>>>>>> When I make bridge to user/extension to verto user, I get user is >>>>>>>>> not registered. Call is transfered, but it looks like freeswitch tries to >>>>>>>>> transfer to verto and sip endpoint. And since sip endpoint is not >>>>>>>>> registered, I get User_not-registered. Everything works, but I get 2 >>>>>>>>> calllogs because of this. >>>>>>>>> >>>>>>>>> I also try to modify dial-string in user params, but same behavior >>>>>>>>> >>>>>>>>> Is this by design? >>>>>>>>> >>>>>>>>> Best regards, Gregor >>>>>>>>> -- >>>>>>>>> Gregor Nanger >>>>>>>>> >>>>>>>>> *CTO* >>>>>>>>> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 >>>>>>>>> • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia >>>>>>>>> • www.infomedia.si >>>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> -- >>>>>>>> Gregor Nanger >>>>>>>> >>>>>>>> *CTO* >>>>>>>> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 >>>>>>>> • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia >>>>>>>> • www.infomedia.si >>>>>>>> ____________________________________________________________ >>>>>>>> _____________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting at freeswitch.org >>>>>>>> http://www.freeswitchsolutions.com >>>>>>>> >>>>>>>> Official FreeSWITCH Sites >>>>>>>> http://www.freeswitch.org >>>>>>>> http://confluence.freeswitch.org >>>>>>>> http://www.cluecon.com >>>>>>>> >>>>>>>> FreeSWITCH-users mailing list >>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>>>>>> switch-users >>>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>>> ____________________________________________________________ >>>>>>> _____________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://confluence.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>>>>> switch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> Gregor Nanger >>>>>> >>>>>> *CTO* >>>>>> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 >>>>>> • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia >>>>>> • www.infomedia.si >>>>>> ____________________________________________________________ >>>>>> _____________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>>>> switch-users >>>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> ____________________________________________________________ >>>>> _____________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>>> switch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> -- >>>> Gregor Nanger >>>> >>>> *CTO* >>>> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 >>>> • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia >>>> • www.infomedia.si >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>> switch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>> switch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Gregor Nanger >> >> *CTO* >> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 >> • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia >> • www.infomedia.si >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Gregor Nanger *CTO* t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia • www.infomedia.si -------------- next part -------------- An HTML attachment was scrubbed... URL: From me at anatoli.ws Fri Sep 22 00:30:40 2017 From: me at anatoli.ws (Anatoli) Date: Thu, 21 Sep 2017 21:30:40 -0300 Subject: [Freeswitch-users] Session recording performance In-Reply-To: References: Message-ID: <48277fcb-6516-3720-858e-b3b8b0b10725@anatoli.ws> Isn't timerfd the default under-the-hood implementation of the "soft" (the default) timer in FS for some years now? In other words, it's quite surprising this param had any effect at all. Are you sure it did it? What are your kernel, glibc (ldd --version) and FS versions? *From:* Stanislav Sinyagin *Sent:* Thursday, September 21, 2017 16:55 *To:* Freeswitch Users Help *Subject:* Re: [Freeswitch-users] Session recording performance that's right, loading mod_timerfd, and setting in the SIP profile has eliminated the problem, and now all outbound RTP has jitter below 1ms. -------------- next part -------------- An HTML attachment was scrubbed... URL: From infos at madovsky.org Fri Sep 22 16:15:41 2017 From: infos at madovsky.org (Madovsky) Date: Fri, 22 Sep 2017 09:15:41 -0700 Subject: [Freeswitch-users] mod_rtmp codecs Message-ID: Hello, is mod_rtmp accepts PCMU as codec from the flex client? thanks Franck From brian at freeswitch.com Fri Sep 22 17:04:13 2017 From: brian at freeswitch.com (Brian West) Date: Fri, 22 Sep 2017 12:04:13 -0500 Subject: [Freeswitch-users] mod_rtmp codecs In-Reply-To: References: Message-ID: email consulting at freeswitch.com /b On Fri, Sep 22, 2017 at 11:15 AM, Madovsky wrote: > Hello, > > is mod_rtmp accepts PCMU as codec from the flex client? > > thanks > > > Franck > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- *Brian West* brian at freeswitch.com *Twitter: @FreeSWITCH , @cluecon* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) -------------- next part -------------- An HTML attachment was scrubbed... URL: From sat at calgaryit.com Fri Sep 22 19:49:01 2017 From: sat at calgaryit.com (George) Date: Fri, 22 Sep 2017 13:49:01 -0600 (MDT) Subject: [Freeswitch-users] CISCO SPA525G2 BLA issue Message-ID: <592060049.102807.1506109741077.JavaMail.zimbra@calgaryit.com> I am having inconsistent BLA issue on these phones, sometimes is works and sometimes not, this is what I have one of the lines set to: fnc=blf+sd+cp;sub=100 at 192.168.0.99;ext=100 at 192.168.0.99 This is my FS version: FreeSWITCH Version 1.9.0-582-c0a02f5~64bit the issue is on inbound calls, all phones are part of a simo-ring group, one person answers, the other phones sometimes show they are on the phone and sometimes not. Thank You, George From ssinyagin at gmail.com Fri Sep 22 20:09:05 2017 From: ssinyagin at gmail.com (Stanislav Sinyagin) Date: Fri, 22 Sep 2017 22:09:05 +0200 Subject: [Freeswitch-users] Session recording performance In-Reply-To: <48277fcb-6516-3720-858e-b3b8b0b10725@anatoli.ws> References: <48277fcb-6516-3720-858e-b3b8b0b10725@anatoli.ws> Message-ID: I thought so too, but I detected a high jitter in outbound RTP (over 10ms) , and this parameter has resolved it. The installation is as follows: The physical host is running under Debian Stretch, kernel version 4.9.0-3-amd64. The VoIP probe is isolated into an LXC container, and the container is running Debian Jessie. The container is not limited in any resources, and FreeSWITCH receives its real-time priority as it should. I tested the priority explicitly by running other processes in normal priority, and their load has no effect on FreeSWITCH performance. On the physical host: ldd (Debian GLIBC 2.24-11+deb9u1) 2.24 Inside the container: ldd (Debian GLIBC 2.19-18+deb8u10) 2.19 freeswitch at fsnic2> status UP 0 years, 0 days, 5 hours, 25 minutes, 25 seconds, 101 milliseconds, 11 microseconds FreeSWITCH (Version 1.6.19 -36-7a77e0b 64bit) is ready 12924 session(s) since startup 16 session(s) - peak 17, last 5min 17 1 session(s) per Sec out of max 30, peak 1, last 5min 1 1000 session(s) max min idle cpu 0.00/89.70 Current Stack Size/Max 240K/8192K So, the only difference from a standard baremetal installation, is a newer kernel version. But all other libraries are from Jessie. I will eventually make additional tests with baremetal servers without LXC, but I don't think it's the source of the issue. On Fri, Sep 22, 2017 at 2:30 AM, Anatoli wrote: > Isn't timerfd the default under-the-hood implementation of the "soft" (the > default) timer in FS for some years now? In other words, it's quite > surprising this param had any effect at all. Are you sure it did it? What > are your kernel, glibc (ldd --version) and FS versions? > > From: Stanislav Sinyagin > Sent: Thursday, September 21, 2017 16:55 > To: Freeswitch Users Help > Subject: Re: [Freeswitch-users] Session recording performance > > that's right, loading mod_timerfd, and setting in the SIP profile > > > has eliminated the problem, and now all outbound RTP has jitter below 1ms. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From bipin at xbipin.com Fri Sep 22 21:26:52 2017 From: bipin at xbipin.com (Bipin Patel) Date: Sat, 23 Sep 2017 01:26:52 +0400 Subject: [Freeswitch-users] tls srtp calls between users white noise Message-ID: <8a5988df-e823-05fb-562e-5bb0620d068d@xbipin.com> hi, i got tls and srtp working between client and FS, client registers to FS using TLS and SRTP and if that call is forwarded to gateway in SIP and RTP everything works fine. Problem arises when 2 clients using zoiper register to FS using TLS and SRTP and any calls routed between them the calls goes through and the phone rings but on answering there is white noise constantly, i checked in FS and it does send the call using TLS and SRTP and key exchange and codec negotiation all is fine using g711u but i cant seem to get rid of this white noise almost making it impossible to each party to hear the other. any idea what could be wrong my dialplan looks like this                                                                                                           my porfile looks like this                                     -- Regards, Bipin ------------------------------------------------------------------------ -------------- next part -------------- An HTML attachment was scrubbed... URL: From michael_cooley at comcast.com Fri Sep 22 21:27:23 2017 From: michael_cooley at comcast.com (mcooley) Date: Fri, 22 Sep 2017 14:27:23 -0700 (MST) Subject: [Freeswitch-users] Modifying RURI for outbound calls Message-ID: <1506115643453-0.post@n2.nabble.com> Hi, How do I modify the RURI for outbound calls? I'd like to add the port, protocol and user=phone So, this: INVITE sip:3035551212 at 10.10.10.10 SIP/2.0 to this: INVITE sip:3035551212 at 10.10.10.10:5060;user=phone;trasport=udp SIP/2.0 thanks, mike -- Sent from: http://freeswitch-users.2379917.n2.nabble.com/ From bipin at xbipin.com Sat Sep 23 08:54:36 2017 From: bipin at xbipin.com (Bipin Patel) Date: Sat, 23 Sep 2017 12:54:36 +0400 Subject: [Freeswitch-users] tls srtp calls between users white noise In-Reply-To: <8a5988df-e823-05fb-562e-5bb0620d068d@xbipin.com> References: <8a5988df-e823-05fb-562e-5bb0620d068d@xbipin.com> Message-ID: hi, dont bother, i managed to figure this out, what i needed to do was export absolute_codec_string with PCMU before the bridge Regards, Bipin ------------------------------------------------------------------------ -------- Original Message -------- Subject: [Freeswitch-users] tls srtp calls between users white noise From: Bipin Patel To: FreeSWITCH Users Help Date: 9/23/2017, 1:26:52 AM > hi, > > i got tls and srtp working between client and FS, client registers to > FS using TLS and SRTP and if that call is forwarded to gateway in SIP > and RTP everything works fine. > > Problem arises when 2 clients using zoiper register to FS using TLS > and SRTP and any calls routed between them the calls goes through and > the phone rings but on answering there is white noise constantly, i > checked in FS and it does send the call using TLS and SRTP and key > exchange and codec negotiation all is fine using g711u but i cant seem > to get rid of this white noise almost making it impossible to each > party to hear the other. > > any idea what could be wrong > > my dialplan looks like this > >     >       >       >           >       >     > >     >       >               >       >     > >     >       >         >         >       >     > > my porfile looks like this > >     >     >     >     >     >     >     >     >     > > > -- > Regards, > Bipin > > > ------------------------------------------------------------------------ > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From mirkobrankovic at gmail.com Sat Sep 23 13:12:46 2017 From: mirkobrankovic at gmail.com (Mirko Brankovic) Date: Sat, 23 Sep 2017 15:12:46 +0200 Subject: [Freeswitch-users] ICE/DTLS handshake In-Reply-To: References: Message-ID: Thanks Mike, all clients are webrtc clients behind same freeswitch. At the same time video rtp/rtcp dtls is instant, but audio is waiting for something, my best guess is for rtcp from client to confirm correct ip.port or to do auto correct. Call scenario is A call B and B transfers(refer) call to C. Also I see in those 5 seconds that first client, A sends total of 6 stun username requests and that C answers to them all at the same time, after 5s. Can this be rtcp problem. I was thinking to go through video ice thread and compare it to audio to see how that one works instantly. thanks, Mirko On Sep 21, 2017 19:55, "Michael Jerris" wrote: its not going to negotiate until we get the stun responses. If we are not, you should look if the client is sending them and something is blocking, or why the client is waiting to send them. Sounds broken on client side from the description. On Sep 21, 2017, at 7:53 AM, Mirko Brankovic wrote: HI, Has anyone experienced DTLS handshake takes 5s to get to SETUP state: > 2017-09-21 11:40:40.021178 [INFO] switch_rtp.c:3515 Changing audio DTLS > state from OFF to HANDSHAKE > 2017-09-21 11:40:45.319457 [INFO] switch_rtp.c:3172 Changing audio DTLS > state from HANDSHAKE to SETUP In network dump i see that answering side is not sending STUN for this 5s and then suddenly answers last 5 STUNs from A side. Has anyone encountered this kind of problem ? I have a pcap if necessary... _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From alfred.heggestad at gmail.com Sat Sep 23 17:30:16 2017 From: alfred.heggestad at gmail.com (Alfred E. Heggestad) Date: Sat, 23 Sep 2017 19:30:16 +0200 Subject: [Freeswitch-users] [OT] baresip Message-ID: <8b028afd-74ce-ec34-9e4c-19f18a748e95@gmail.com> Hi, My apologies for this off-topic post, but I hope that this can be of interest to the nice Freeswitch community. We have released a new version of Baresip which is a open source SIP client that is compatible with Freeswitch. you can find the source code here: https://github.com/alfredh/baresip/releases/tag/v0.5.5 We are doing regular interop-testing with Freeswitch, mainly SIP and RTP related features. From the Baresip project we send the best greetings to the nice people on this list :) /alfred From gmaruzz at gmail.com Sun Sep 24 05:24:08 2017 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Sun, 24 Sep 2017 07:24:08 +0200 Subject: [Freeswitch-users] [OT] baresip In-Reply-To: <8b028afd-74ce-ec34-9e4c-19f18a748e95@gmail.com> References: <8b028afd-74ce-ec34-9e4c-19f18a748e95@gmail.com> Message-ID: Thanks Alfred! -giovanni On 23 September 2017 at 19:30, Alfred E. Heggestad < alfred.heggestad at gmail.com> wrote: > Hi, > > > My apologies for this off-topic post, but I hope that this > can be of interest to the nice Freeswitch community. > > > We have released a new version of Baresip which is a > open source SIP client that is compatible with Freeswitch. > you can find the source code here: > > https://github.com/alfredh/baresip/releases/tag/v0.5.5 > > > We are doing regular interop-testing with Freeswitch, > mainly SIP and RTP related features. > > > From the Baresip project we send the best greetings to > the nice people on this list :) > > > > > /alfred > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From rick at magicmail.mooo.com Sun Sep 24 15:17:34 2017 From: rick at magicmail.mooo.com (Rick Jarvis) Date: Sun, 24 Sep 2017 16:17:34 +0100 Subject: [Freeswitch-users] JSON API Message-ID: Did anything ever come of this http://lists.freeswitch.org/pipermail/freeswitch-users/2015-August/115482.html ? I’ve given up with trying to get information over WSS, so am back to using Event Socket - it would be good to know exactly which commands I can use to get JSON, or whether I’ll need to write a parser… ‘show channels’ for instance just errors... -------------- next part -------------- An HTML attachment was scrubbed... URL: From jeff.chua.linux at gmail.com Sun Sep 24 16:44:35 2017 From: jeff.chua.linux at gmail.com (Jeff Chua) Date: Mon, 25 Sep 2017 00:44:35 +0800 Subject: [Freeswitch-users] gsmopen one-way voice issue In-Reply-To: References: Message-ID: On Wed, Sep 20, 2017 at 6:44 PM, Giovanni Maruzzelli wrote: > try to change the interface devices that are used by gsopen, many times > modem (dongle) vendor are messing with them. Rerea the documentation, and > check out how to change those devices, eg, in various permutations. > This is the only thing that comes to my mind. If this do not work, you will > need to debug it deeply. Giovanni, I kept getting this message ... could this be causing the no-sound from cellphone to freeswitch? 2017-09-25 00:21:48.187031 [DEBUG] mod_gsmopen.cpp:688 rev 1.9.0~64bit [(nil)|37 ][DEBUG_GSMOPEN 688 ][gsm01 ][-1, 5, 5] read more than 320, samples=640 Thanks, Jeff. From gmaruzz at gmail.com Sun Sep 24 17:06:29 2017 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Sun, 24 Sep 2017 19:06:29 +0200 Subject: [Freeswitch-users] gsmopen one-way voice issue In-Reply-To: References: Message-ID: On 24 September 2017 at 18:44, Jeff Chua wrote: > On Wed, Sep 20, 2017 at 6:44 PM, Giovanni Maruzzelli > wrote: > > try to change the interface devices that are used by gsopen, many times > > modem (dongle) vendor are messing with them. Rerea the documentation, and > > check out how to change those devices, eg, in various permutations. > > This is the only thing that comes to my mind. If this do not work, you > will > > need to debug it deeply. > > Giovanni, > > I kept getting this message ... could this be causing the no-sound > from cellphone to freeswitch? > > 2017-09-25 00:21:48.187031 [DEBUG] mod_gsmopen.cpp:688 rev 1.9.0~64bit > [(nil)|37 ][DEBUG_GSMOPEN 688 ][gsm01 ][-1, 5, 5] read more > than 320, samples=640 > No idea. The only suggestion/hint comes to my mind is: check if during initialization, and then when calls are made or received, it uses exactly the same AT commands and arguments as in the Asterisk channel you saw is working with your modem. If not, patch it to do the same when your modem model is detected, and then post a pull request, or send me a patch. > > > Thanks, > Jeff. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From kamal.mustafa at gmail.com Mon Sep 25 02:20:21 2017 From: kamal.mustafa at gmail.com (Kamal Mustafa) Date: Mon, 25 Sep 2017 10:20:21 +0800 Subject: [Freeswitch-users] Python-ESL: ESLconnection.execute() blocked when handling channel_park under thread In-Reply-To: References: Message-ID: On 10 May 2017 at 02:39, Lesley Pervis wrote: > Late to answer. Seems no one did, so I'd say don't write event listeners > that run in a Python interpreter that FreeSWITCH knows nothing about and > uses threading. Hi, thanks for the reply and advice. I'd appreciate that. It's been quite sometimes after I'm working on this (forgot about this email already) but with recent problem I have, I think we have to open new connection in the handler thread if we want to send another command over the esl socket. We can't reuse the same connection in the main thread. Since we're at this, what you think about doing billing in python ? It sound crazy but after seeing cgrates also doing the billing over the event socket, I think maybe it feasible - https://github.com/cgrates/cgrates/blob/master/docs/freeswitch.rst#811-sessionmanager I've started a PoC of billing in python and it seem to work, although I haven't put it under load and concurrent tests yet. From rfmundkowsky at yahoo.com Mon Sep 25 03:32:02 2017 From: rfmundkowsky at yahoo.com (Robert Mundkowsky) Date: Sun, 24 Sep 2017 23:32:02 -0400 Subject: [Freeswitch-users] Python-ESL: ESLconnection.execute() blocked when handling channel_park under thread Message-ID: Any reason you are against Python event handlers? -------- Original message -------- From: Kamal Mustafa Date: 9/24/17 10:20 PM (GMT-05:00) To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Python-ESL: ESLconnection.execute() blocked when handling channel_park under thread On 10 May 2017 at 02:39, Lesley Pervis wrote: > Late to answer. Seems no one did, so I'd say don't write event listeners > that run in a Python interpreter that FreeSWITCH knows nothing about and > uses threading. Hi, thanks for the reply and advice. I'd appreciate that. It's been quite sometimes after I'm working on this (forgot about this email already) but with recent problem I have, I think we have to open new connection in the handler thread if we want to send another command over the esl socket. We can't reuse the same connection in the main thread. Since we're at this, what you think about doing billing in python ? It sound crazy but after seeing cgrates also doing the billing over the event socket, I think maybe it feasible - https://github.com/cgrates/cgrates/blob/master/docs/freeswitch.rst#811-sessionmanager I've started a PoC of billing in python and it seem to work, although I haven't put it under load and concurrent tests yet. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From josedavid at zennio.com Mon Sep 25 08:28:14 2017 From: josedavid at zennio.com (Jose David Jurado Alonso) Date: Mon, 25 Sep 2017 10:28:14 +0200 Subject: [Freeswitch-users] How can I use an external database to create new users? Message-ID: Hi, I'm thinking to create a web application to create, edit and delete users in a simpler way than creating XML by hand but I don't find information about it. I've seen something about the module "mod_xml_curl" but I'm not sure how it interacts with the database nor if it's necessary some additional module more. I would like to use MySQL or postgreSQL, does anyone know how you can do it or some tutorial or reference application? Thanks, José D. -------------- next part -------------- An HTML attachment was scrubbed... URL: From tg-maillistings at level5.de Mon Sep 25 10:44:48 2017 From: tg-maillistings at level5.de (=?UTF-8?Q?Thorsten_G=c3=b6llner?=) Date: Mon, 25 Sep 2017 12:44:48 +0200 Subject: [Freeswitch-users] How can I use an external database to create new users? In-Reply-To: References: Message-ID: <2b6527f0-b831-fb2b-4a8e-47b43cdb4c87@level5.de> Hi, did you take a look at existing (Web)Guis like FusionPBX? https://www.fusionpbx.com/ Thorsten Am 25.09.2017 um 10:28 schrieb Jose David Jurado Alonso: > Hi, > > I'm thinking to create a web application to create, edit and delete > users in a simpler way than creating XML by hand but I don't find > information about it.  > > I've seen something about the module "mod_xml_curl" but I'm not sure > how it interacts with the database nor if it's necessary some > additional module more. > > I would like to use MySQL or postgreSQL, does anyone know how you can > do it or some tutorial or reference application? > > Thanks, > > José D. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From josedavid at zennio.com Mon Sep 25 11:21:27 2017 From: josedavid at zennio.com (Jose David Jurado Alonso) Date: Mon, 25 Sep 2017 13:21:27 +0200 Subject: [Freeswitch-users] How can I use an external database to create new users? In-Reply-To: <2b6527f0-b831-fb2b-4a8e-47b43cdb4c87@level5.de> References: <2b6527f0-b831-fb2b-4a8e-47b43cdb4c87@level5.de> Message-ID: Hi, Yes, but I need a custom web app to users management. I tried using postgresql as core-db-dsn and I work properly in some cases (register appears) but no show any about users. José David Jurado Alonso *Área de Desarrollo de Software de Alto Nivel* *Dpto. Ingeniería* [image: Descripción: cid:image010.jpg at 01D27631.DF7E30F0] Zennio Avance y Tecnología, S.L. C/ Rio Jarama,132. 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Los datos por usted suministrados serán empleados con fines de gestión, Zennio Avance y Tecnología S.L ha adoptado las medidas de seguridad exigidas en función del nivel de los datos suministrados, instalando las medidas técnicas y organizativas necesarias, habida cuenta del estado de la tecnología, a fin de evitar su pérdida, alteración, uso inadecuado o accesos no autorizados a los mismos. Para el ejercicio de sus derechos de acceso, rectificación, cancelación y oposición deberá dirigirse a la dirección del Responsable de Fichero Zennio Avance y Tecnología S.L C/ Rio Jarama, 132. Nave P-8.11, 45007, TOLEDO o a la dirección de correo electrónico: info at zennio.com Please, consider the environment before printing this e-mail... Save energy! Por favor, piensa en el medio ambiente antes de imprimir este e-mail... ¡Ahorra energía! Disclaimer: This message and any attached files transmitted with it, is confidential, especially as regards personal data. 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Advertencia legal: Este mensaje y, en su caso, los ficheros anexos son confidenciales, especialmente en lo que respecta a los datos personales, y se dirigen exclusivamente al destinatario referenciado. Si usted no lo es y lo ha recibido por error o tiene conocimiento del mismo por cualquier motivo, le rogamos que nos lo comunique por este medio y proceda a destruirlo o borrarlo, y que en todo caso se abstenga de utilizar, reproducir, alterar, archivar o comunicar a terceros el presente mensaje y ficheros anexos, todo ello bajo pena de incurrir en responsabilidades legales. Las opiniones contenidas en este mensaje y en los archivos adjuntos, pertenecen exclusivamente a su remitente y no representan la opinión de la empresa salvo que se diga expresamente y el remitente esté autorizado para ello. El emisor no garantiza la integridad, rapidez o seguridad del presente correo, ni se responsabiliza de posibles perjuicios derivados de la captura, incorporaciones de virus o cualesquiera otras manipulaciones efectuadas por terceros. 2017-09-25 12:44 GMT+02:00 Thorsten Göllner : > Hi, > > did you take a look at existing (Web)Guis like FusionPBX? > > https://www.fusionpbx.com/ > > Thorsten > > > Am 25.09.2017 um 10:28 schrieb Jose David Jurado Alonso: > > Hi, > > I'm thinking to create a web application to create, edit and delete users > in a simpler way than creating XML by hand but I don't find information > about it. > > I've seen something about the module "mod_xml_curl" but I'm not sure how > it interacts with the database nor if it's necessary some additional module > more. > > I would like to use MySQL or postgreSQL, does anyone know how you can do > it or some tutorial or reference application? > > Thanks, > > José D. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > -------------- next part -------------- An HTML attachment was scrubbed... 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Name: image004.jpg Type: image/jpeg Size: 966 bytes Desc: not available URL: From igorolhovskiy at gmail.com Mon Sep 25 13:05:45 2017 From: igorolhovskiy at gmail.com (Igor Olhovskiy) Date: Mon, 25 Sep 2017 16:05:45 +0300 Subject: [Freeswitch-users] How can I use an external database to create new users? In-Reply-To: References: <2b6527f0-b831-fb2b-4a8e-47b43cdb4c87@level5.de> Message-ID: <7b4a8ce4-0136-465c-af1b-e7c62ec1dbe6@Spark> mod_xml_curl will ask some http server with username and domain and will require as a response XML-formatted directory entry. So you need to write some intermediate layer, that will on request go to database and provide back info in necessary format. Regards, Igor On 25 сент. 2017 г., 14:22 +0300, Jose David Jurado Alonso , wrote: > Hi, > > Yes, but I need a custom web app to users management. > > I tried using postgresql as core-db-dsn and I work properly in some cases (register appears) but no show any about users. > > José David Jurado Alonso > Área de Desarrollo de Software de Alto Nivel > Dpto. Ingeniería > > Zennio Avance y Tecnología, S.L. > C/ Rio Jarama,132. Nave P-8.11 > 45007 - Toledo (Spain) > T: +34 925 232 002 > www.zennio.com > > Zennio Avance y Tecnología S.L le informa de los siguientes extremos: > Los datos por usted suministrados pasarán a formar parte de un fichero cuyo responsable es Zennio Avance y Tecnología S.L. Dicho fichero se encuentra legalmente inscrito en el Registro General de Protección de Datos de la Agencia Española de Protección de Datos. Los datos por usted suministrados serán empleados con fines de gestión, Zennio Avance y Tecnología S.L ha adoptado las medidas de seguridad exigidas en función del nivel de los datos suministrados, instalando las medidas técnicas y organizativas necesarias, habida cuenta del estado de la tecnología, a fin de evitar su pérdida, alteración, uso inadecuado o accesos no autorizados a los mismos. > Para el ejercicio de sus derechos de acceso, rectificación, cancelación y oposición deberá dirigirse a la dirección del Responsable de Fichero Zennio Avance y Tecnología S.L C/ Rio Jarama, 132. Nave P-8.11, 45007, TOLEDO o a la dirección de correo electrónico: info at zennio.com > Please, consider the environment before printing this e-mail... Save energy! > Por favor, piensa en el medio ambiente antes de imprimir este e-mail... ¡Ahorra energía! > Disclaimer: > This message and any attached files transmitted with it, is confidential, especially as regards personal data. It is intended solely for the use of the individual or entity to whom it is addressed. If you are not the intended recipient and have received this information in error or have accessed it for any reason, please notify us of this fact by email reply and then destroy or delete the message, refraining from any reproduction, use, alteration, filing or communication to third parties of this message and attached files on penalty of incurring legal responsibilities. The opinions contained in this message and the attached archives, belong exclusively to their sender and they do not represent the opinion of the company unless it is said specifically and the sender is authorized for it. The sender does not guarantee the integrity, the accuracy, the swift delivery or the security of this email transmission, and assumes no responsibility for any possible damage incurred through data capture, virus incorporation or any manipulation carried out by third parties. > Advertencia legal: > Este mensaje y, en su caso, los ficheros anexos son confidenciales, especialmente en lo que respecta a los datos personales, y se dirigen exclusivamente al destinatario referenciado. Si usted no lo es y lo ha recibido por error o tiene conocimiento del mismo por cualquier motivo, le rogamos que nos lo comunique por este medio y proceda a destruirlo o borrarlo, y que en todo caso se abstenga de utilizar, reproducir, alterar, archivar o comunicar a terceros el presente mensaje y ficheros anexos, todo ello bajo pena de incurrir en responsabilidades legales. Las opiniones contenidas en este mensaje y en los archivos adjuntos, pertenecen exclusivamente a su remitente y no representan la opinión de la empresa salvo que se diga expresamente y el remitente esté autorizado para ello. El emisor no garantiza la integridad, rapidez o seguridad del presente correo, ni se responsabiliza de posibles perjuicios derivados de la captura, incorporaciones de virus o cualesquiera otras manipulaciones efectuadas por terceros. > > > 2017-09-25 12:44 GMT+02:00 Thorsten Göllner : > > > Hi, > > > > > > did you take a look at existing (Web)Guis like FusionPBX? > > > > > > https://www.fusionpbx.com/ > > > > > > Thorsten > > > > > > > > > Am 25.09.2017 um 10:28 schrieb Jose David Jurado Alonso: > > > > Hi, > > > > > > > > I'm thinking to create a web application to create, edit and delete users in a simpler way than creating XML by hand but I don't find information about it. > > > > > > > > I've seen something about the module "mod_xml_curl" but I'm not sure how it interacts with the database nor if it's necessary some additional module more. > > > > > > > > I would like to use MySQL or postgreSQL, does anyone know how you can do it or some tutorial or reference application? > > > > > > > > Thanks, > > > > > > > > José D. > > > > > > > > > > > > _________________________________________________________________________ > > > > Professional FreeSWITCH Consulting Services: > > > > consulting at freeswitch.org > > > > http://www.freeswitchsolutions.com > > > > > > > > Official FreeSWITCH Sites > > > > http://www.freeswitch.org > > > > http://confluence.freeswitch.org > > > > http://www.cluecon.com > > > > > > > > FreeSWITCH-users mailing list > > > > FreeSWITCH-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... 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Name: 523FFCFF7BF046C1B5CFA91015D2762F.png Type: image/png Size: 3473 bytes Desc: not available URL: From bipin at xbipin.com Mon Sep 25 13:44:01 2017 From: bipin at xbipin.com (Bipin Patel) Date: Mon, 25 Sep 2017 17:44:01 +0400 Subject: [Freeswitch-users] How can I use an external database to create new users? In-Reply-To: <7b4a8ce4-0136-465c-af1b-e7c62ec1dbe6@Spark> References: <2b6527f0-b831-fb2b-4a8e-47b43cdb4c87@level5.de> <7b4a8ce4-0136-465c-af1b-e7c62ec1dbe6@Spark> Message-ID: <0db247db-dc52-cb83-f818-ab54c633de8e@xbipin.com> hi, we use a different switch with its db running on mysql but sadly that doesnt supports tls and srtp so we use freeswitch for that so users register to FS and we use xml curl for directory users where FS calls a php script which inturn fetches the id/pass of the user and returns it to FS which then allows users to register to FS. The script is only called when users register to FS, its quiet simple if u ask me. Regards, Bipin ------------------------------------------------------------------------ -------- Original Message -------- Subject: Re: [Freeswitch-users] How can I use an external database to create new users? From: Igor Olhovskiy To: Thorsten Göllner , FreeSWITCH Users Help Date: 9/25/2017, 5:05:45 PM > mod_xml_curl will ask some http server with username and domain and > will require as a response XML-formatted directory entry. > So you need to write some intermediate layer, that will on request go > to database and provide back info in necessary format. > > Regards, Igor > > On 25 сент. 2017 г., 14:22 +0300, Jose David Jurado Alonso > , wrote: >> Hi, >> >> Yes, but I need a custom web app to users management. >> >> I tried using postgresql as core-db-dsn and I work properly in some >> cases (register appears) but no show any about users. >> >> José David Jurado Alonso > > >> >> /Área de Desarrollo de Software de Alto Nivel/ >> >> *Dpto. Ingeniería* >> >> Descripción: cid:image010.jpg at 01D27631.DF7E30F0 >> >> Zennio Avance y Tecnología, S.L. >> >> C/ Rio Jarama,132. Nave P-8.11 >> >> 45007 - Toledo (Spain) >> >> T: +34 925 232 002 >> >> www.zennio.com >> >> Descripción: Descripción: C:\Users\jjmanjarres\Desktop\Z zennio.jpg >> Descripción: Descripción: >> C:\Users\jjmanjarres\Desktop\twitter.jpg >> Descripción: Descripción: >> C:\Users\jjmanjarres\Desktop\descarga.jpg >> Descripción: >> Descripción: C:\Users\jjmanjarres\Desktop\linkedin.png >> Descripción: Descripción: >> C:\Users\jjmanjarres\Desktop\youtube.jpg >> >> >> Zennio Avance y Tecnología S.L le informa de los siguientes extremos: >> Los datos por usted suministrados pasarán a formar parte de un >> fichero cuyo responsable es Zennio Avance y Tecnología S.L. Dicho >> fichero se encuentra legalmente inscrito en el Registro General de >> Protección de Datos de la Agencia Española de Protección de Datos. >> Los datos por usted suministrados serán empleados con fines de >> gestión, Zennio Avance y Tecnología S.L ha adoptado las medidas de >> seguridad exigidas en función del nivel de los datos suministrados, >> instalando las medidas técnicas y organizativas necesarias, habida >> cuenta del estado de la tecnología, a fin de evitar su pérdida, >> alteración, uso inadecuado o accesos no autorizados a los mismos. >> Para el ejercicio de sus derechos de acceso, rectificación, >> cancelación y oposición deberá dirigirse a la dirección del >> Responsable de Fichero Zennio Avance y Tecnología S.L C/ Rio Jarama, >> 132. Nave P-8.11, 45007, TOLEDO o a la dirección de correo >> electrónico: info at zennio.com >> >> Please, consider the environment before printing this e-mail... Save >> energy! >> >> Por favor, piensa en el medio ambiente antes de imprimir este >> e-mail... ¡Ahorra energía! >> >> Disclaimer: >> This message and any attached files transmitted with it, is >> confidential, especially as regards personal data. It is intended >> solely for the use of the individual or entity to whom it is >> addressed. If you are not the intended recipient and have received >> this information in error or have accessed it for any reason, please >> notify us of this fact by email reply and then destroy or delete the >> message, refraining from any reproduction, use, alteration, filing or >> communication to third parties of this message and attached files on >> penalty of incurring legal responsibilities. The opinions contained >> in this message and the attached archives, belong exclusively to >> their sender and they do not represent the opinion of the company >> unless it is said specifically and the sender is authorized for it. >> The sender does not guarantee the integrity, the accuracy, the swift >> delivery or the security of this email transmission, and assumes no >> responsibility for any possible damage incurred through data capture, >> virus incorporation or any manipulation carried out by third parties. >> >> Advertencia legal: >> Este mensaje y, en su caso, los ficheros anexos son confidenciales, >> especialmente en lo que respecta a los datos personales, y se dirigen >> exclusivamente al destinatario referenciado. Si usted no lo es y lo >> ha recibido por error o tiene conocimiento del mismo por cualquier >> motivo, le rogamos que nos lo comunique por este medio y proceda a >> destruirlo o borrarlo, y que en todo caso se abstenga de utilizar, >> reproducir, alterar, archivar o comunicar a terceros el presente >> mensaje y ficheros anexos, todo ello bajo pena de incurrir en >> responsabilidades legales. Las opiniones contenidas en este mensaje y >> en los archivos adjuntos, pertenecen exclusivamente a su remitente y >> no representan la opinión de la empresa salvo que se diga >> expresamente y el remitente esté autorizado para ello. El emisor no >> garantiza la integridad, rapidez o seguridad del presente correo, ni >> se responsabiliza de posibles perjuicios derivados de la captura, >> incorporaciones de virus o cualesquiera otras manipulaciones >> efectuadas por terceros. >> >> >> 2017-09-25 12:44 GMT+02:00 Thorsten Göllner >> >: >> >> Hi, >> >> did you take a look at existing (Web)Guis like FusionPBX? >> >> https://www.fusionpbx.com/ >> >> Thorsten >> >> >> Am 25.09.2017 um 10:28 schrieb Jose David Jurado Alonso: >>> Hi, >>> >>> I'm thinking to create a web application to create, edit and >>> delete users in a simpler way than creating XML by hand but I >>> don't find information about it. >>> >>> I've seen something about the module "mod_xml_curl" but I'm not >>> sure how it interacts with the database nor if it's necessary >>> some additional module more. >>> >>> I would like to use MySQL or postgreSQL, does anyone know how >>> you can do it or some tutorial or reference application? >>> >>> Thanks, >>> >>> José D. >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >>> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... 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Name: 523FFCFF7BF046C1B5CFA91015D2762F.png Type: image/png Size: 3473 bytes Desc: not available URL: From colin.morelli at gmail.com Mon Sep 25 15:10:31 2017 From: colin.morelli at gmail.com (Colin Morelli) Date: Mon, 25 Sep 2017 11:10:31 -0400 Subject: [Freeswitch-users] Temporary Equivalent of fs_path Message-ID: Hey all, Trying to figure out how to get the equivalent behavior of fs_path, but only for a single transaction. In other words, I want to start a SIP request hitting a particular proxy, and then simply let Record-Route headers determine where the call should be routed after that. I might be completely overthinking this, but everything I've tried so far (sip_route_uri, fs_path), have resulted in Freeswitch continuing to use the given route for all subsequent requests, rather than simply falling back to the session route. Am I missing something? Best, Colin -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Mon Sep 25 17:11:43 2017 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Mon, 25 Sep 2017 19:11:43 +0200 Subject: [Freeswitch-users] How can I use an external database to create new users? In-Reply-To: <0db247db-dc52-cb83-f818-ab54c633de8e@xbipin.com> References: <2b6527f0-b831-fb2b-4a8e-47b43cdb4c87@level5.de> <7b4a8ce4-0136-465c-af1b-e7c62ec1dbe6@Spark> <0db247db-dc52-cb83-f818-ab54c633de8e@xbipin.com> Message-ID: You can check http://freeswitch.org/confluence searching for mod_xml_curl Also, if you git clone freeswitch.contrib you will fond many examples, particularly intralanman php Obviously, you can also read the many FreeSWITCH books published by Packt. -giovanni sent from mobile cell: +39 347 266 56 18 Giovanni Maruzzelli OpenTelecom.IT On Sep 25, 2017 15:44, "Bipin Patel" wrote: > hi, > > we use a different switch with its db running on mysql but sadly that > doesnt supports tls and srtp so we use freeswitch for that so users > register to FS and we use xml curl for directory users where FS calls a php > script which inturn fetches the id/pass of the user and returns it to FS > which then allows users to register to FS. The script is only called when > users register to FS, its quiet simple if u ask me. > > > Regards, > Bipin > > > ------------------------------ > -------- Original Message -------- > Subject: Re: [Freeswitch-users] How can I use an external database to > create new users? > From: Igor Olhovskiy > To: Thorsten Göllner > , FreeSWITCH Users Help > > > Date: 9/25/2017, 5:05:45 PM > > mod_xml_curl will ask some http server with username and domain and will > require as a response XML-formatted directory entry. > So you need to write some intermediate layer, that will on request go to > database and provide back info in necessary format. > > Regards, Igor > > On 25 сент. 2017 г., 14:22 +0300, Jose David Jurado Alonso > , wrote: > > Hi, > > Yes, but I need a custom web app to users management. > > I tried using postgresql as core-db-dsn and I work properly in some cases > (register appears) but no show any about users. > > José David Jurado Alonso > > *Área de Desarrollo de Software de Alto Nivel* > > *Dpto. Ingeniería* > > [image: Descripción: cid:image010.jpg at 01D27631.DF7E30F0] > > Zennio Avance y Tecnología, S.L. > > C/ Rio Jarama,132. Nave P-8.11 > > 45007 - Toledo (Spain) > > T: +34 925 232 002 <+34%20925%2023%2020%2002> > > www.zennio.com > > [image: Descripción: Descripción: C:\Users\jjmanjarres\Desktop\Z > zennio.jpg] [image: Descripción: Descripción: > C:\Users\jjmanjarres\Desktop\twitter.jpg] > [image: Descripción: Descripción: > C:\Users\jjmanjarres\Desktop\descarga.jpg] > [image: Descripción: > Descripción: C:\Users\jjmanjarres\Desktop\linkedin.png] > [image: Descripción: > Descripción: C:\Users\jjmanjarres\Desktop\youtube.jpg] > > > Zennio Avance y Tecnología S.L le informa de los siguientes extremos: > Los datos por usted suministrados pasarán a formar parte de un fichero > cuyo responsable es Zennio Avance y Tecnología S.L. Dicho fichero se > encuentra legalmente inscrito en el Registro General de Protección de Datos > de la Agencia Española de Protección de Datos. Los datos por usted > suministrados serán empleados con fines de gestión, Zennio Avance y > Tecnología S.L ha adoptado las medidas de seguridad exigidas en función del > nivel de los datos suministrados, instalando las medidas técnicas y > organizativas necesarias, habida cuenta del estado de la tecnología, a fin > de evitar su pérdida, alteración, uso inadecuado o accesos no autorizados a > los mismos. > Para el ejercicio de sus derechos de acceso, rectificación, cancelación y > oposición deberá dirigirse a la dirección del Responsable de Fichero Zennio > Avance y Tecnología S.L C/ Rio Jarama, 132. Nave P-8.11, 45007, TOLEDO o a > la dirección de correo electrónico: info at zennio.com > > Please, consider the environment before printing this e-mail... Save > energy! > > Por favor, piensa en el medio ambiente antes de imprimir este e-mail... ¡Ahorra > energía! > > Disclaimer: > This message and any attached files transmitted with it, is confidential, > especially as regards personal data. It is intended solely for the use of > the individual or entity to whom it is addressed. If you are not the > intended recipient and have received this information in error or have > accessed it for any reason, please notify us of this fact by email reply > and then destroy or delete the message, refraining from any reproduction, > use, alteration, filing or communication to third parties of this message > and attached files on penalty of incurring legal responsibilities. The > opinions contained in this message and the attached archives, belong > exclusively to their sender and they do not represent the opinion of the > company unless it is said specifically and the sender is authorized for it. > The sender does not guarantee the integrity, the accuracy, the swift > delivery or the security of this email transmission, and assumes no > responsibility for any possible damage incurred through data capture, virus > incorporation or any manipulation carried out by third parties. > > Advertencia legal: > Este mensaje y, en su caso, los ficheros anexos son confidenciales, > especialmente en lo que respecta a los datos personales, y se dirigen > exclusivamente al destinatario referenciado. 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Name: 523FFCFF7BF046C1B5CFA91015D2762F.png Type: image/png Size: 3473 bytes Desc: not available URL: From voransoy at gmail.com Mon Sep 25 19:21:48 2017 From: voransoy at gmail.com (Volkan Oransoy) Date: Mon, 25 Sep 2017 22:21:48 +0300 Subject: [Freeswitch-users] Opus problem Message-ID: <95869C0B-F054-442D-8E5F-E788A5B439A6@gmail.com> Hi All, I am trying to transcode between OPUS and G711. I encountered a strange problem which I couldn’t find any clue about the cause. After the channel has been answered with opus, I hear a strange sound (highly distorted version of my voice) for 10 seconds and it starts to operate properly with the message "[DEBUG] switch_core_media_bug.c:1198 Removing BUG from…” . I can reproduce same problem with echo application. Here is my logs and opus.conf.xml files. Did anybody experienced this issue? Thanks. /Volkan My FreeSWITH Version: FreeSWITCH Version 1.6.16+git~20170407T164913Z~3da6bd0108~64bit (git 3da6bd0 2017-04-07 16:49:13Z 64bit) Opus.xml config: Freeswitch Log: 2017-09-25 22:01:06.277462 [NOTICE] switch_channel.c:1104 New Channel sofia/internalopus/2223 at sip.example.com [1048c869-8d71-41fd-b0b7-f8fbd01a0d93] 2017-09-25 22:01:06.277462 [DEBUG] switch_core_state_machine.c:584 (sofia/internalopus/2223 at sip.example.com) Running State Change CS_NEW (Cur 2 Tot 3196652) 2017-09-25 22:01:06.277462 [DEBUG] sofia.c:9828 sofia/internalopus/2223 at sip.example.com receiving invite from 192.168.100.2:5060 version: 1.6.16 git 3da6bd0 2017-04-07 16:49:13Z 64bit 2017-09-25 22:01:06.277462 [DEBUG] sofia.c:7048 Channel sofia/internalopus/2223 at sip.example.com entering state [received][100] 2017-09-25 22:01:06.277462 [DEBUG] sofia.c:7058 Remote SDP: v=0 o=- 8070049943 45777 IN IP4 192.168.100.2 s=dvjtiwu c=IN IP4 192.168.100.2 t=0 0 m=audio 10036 RTP/AVP 103 8 0 9 101 a=rtpmap:103 opus/48000/2 a=fmtp:103 maxplaybackrate=8000;maxaveragebitrate=15500;useinbandfec=1;usedtx=1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=nortpproxy:yes . . . . . 2017-09-25 22:01:06.337394 [INFO] mod_dialplan_xml.c:637 Processing 2223 <2223>->*71 in context default Dialplan: sofia/internalopus/2223 at sip.example.com parsing [default->echo] continue=false Dialplan: sofia/internalopus/2223 at sip.example.com Regex (PASS) [echo] destination_number(*71) =~ /^\*71$/ break=on-false Dialplan: sofia/internalopus/2223 at sip.example.com Action answer() Dialplan: sofia/internalopus/2223 at sip.example.com Action echo() 2017-09-25 22:01:06.357410 [DEBUG] switch_core_state_machine.c:286 (sofia/internalopus/2223 at sip.example.com) State Change CS_ROUTING -> CS_EXECUTE 2017-09-25 22:01:06.357410 [DEBUG] switch_core_state_machine.c:643 (sofia/internalopus/2223 at sip.example.com) State ROUTING going to sleep 2017-09-25 22:01:06.357410 [DEBUG] switch_core_state_machine.c:584 (sofia/internalopus/2223 at sip.example.com) Running State Change CS_EXECUTE (Cur 2 Tot 3196652) 2017-09-25 22:01:06.357410 [DEBUG] switch_core_state_machine.c:650 (sofia/internalopus/2223 at sip.example.com) State EXECUTE 2017-09-25 22:01:06.357410 [DEBUG] mod_sofia.c:198 sofia/internalopus/2223 at sip.example.com SOFIA EXECUTE 2017-09-25 22:01:06.357410 [DEBUG] switch_core_state_machine.c:328 sofia/internalopus/2223 at sip.example.com Standard EXECUTE EXECUTE sofia/internalopus/2223 at sip.example.com answer() 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [opus:103:48000:20:0:1]/[opus:116:48000:20:0:1] 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4484 Audio Codec Compare [opus:116:48000:20:0:1] ++++ is saved as a match 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [opus:103:48000:20:0:1]/[PCMU:0:8000:20:64000:1] 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [opus:103:48000:20:0:1]/[PCMA:8:8000:20:64000:1] 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[opus:116:48000:20:0:1] 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4484 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[opus:116:48000:20:0:1] 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4484 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [G722:9:8000:20:64000:1]/[opus:116:48000:20:0:1] 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4345 Set telephone-event payload to 101 at 8000 2017-09-25 22:01:06.357410 [DEBUG] mod_opus.c:596 Opus encoder: set bitrate based on maxaveragebitrate value found in SDP or local config [14400bps] 2017-09-25 22:01:06.357410 [DEBUG] mod_opus.c:613 Opus encoder: set audio bandwidth to [NARROWBAND] based on maxplaybackrate value found in SDP or local config [8000Hz] 2017-09-25 22:01:06.357410 [DEBUG] mod_opus.c:596 Opus encoder: set bitrate based on maxaveragebitrate value found in SDP or local config [14400bps] 2017-09-25 22:01:06.357410 [DEBUG] mod_opus.c:613 Opus encoder: set audio bandwidth to [NARROWBAND] based on maxplaybackrate value found in SDP or local config [8000Hz] 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:3049 Set Codec sofia/internalopus/2223 at sip.example.com opus/8000 20 ms 960 samples 0 bits 1 channels 2017-09-25 22:01:06.357410 [DEBUG] switch_core_codec.c:111 sofia/internalopus/2223 at sip.example.com Original read codec set to opus:116 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4747 sofia/internalopus/2223 at sip.example.com Set 2833 dtmf send payload to 101 recv payload to 101 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:6803 AUDIO RTP [sofia/internalopus/2223 at sip.example.com] 192.168.100.12 port 23594 -> 192.168.100.2 port 10036 codec: 103 ms: 20 2017-09-25 22:01:06.357410 [DEBUG] switch_rtp.c:4096 Starting timer [soft] 960 bytes per 20ms 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:7109 sofia/internalopus/2223 at sip.example.com Set 2833 dtmf send payload to 101 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:7116 sofia/internalopus/2223 at sip.example.com Set 2833 dtmf receive payload to 101 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:7139 sofia/internalopus/2223 at sip.example.com Set rtp dtmf delay to 40 2017-09-25 22:01:06.357410 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internalopus/2223 at sip.example.com! 2017-09-25 22:01:06.357410 [DEBUG] switch_channel.c:3473 (sofia/internalopus/2223 at sip.example.com) Callstate Change RINGING -> EARLY 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:6786 Audio params are unchanged for sofia/internalopus/2223 at sip.example.com. 2017-09-25 22:01:06.357410 [DEBUG] mod_sofia.c:850 Local SDP sofia/internalopus/2223 at sip.example.com: v=0 o=FreeSWITCH 1506342472 1506342473 IN IP4 192.168.100.12 s=FreeSWITCH c=IN IP4 192.168.100.12 t=0 0 m=audio 23594 RTP/AVP 103 101 a=rtpmap:103 opus/48000/2 a=fmtp:103 useinbandfec=1; usedtx=1; maxaveragebitrate=14400; maxplaybackrate=8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv 2017-09-25 22:01:06.357410 [NOTICE] mod_dptools.c:1312 Channel [sofia/internalopus/2223 at sip.example.com] has been answered 2017-09-25 22:01:06.357410 [DEBUG] sofia.c:7048 Channel sofia/internalopus/2223 at sip.example.com entering state [completed][200] 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media_bug.c:828 Attaching BUG to sofia/internalopus/2223 at sip.example.com 2017-09-25 22:01:06.357410 [DEBUG] switch_channel.c:3772 (sofia/internalopus/2223 at sip.example.com) Callstate Change EARLY -> ACTIVE EXECUTE sofia/internalopus/2223 at sip.example.com echo() 2017-09-25 22:01:06.737463 [DEBUG] sofia.c:7048 Channel sofia/internalopus/2223 at sip.example.com entering state [ready][200] 2017-09-25 22:01:06.837459 [DEBUG] switch_rtp.c:7229 Correct audio ip/port confirmed. 2017-09-25 22:01:06.837459 [DEBUG] switch_core_io.c:448 Setting BUG Codec opus:116 2017-09-25 22:01:06.837459 [DEBUG] mod_opus.c:596 Opus encoder: set bitrate based on maxaveragebitrate value found in SDP or local config [14400bps] 2017-09-25 22:01:06.837459 [DEBUG] mod_opus.c:613 Opus encoder: set audio bandwidth to [NARROWBAND] based on maxplaybackrate value found in SDP or local config [8000Hz] 2017-09-25 22:01:06.837459 [DEBUG] switch_core_io.c:780 Engaging Read Buffer at 320 bytes vs 15 [...CREEPY VOICE STARTS HERE AND ENDS WITH THE BELOW MESSAGE...] 2017-09-25 22:01:16.017463 [DEBUG] switch_core_media_bug.c:1198 Removing BUG from sofia/internalopus/2223 at sip.example.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Mon Sep 25 19:35:08 2017 From: mike at jerris.com (Michael Jerris) Date: Mon, 25 Sep 2017 19:35:08 +0000 Subject: [Freeswitch-users] Opus problem In-Reply-To: <95869C0B-F054-442D-8E5F-E788A5B439A6@gmail.com> References: <95869C0B-F054-442D-8E5F-E788A5B439A6@gmail.com> Message-ID: This is due to an incomplete implementation of the asymmetric sample rate feature. It should work fine without that setting. On Mon, Sep 25, 2017 at 2:22 PM Volkan Oransoy wrote: > Hi All, > > I am trying to transcode between OPUS and G711. I encountered a strange > problem which I couldn’t find any clue about the cause. After the channel > has been answered with opus, I hear a strange sound (highly distorted > version of my voice) for 10 seconds and it starts to operate properly with > the message "[DEBUG] switch_core_media_bug.c:1198 Removing BUG from…” . I > can reproduce same problem with echo application. Here is my logs and > opus.conf.xml files. Did anybody experienced this issue? > > Thanks. > > /Volkan > > > My FreeSWITH Version: > FreeSWITCH Version 1.6.16+git~20170407T164913Z~3da6bd0108~64bit (git > 3da6bd0 2017-04-07 16:49:13Z 64bit) > > Opus.xml config: > > > > > > > > > > > > > > > > Freeswitch Log: > 2017-09-25 22:01:06.277462 [NOTICE] switch_channel.c:1104 New Channel > sofia/internalopus/2223 at sip.example.com > [1048c869-8d71-41fd-b0b7-f8fbd01a0d93] > 2017-09-25 22:01:06.277462 [DEBUG] switch_core_state_machine.c:584 ( > sofia/internalopus/2223 at sip.example.com) Running State Change CS_NEW (Cur > 2 Tot 3196652) > 2017-09-25 22:01:06.277462 [DEBUG] sofia.c:9828 > sofia/internalopus/2223 at sip.example.com receiving invite from > 192.168.100.2:5060 version: 1.6.16 git 3da6bd0 2017-04-07 16:49:13Z 64bit > 2017-09-25 22:01:06.277462 [DEBUG] sofia.c:7048 Channel > sofia/internalopus/2223 at sip.example.com entering state [received][100] > 2017-09-25 22:01:06.277462 [DEBUG] sofia.c:7058 Remote SDP: > v=0 > o=- 8070049943 45777 IN IP4 192.168.100.2 > s=dvjtiwu > c=IN IP4 192.168.100.2 > t=0 0 > m=audio 10036 RTP/AVP 103 8 0 9 101 > a=rtpmap:103 opus/48000/2 > a=fmtp:103 > maxplaybackrate=8000;maxaveragebitrate=15500;useinbandfec=1;usedtx=1 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=ptime:20 > a=nortpproxy:yes > . > . > . > . > . > 2017-09-25 22:01:06.337394 [INFO] mod_dialplan_xml.c:637 Processing 2223 > <2223>->*71 in context default > Dialplan: sofia/internalopus/2223 at sip.example.com parsing [default->echo] > continue=false > Dialplan: sofia/internalopus/2223 at sip.example.com Regex (PASS) [echo] > destination_number(*71) =~ /^\*71$/ break=on-false > Dialplan: sofia/internalopus/2223 at sip.example.com Action answer() > Dialplan: sofia/internalopus/2223 at sip.example.com Action echo() > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_state_machine.c:286 ( > sofia/internalopus/2223 at sip.example.com) State Change CS_ROUTING -> > CS_EXECUTE > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_state_machine.c:643 ( > sofia/internalopus/2223 at sip.example.com) State ROUTING going to sleep > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_state_machine.c:584 ( > sofia/internalopus/2223 at sip.example.com) Running State Change CS_EXECUTE > (Cur 2 Tot 3196652) > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_state_machine.c:650 ( > sofia/internalopus/2223 at sip.example.com) State EXECUTE > 2017-09-25 22:01:06.357410 [DEBUG] mod_sofia.c:198 > sofia/internalopus/2223 at sip.example.com SOFIA EXECUTE > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_state_machine.c:328 > sofia/internalopus/2223 at sip.example.com Standard EXECUTE > EXECUTE sofia/internalopus/2223 at sip.example.com answer() > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec > Compare [opus:103:48000:20:0:1]/[opus:116:48000:20:0:1] > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4484 Audio Codec > Compare [opus:116:48000:20:0:1] ++++ is saved as a match > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec > Compare [opus:103:48000:20:0:1]/[PCMU:0:8000:20:64000:1] > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec > Compare [opus:103:48000:20:0:1]/[PCMA:8:8000:20:64000:1] > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec > Compare [PCMA:8:8000:20:64000:1]/[opus:116:48000:20:0:1] > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec > Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec > Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4484 Audio Codec > Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec > Compare [PCMU:0:8000:20:64000:1]/[opus:116:48000:20:0:1] > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec > Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4484 Audio Codec > Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec > Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec > Compare [G722:9:8000:20:64000:1]/[opus:116:48000:20:0:1] > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec > Compare [G722:9:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec > Compare [G722:9:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4345 Set > telephone-event payload to 101 at 8000 > 2017-09-25 22:01:06.357410 [DEBUG] mod_opus.c:596 Opus encoder: set > bitrate based on maxaveragebitrate value found in SDP or local config > [14400bps] > 2017-09-25 22:01:06.357410 [DEBUG] mod_opus.c:613 Opus encoder: set audio > bandwidth to [NARROWBAND] based on maxplaybackrate value found in SDP or > local config [8000Hz] > 2017-09-25 22:01:06.357410 [DEBUG] mod_opus.c:596 Opus encoder: set > bitrate based on maxaveragebitrate value found in SDP or local config > [14400bps] > 2017-09-25 22:01:06.357410 [DEBUG] mod_opus.c:613 Opus encoder: set audio > bandwidth to [NARROWBAND] based on maxplaybackrate value found in SDP or > local config [8000Hz] > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:3049 Set Codec > sofia/internalopus/2223 at sip.example.com opus/8000 20 ms 960 samples 0 > bits 1 channels > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_codec.c:111 > sofia/internalopus/2223 at sip.example.com Original read codec set to > opus:116 > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4747 > sofia/internalopus/2223 at sip.example.com Set 2833 dtmf send payload to 101 > recv payload to 101 > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:6803 AUDIO RTP [ > sofia/internalopus/2223 at sip.example.com] 192.168.100.12 port 23594 -> > 192.168.100.2 port 10036 codec: 103 ms: 20 > 2017-09-25 22:01:06.357410 [DEBUG] switch_rtp.c:4096 Starting timer [soft] > 960 bytes per 20ms > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:7109 > sofia/internalopus/2223 at sip.example.com Set 2833 dtmf send payload to 101 > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:7116 > sofia/internalopus/2223 at sip.example.com Set 2833 dtmf receive payload to > 101 > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:7139 > sofia/internalopus/2223 at sip.example.com Set rtp dtmf delay to 40 > 2017-09-25 22:01:06.357410 [NOTICE] sofia_media.c:92 Pre-Answer > sofia/internalopus/2223 at sip.example.com! > 2017-09-25 22:01:06.357410 [DEBUG] switch_channel.c:3473 ( > sofia/internalopus/2223 at sip.example.com) Callstate Change RINGING -> EARLY > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:6786 Audio params > are unchanged for sofia/internalopus/2223 at sip.example.com. > > 2017-09-25 22:01:06.357410 [DEBUG] mod_sofia.c:850 Local SDP > sofia/internalopus/2223 at sip.example.com: > v=0 > o=FreeSWITCH 1506342472 1506342473 IN IP4 192.168.100.12 > s=FreeSWITCH > c=IN IP4 192.168.100.12 > t=0 0 > m=audio 23594 RTP/AVP 103 101 > a=rtpmap:103 opus/48000/2 > a=fmtp:103 useinbandfec=1; usedtx=1; maxaveragebitrate=14400; > maxplaybackrate=8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > a=sendrecv > > 2017-09-25 22:01:06.357410 [NOTICE] mod_dptools.c:1312 Channel [ > sofia/internalopus/2223 at sip.example.com] has been answered > 2017-09-25 22:01:06.357410 [DEBUG] sofia.c:7048 Channel > sofia/internalopus/2223 at sip.example.com entering state [completed][200] > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media_bug.c:828 Attaching > BUG to sofia/internalopus/2223 at sip.example.com > 2017-09-25 22:01:06.357410 [DEBUG] switch_channel.c:3772 ( > sofia/internalopus/2223 at sip.example.com) Callstate Change EARLY -> ACTIVE > EXECUTE sofia/internalopus/2223 at sip.example.com echo() > 2017-09-25 22:01:06.737463 [DEBUG] sofia.c:7048 Channel > sofia/internalopus/2223 at sip.example.com entering state [ready][200] > 2017-09-25 22:01:06.837459 [DEBUG] switch_rtp.c:7229 Correct audio ip/port > confirmed. > 2017-09-25 22:01:06.837459 [DEBUG] switch_core_io.c:448 Setting BUG Codec > opus:116 > 2017-09-25 22:01:06.837459 [DEBUG] mod_opus.c:596 Opus encoder: set > bitrate based on maxaveragebitrate value found in SDP or local config > [14400bps] > 2017-09-25 22:01:06.837459 [DEBUG] mod_opus.c:613 Opus encoder: set audio > bandwidth to [NARROWBAND] based on maxplaybackrate value found in SDP or > local config [8000Hz] > 2017-09-25 22:01:06.837459 [DEBUG] switch_core_io.c:780 Engaging Read > Buffer at 320 bytes vs 15 > [...CREEPY VOICE STARTS HERE AND ENDS WITH THE BELOW MESSAGE...] > 2017-09-25 22:01:16.017463 [DEBUG] switch_core_media_bug.c:1198 Removing > BUG from sofia/internalopus/2223 at sip.example.com > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From voransoy at gmail.com Mon Sep 25 19:46:25 2017 From: voransoy at gmail.com (Volkan Oransoy) Date: Mon, 25 Sep 2017 22:46:25 +0300 Subject: [Freeswitch-users] Opus problem In-Reply-To: References: <95869C0B-F054-442D-8E5F-E788A5B439A6@gmail.com> Message-ID: Michael, Thank you for your reply. I have removed that directive and it is still the same. In fact I started with the example config at https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+And+The+Opus+Audio+Codec and I tried different directives like asymmetric sample rate which may effect this issue. Right now my current config is the same as the reference document, but the issue persists. /Volkan > On 25 Sep 2017, at 22:35, Michael Jerris wrote: > > This is due to an incomplete implementation of the asymmetric sample rate feature. It should work fine without that setting. > > On Mon, Sep 25, 2017 at 2:22 PM Volkan Oransoy > wrote: > Hi All, > > I am trying to transcode between OPUS and G711. I encountered a strange problem which I couldn’t find any clue about the cause. After the channel has been answered with opus, I hear a strange sound (highly distorted version of my voice) for 10 seconds and it starts to operate properly with the message "[DEBUG] switch_core_media_bug.c:1198 Removing BUG from…” . I can reproduce same problem with echo application. Here is my logs and opus.conf.xml files. Did anybody experienced this issue? > > Thanks. > > /Volkan > > > My FreeSWITH Version: > FreeSWITCH Version 1.6.16+git~20170407T164913Z~3da6bd0108~64bit (git 3da6bd0 2017-04-07 16:49:13Z 64bit) > > Opus.xml config: > > > > > > > > > > > > > > > > Freeswitch Log: > 2017-09-25 22:01:06.277462 [NOTICE] switch_channel.c:1104 New Channel sofia/internalopus/2223 at sip.example.com [1048c869-8d71-41fd-b0b7-f8fbd01a0d93] > 2017-09-25 22:01:06.277462 [DEBUG] switch_core_state_machine.c:584 (sofia/internalopus/2223 at sip.example.com ) Running State Change CS_NEW (Cur 2 Tot 3196652) > 2017-09-25 22:01:06.277462 [DEBUG] sofia.c:9828 sofia/internalopus/2223 at sip.example.com receiving invite from 192.168.100.2:5060 version: 1.6.16 git 3da6bd0 2017-04-07 16:49:13Z 64bit > 2017-09-25 22:01:06.277462 [DEBUG] sofia.c:7048 Channel sofia/internalopus/2223 at sip.example.com entering state [received][100] > 2017-09-25 22:01:06.277462 [DEBUG] sofia.c:7058 Remote SDP: > v=0 > o=- 8070049943 45777 IN IP4 192.168.100.2 > s=dvjtiwu > c=IN IP4 192.168.100.2 > t=0 0 > m=audio 10036 RTP/AVP 103 8 0 9 101 > a=rtpmap:103 opus/48000/2 > a=fmtp:103 maxplaybackrate=8000;maxaveragebitrate=15500;useinbandfec=1;usedtx=1 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=ptime:20 > a=nortpproxy:yes > . > . > . > . > . > 2017-09-25 22:01:06.337394 [INFO] mod_dialplan_xml.c:637 Processing 2223 <2223>->*71 in context default > Dialplan: sofia/internalopus/2223 at sip.example.com parsing [default->echo] continue=false > Dialplan: sofia/internalopus/2223 at sip.example.com Regex (PASS) [echo] destination_number(*71) =~ /^\*71$/ break=on-false > Dialplan: sofia/internalopus/2223 at sip.example.com Action answer() > Dialplan: sofia/internalopus/2223 at sip.example.com Action echo() > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_state_machine.c:286 (sofia/internalopus/2223 at sip.example.com ) State Change CS_ROUTING -> CS_EXECUTE > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_state_machine.c:643 (sofia/internalopus/2223 at sip.example.com ) State ROUTING going to sleep > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_state_machine.c:584 (sofia/internalopus/2223 at sip.example.com ) Running State Change CS_EXECUTE (Cur 2 Tot 3196652) > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_state_machine.c:650 (sofia/internalopus/2223 at sip.example.com ) State EXECUTE > 2017-09-25 22:01:06.357410 [DEBUG] mod_sofia.c:198 sofia/internalopus/2223 at sip.example.com SOFIA EXECUTE > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_state_machine.c:328 sofia/internalopus/2223 at sip.example.com Standard EXECUTE > EXECUTE sofia/internalopus/2223 at sip.example.com answer() > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [opus:103:48000:20:0:1]/[opus:116:48000:20:0:1] > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4484 Audio Codec Compare [opus:116:48000:20:0:1] ++++ is saved as a match > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [opus:103:48000:20:0:1]/[PCMU:0:8000:20:64000:1] > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [opus:103:48000:20:0:1]/[PCMA:8:8000:20:64000:1] > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[opus:116:48000:20:0:1] > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4484 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[opus:116:48000:20:0:1] > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4484 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [G722:9:8000:20:64000:1]/[opus:116:48000:20:0:1] > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4345 Set telephone-event payload to 101 at 8000 > 2017-09-25 22:01:06.357410 [DEBUG] mod_opus.c:596 Opus encoder: set bitrate based on maxaveragebitrate value found in SDP or local config [14400bps] > 2017-09-25 22:01:06.357410 [DEBUG] mod_opus.c:613 Opus encoder: set audio bandwidth to [NARROWBAND] based on maxplaybackrate value found in SDP or local config [8000Hz] > 2017-09-25 22:01:06.357410 [DEBUG] mod_opus.c:596 Opus encoder: set bitrate based on maxaveragebitrate value found in SDP or local config [14400bps] > 2017-09-25 22:01:06.357410 [DEBUG] mod_opus.c:613 Opus encoder: set audio bandwidth to [NARROWBAND] based on maxplaybackrate value found in SDP or local config [8000Hz] > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:3049 Set Codec sofia/internalopus/2223 at sip.example.com opus/8000 20 ms 960 samples 0 bits 1 channels > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_codec.c:111 sofia/internalopus/2223 at sip.example.com Original read codec set to opus:116 > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4747 sofia/internalopus/2223 at sip.example.com Set 2833 dtmf send payload to 101 recv payload to 101 > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:6803 AUDIO RTP [sofia/internalopus/2223 at sip.example.com ] 192.168.100.12 port 23594 -> 192.168.100.2 port 10036 codec: 103 ms: 20 > 2017-09-25 22:01:06.357410 [DEBUG] switch_rtp.c:4096 Starting timer [soft] 960 bytes per 20ms > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:7109 sofia/internalopus/2223 at sip.example.com Set 2833 dtmf send payload to 101 > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:7116 sofia/internalopus/2223 at sip.example.com Set 2833 dtmf receive payload to 101 > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:7139 sofia/internalopus/2223 at sip.example.com Set rtp dtmf delay to 40 > 2017-09-25 22:01:06.357410 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internalopus/2223 at sip.example.com ! > 2017-09-25 22:01:06.357410 [DEBUG] switch_channel.c:3473 (sofia/internalopus/2223 at sip.example.com ) Callstate Change RINGING -> EARLY > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:6786 Audio params are unchanged for sofia/internalopus/2223 at sip.example.com . > > 2017-09-25 22:01:06.357410 [DEBUG] mod_sofia.c:850 Local SDP sofia/internalopus/2223 at sip.example.com : > v=0 > o=FreeSWITCH 1506342472 1506342473 IN IP4 192.168.100.12 > s=FreeSWITCH > c=IN IP4 192.168.100.12 > t=0 0 > m=audio 23594 RTP/AVP 103 101 > a=rtpmap:103 opus/48000/2 > a=fmtp:103 useinbandfec=1; usedtx=1; maxaveragebitrate=14400; maxplaybackrate=8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > a=sendrecv > > 2017-09-25 22:01:06.357410 [NOTICE] mod_dptools.c:1312 Channel [sofia/internalopus/2223 at sip.example.com ] has been answered > 2017-09-25 22:01:06.357410 [DEBUG] sofia.c:7048 Channel sofia/internalopus/2223 at sip.example.com entering state [completed][200] > 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media_bug.c:828 Attaching BUG to sofia/internalopus/2223 at sip.example.com > 2017-09-25 22:01:06.357410 [DEBUG] switch_channel.c:3772 (sofia/internalopus/2223 at sip.example.com ) Callstate Change EARLY -> ACTIVE > EXECUTE sofia/internalopus/2223 at sip.example.com echo() > 2017-09-25 22:01:06.737463 [DEBUG] sofia.c:7048 Channel sofia/internalopus/2223 at sip.example.com entering state [ready][200] > 2017-09-25 22:01:06.837459 [DEBUG] switch_rtp.c:7229 Correct audio ip/port confirmed. > 2017-09-25 22:01:06.837459 [DEBUG] switch_core_io.c:448 Setting BUG Codec opus:116 > 2017-09-25 22:01:06.837459 [DEBUG] mod_opus.c:596 Opus encoder: set bitrate based on maxaveragebitrate value found in SDP or local config [14400bps] > 2017-09-25 22:01:06.837459 [DEBUG] mod_opus.c:613 Opus encoder: set audio bandwidth to [NARROWBAND] based on maxplaybackrate value found in SDP or local config [8000Hz] > 2017-09-25 22:01:06.837459 [DEBUG] switch_core_io.c:780 Engaging Read Buffer at 320 bytes vs 15 > [...CREEPY VOICE STARTS HERE AND ENDS WITH THE BELOW MESSAGE...] > 2017-09-25 22:01:16.017463 [DEBUG] switch_core_media_bug.c:1198 Removing BUG from sofia/internalopus/2223 at sip.example.com > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From josedavid at zennio.com Tue Sep 26 06:41:41 2017 From: josedavid at zennio.com (Jose David Jurado Alonso) Date: Tue, 26 Sep 2017 08:41:41 +0200 Subject: [Freeswitch-users] How can I use an external database to create new users? In-Reply-To: <0db247db-dc52-cb83-f818-ab54c633de8e@xbipin.com> References: <2b6527f0-b831-fb2b-4a8e-47b43cdb4c87@level5.de> <7b4a8ce4-0136-465c-af1b-e7c62ec1dbe6@Spark> <0db247db-dc52-cb83-f818-ab54c633de8e@xbipin.com> Message-ID: Hi Bipin, I found this tutorial searching about the comment of other response: https://github.com/intralanman/fs_curl (It's looks fine but I think that are too complex) I just look for the way to force FS to load and register users (previously created) from a database. The dialplan and others configuration must be used as usual (from XML configuration files). Can you explain me in more detail your solution and pass me the register script? Thanks! José David Jurado Alonso *Área de Desarrollo de Software de Alto Nivel* *Dpto. Ingeniería* [image: Descripción: cid:image010.jpg at 01D27631.DF7E30F0] Zennio Avance y Tecnología, S.L. C/ Rio Jarama,132. Nave P-8.11 45007 - Toledo (Spain) T: +34 925 232 002 www.zennio.com [image: Descripción: Descripción: C:\Users\jjmanjarres\Desktop\Z zennio.jpg] [image: Descripción: Descripción: C:\Users\jjmanjarres\Desktop\twitter.jpg] [image: Descripción: Descripción: C:\Users\jjmanjarres\Desktop\descarga.jpg] [image: Descripción: Descripción: C:\Users\jjmanjarres\Desktop\linkedin.png] [image: Descripción: Descripción: C:\Users\jjmanjarres\Desktop\youtube.jpg] Zennio Avance y Tecnología S.L le informa de los siguientes extremos: Los datos por usted suministrados pasarán a formar parte de un fichero cuyo responsable es Zennio Avance y Tecnología S.L. Dicho fichero se encuentra legalmente inscrito en el Registro General de Protección de Datos de la Agencia Española de Protección de Datos. 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Advertencia legal: Este mensaje y, en su caso, los ficheros anexos son confidenciales, especialmente en lo que respecta a los datos personales, y se dirigen exclusivamente al destinatario referenciado. Si usted no lo es y lo ha recibido por error o tiene conocimiento del mismo por cualquier motivo, le rogamos que nos lo comunique por este medio y proceda a destruirlo o borrarlo, y que en todo caso se abstenga de utilizar, reproducir, alterar, archivar o comunicar a terceros el presente mensaje y ficheros anexos, todo ello bajo pena de incurrir en responsabilidades legales. Las opiniones contenidas en este mensaje y en los archivos adjuntos, pertenecen exclusivamente a su remitente y no representan la opinión de la empresa salvo que se diga expresamente y el remitente esté autorizado para ello. El emisor no garantiza la integridad, rapidez o seguridad del presente correo, ni se responsabiliza de posibles perjuicios derivados de la captura, incorporaciones de virus o cualesquiera otras manipulaciones efectuadas por terceros. 2017-09-25 15:44 GMT+02:00 Bipin Patel : > hi, > > we use a different switch with its db running on mysql but sadly that > doesnt supports tls and srtp so we use freeswitch for that so users > register to FS and we use xml curl for directory users where FS calls a php > script which inturn fetches the id/pass of the user and returns it to FS > which then allows users to register to FS. The script is only called when > users register to FS, its quiet simple if u ask me. > > > Regards, > Bipin > > > ------------------------------ > -------- Original Message -------- > Subject: Re: [Freeswitch-users] How can I use an external database to > create new users? > From: Igor Olhovskiy > To: Thorsten Göllner > , FreeSWITCH Users Help > > > Date: 9/25/2017, 5:05:45 PM > > mod_xml_curl will ask some http server with username and domain and will > require as a response XML-formatted directory entry. > So you need to write some intermediate layer, that will on request go to > database and provide back info in necessary format. > > Regards, Igor > > On 25 сент. 2017 г., 14:22 +0300, Jose David Jurado Alonso > , wrote: > > Hi, > > Yes, but I need a custom web app to users management. > > I tried using postgresql as core-db-dsn and I work properly in some cases > (register appears) but no show any about users. > > José David Jurado Alonso > > *Área de Desarrollo de Software de Alto Nivel* > > *Dpto. Ingeniería* > > [image: Descripción: cid:image010.jpg at 01D27631.DF7E30F0] > > Zennio Avance y Tecnología, S.L. > > C/ Rio Jarama,132. Nave P-8.11 > > 45007 - Toledo (Spain) > > T: +34 925 232 002 <+34%20925%2023%2020%2002> > > www.zennio.com > > [image: Descripción: Descripción: C:\Users\jjmanjarres\Desktop\Z > zennio.jpg] [image: Descripción: Descripción: > C:\Users\jjmanjarres\Desktop\twitter.jpg] > [image: Descripción: Descripción: > C:\Users\jjmanjarres\Desktop\descarga.jpg] > [image: Descripción: > Descripción: C:\Users\jjmanjarres\Desktop\linkedin.png] > [image: Descripción: > Descripción: C:\Users\jjmanjarres\Desktop\youtube.jpg] > > > Zennio Avance y Tecnología S.L le informa de los siguientes extremos: > Los datos por usted suministrados pasarán a formar parte de un fichero > cuyo responsable es Zennio Avance y Tecnología S.L. Dicho fichero se > encuentra legalmente inscrito en el Registro General de Protección de Datos > de la Agencia Española de Protección de Datos. Los datos por usted > suministrados serán empleados con fines de gestión, Zennio Avance y > Tecnología S.L ha adoptado las medidas de seguridad exigidas en función del > nivel de los datos suministrados, instalando las medidas técnicas y > organizativas necesarias, habida cuenta del estado de la tecnología, a fin > de evitar su pérdida, alteración, uso inadecuado o accesos no autorizados a > los mismos. > Para el ejercicio de sus derechos de acceso, rectificación, cancelación y > oposición deberá dirigirse a la dirección del Responsable de Fichero Zennio > Avance y Tecnología S.L C/ Rio Jarama, 132. Nave P-8.11, 45007, TOLEDO o a > la dirección de correo electrónico: info at zennio.com > > Please, consider the environment before printing this e-mail... Save > energy! > > Por favor, piensa en el medio ambiente antes de imprimir este e-mail... ¡Ahorra > energía! > > Disclaimer: > This message and any attached files transmitted with it, is confidential, > especially as regards personal data. It is intended solely for the use of > the individual or entity to whom it is addressed. If you are not the > intended recipient and have received this information in error or have > accessed it for any reason, please notify us of this fact by email reply > and then destroy or delete the message, refraining from any reproduction, > use, alteration, filing or communication to third parties of this message > and attached files on penalty of incurring legal responsibilities. The > opinions contained in this message and the attached archives, belong > exclusively to their sender and they do not represent the opinion of the > company unless it is said specifically and the sender is authorized for it. > The sender does not guarantee the integrity, the accuracy, the swift > delivery or the security of this email transmission, and assumes no > responsibility for any possible damage incurred through data capture, virus > incorporation or any manipulation carried out by third parties. > > Advertencia legal: > Este mensaje y, en su caso, los ficheros anexos son confidenciales, > especialmente en lo que respecta a los datos personales, y se dirigen > exclusivamente al destinatario referenciado. Si usted no lo es y lo ha > recibido por error o tiene conocimiento del mismo por cualquier motivo, le > rogamos que nos lo comunique por este medio y proceda a destruirlo o > borrarlo, y que en todo caso se abstenga de utilizar, reproducir, alterar, > archivar o comunicar a terceros el presente mensaje y ficheros anexos, todo > ello bajo pena de incurrir en responsabilidades legales. Las opiniones > contenidas en este mensaje y en los archivos adjuntos, pertenecen > exclusivamente a su remitente y no representan la opinión de la empresa > salvo que se diga expresamente y el remitente esté autorizado para ello. El > emisor no garantiza la integridad, rapidez o seguridad del presente correo, > ni se responsabiliza de posibles perjuicios derivados de la captura, > incorporaciones de virus o cualesquiera otras manipulaciones efectuadas por > terceros. > > 2017-09-25 12:44 GMT+02:00 Thorsten Göllner : > >> Hi, >> >> did you take a look at existing (Web)Guis like FusionPBX? >> >> https://www.fusionpbx.com/ >> >> Thorsten >> >> >> Am 25.09.2017 um 10:28 schrieb Jose David Jurado Alonso: >> >> Hi, >> >> I'm thinking to create a web application to create, edit and delete users >> in a simpler way than creating XML by hand but I don't find information >> about it. >> >> I've seen something about the module "mod_xml_curl" but I'm not sure how >> it interacts with the database nor if it's necessary some additional module >> more. >> >> I would like to use MySQL or postgreSQL, does anyone know how you can do >> it or some tutorial or reference application? >> >> Thanks, >> >> José D. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >> >> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com >> >> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... 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Name: 1D0CFFE6F50B4129BE6BF73A214643FC.png Type: image/png Size: 2374 bytes Desc: not available URL: From voransoy at gmail.com Tue Sep 26 08:18:30 2017 From: voransoy at gmail.com (Volkan Oransoy) Date: Tue, 26 Sep 2017 11:18:30 +0300 Subject: [Freeswitch-users] Opus problem In-Reply-To: References: <95869C0B-F054-442D-8E5F-E788A5B439A6@gmail.com> Message-ID: <812F987B-4153-4FD7-B152-AD2161CD88DD@gmail.com> Hi, I have solved the issue. It is caused by the variable I set before execute_on_answer=spandsp_start_fax_detect transfer '*11 XML default’ 10 Somehow the bug placed by this directive effects the decoder when the codec is opus. When 10 seconds expire, it turns back to normal. Thanks, /Volkan > On 25 Sep 2017, at 22:46, Volkan Oransoy wrote: > > Michael, > > Thank you for your reply. I have removed that directive and it is still the same. > > In fact I started with the example config at https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+And+The+Opus+Audio+Codec and I tried different directives like asymmetric sample rate which may effect this issue. Right now my current config is the same as the reference document, but the issue persists. > > > > > > > > > > > > > > > > /Volkan > > > >> On 25 Sep 2017, at 22:35, Michael Jerris > wrote: >> >> This is due to an incomplete implementation of the asymmetric sample rate feature. It should work fine without that setting. >> >> On Mon, Sep 25, 2017 at 2:22 PM Volkan Oransoy > wrote: >> Hi All, >> >> I am trying to transcode between OPUS and G711. I encountered a strange problem which I couldn’t find any clue about the cause. After the channel has been answered with opus, I hear a strange sound (highly distorted version of my voice) for 10 seconds and it starts to operate properly with the message "[DEBUG] switch_core_media_bug.c:1198 Removing BUG from…” . I can reproduce same problem with echo application. Here is my logs and opus.conf.xml files. Did anybody experienced this issue? >> >> Thanks. >> >> /Volkan >> >> >> My FreeSWITH Version: >> FreeSWITCH Version 1.6.16+git~20170407T164913Z~3da6bd0108~64bit (git 3da6bd0 2017-04-07 16:49:13Z 64bit) >> >> Opus.xml config: >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> Freeswitch Log: >> 2017-09-25 22:01:06.277462 [NOTICE] switch_channel.c:1104 New Channel sofia/internalopus/2223 at sip.example.com [1048c869-8d71-41fd-b0b7-f8fbd01a0d93] >> 2017-09-25 22:01:06.277462 [DEBUG] switch_core_state_machine.c:584 (sofia/internalopus/2223 at sip.example.com ) Running State Change CS_NEW (Cur 2 Tot 3196652) >> 2017-09-25 22:01:06.277462 [DEBUG] sofia.c:9828 sofia/internalopus/2223 at sip.example.com receiving invite from 192.168.100.2:5060 version: 1.6.16 git 3da6bd0 2017-04-07 16:49:13Z 64bit >> 2017-09-25 22:01:06.277462 [DEBUG] sofia.c:7048 Channel sofia/internalopus/2223 at sip.example.com entering state [received][100] >> 2017-09-25 22:01:06.277462 [DEBUG] sofia.c:7058 Remote SDP: >> v=0 >> o=- 8070049943 45777 IN IP4 192.168.100.2 >> s=dvjtiwu >> c=IN IP4 192.168.100.2 >> t=0 0 >> m=audio 10036 RTP/AVP 103 8 0 9 101 >> a=rtpmap:103 opus/48000/2 >> a=fmtp:103 maxplaybackrate=8000;maxaveragebitrate=15500;useinbandfec=1;usedtx=1 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> a=ptime:20 >> a=nortpproxy:yes >> . >> . >> . >> . >> . >> 2017-09-25 22:01:06.337394 [INFO] mod_dialplan_xml.c:637 Processing 2223 <2223>->*71 in context default >> Dialplan: sofia/internalopus/2223 at sip.example.com parsing [default->echo] continue=false >> Dialplan: sofia/internalopus/2223 at sip.example.com Regex (PASS) [echo] destination_number(*71) =~ /^\*71$/ break=on-false >> Dialplan: sofia/internalopus/2223 at sip.example.com Action answer() >> Dialplan: sofia/internalopus/2223 at sip.example.com Action echo() >> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_state_machine.c:286 (sofia/internalopus/2223 at sip.example.com ) State Change CS_ROUTING -> CS_EXECUTE >> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_state_machine.c:643 (sofia/internalopus/2223 at sip.example.com ) State ROUTING going to sleep >> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_state_machine.c:584 (sofia/internalopus/2223 at sip.example.com ) Running State Change CS_EXECUTE (Cur 2 Tot 3196652) >> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_state_machine.c:650 (sofia/internalopus/2223 at sip.example.com ) State EXECUTE >> 2017-09-25 22:01:06.357410 [DEBUG] mod_sofia.c:198 sofia/internalopus/2223 at sip.example.com SOFIA EXECUTE >> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_state_machine.c:328 sofia/internalopus/2223 at sip.example.com Standard EXECUTE >> EXECUTE sofia/internalopus/2223 at sip.example.com answer() >> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [opus:103:48000:20:0:1]/[opus:116:48000:20:0:1] >> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4484 Audio Codec Compare [opus:116:48000:20:0:1] ++++ is saved as a match >> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [opus:103:48000:20:0:1]/[PCMU:0:8000:20:64000:1] >> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [opus:103:48000:20:0:1]/[PCMA:8:8000:20:64000:1] >> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[opus:116:48000:20:0:1] >> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] >> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] >> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4484 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match >> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[opus:116:48000:20:0:1] >> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] >> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4484 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match >> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] >> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [G722:9:8000:20:64000:1]/[opus:116:48000:20:0:1] >> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] >> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] >> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4345 Set telephone-event payload to 101 at 8000 >> 2017-09-25 22:01:06.357410 [DEBUG] mod_opus.c:596 Opus encoder: set bitrate based on maxaveragebitrate value found in SDP or local config [14400bps] >> 2017-09-25 22:01:06.357410 [DEBUG] mod_opus.c:613 Opus encoder: set audio bandwidth to [NARROWBAND] based on maxplaybackrate value found in SDP or local config [8000Hz] >> 2017-09-25 22:01:06.357410 [DEBUG] mod_opus.c:596 Opus encoder: set bitrate based on maxaveragebitrate value found in SDP or local config [14400bps] >> 2017-09-25 22:01:06.357410 [DEBUG] mod_opus.c:613 Opus encoder: set audio bandwidth to [NARROWBAND] based on maxplaybackrate value found in SDP or local config [8000Hz] >> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:3049 Set Codec sofia/internalopus/2223 at sip.example.com opus/8000 20 ms 960 samples 0 bits 1 channels >> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_codec.c:111 sofia/internalopus/2223 at sip.example.com Original read codec set to opus:116 >> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4747 sofia/internalopus/2223 at sip.example.com Set 2833 dtmf send payload to 101 recv payload to 101 >> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:6803 AUDIO RTP [sofia/internalopus/2223 at sip.example.com ] 192.168.100.12 port 23594 -> 192.168.100.2 port 10036 codec: 103 ms: 20 >> 2017-09-25 22:01:06.357410 [DEBUG] switch_rtp.c:4096 Starting timer [soft] 960 bytes per 20ms >> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:7109 sofia/internalopus/2223 at sip.example.com Set 2833 dtmf send payload to 101 >> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:7116 sofia/internalopus/2223 at sip.example.com Set 2833 dtmf receive payload to 101 >> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:7139 sofia/internalopus/2223 at sip.example.com Set rtp dtmf delay to 40 >> 2017-09-25 22:01:06.357410 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internalopus/2223 at sip.example.com ! >> 2017-09-25 22:01:06.357410 [DEBUG] switch_channel.c:3473 (sofia/internalopus/2223 at sip.example.com ) Callstate Change RINGING -> EARLY >> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:6786 Audio params are unchanged for sofia/internalopus/2223 at sip.example.com . >> >> 2017-09-25 22:01:06.357410 [DEBUG] mod_sofia.c:850 Local SDP sofia/internalopus/2223 at sip.example.com : >> v=0 >> o=FreeSWITCH 1506342472 1506342473 IN IP4 192.168.100.12 >> s=FreeSWITCH >> c=IN IP4 192.168.100.12 >> t=0 0 >> m=audio 23594 RTP/AVP 103 101 >> a=rtpmap:103 opus/48000/2 >> a=fmtp:103 useinbandfec=1; usedtx=1; maxaveragebitrate=14400; maxplaybackrate=8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=ptime:20 >> a=sendrecv >> >> 2017-09-25 22:01:06.357410 [NOTICE] mod_dptools.c:1312 Channel [sofia/internalopus/2223 at sip.example.com ] has been answered >> 2017-09-25 22:01:06.357410 [DEBUG] sofia.c:7048 Channel sofia/internalopus/2223 at sip.example.com entering state [completed][200] >> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media_bug.c:828 Attaching BUG to sofia/internalopus/2223 at sip.example.com >> 2017-09-25 22:01:06.357410 [DEBUG] switch_channel.c:3772 (sofia/internalopus/2223 at sip.example.com ) Callstate Change EARLY -> ACTIVE >> EXECUTE sofia/internalopus/2223 at sip.example.com echo() >> 2017-09-25 22:01:06.737463 [DEBUG] sofia.c:7048 Channel sofia/internalopus/2223 at sip.example.com entering state [ready][200] >> 2017-09-25 22:01:06.837459 [DEBUG] switch_rtp.c:7229 Correct audio ip/port confirmed. >> 2017-09-25 22:01:06.837459 [DEBUG] switch_core_io.c:448 Setting BUG Codec opus:116 >> 2017-09-25 22:01:06.837459 [DEBUG] mod_opus.c:596 Opus encoder: set bitrate based on maxaveragebitrate value found in SDP or local config [14400bps] >> 2017-09-25 22:01:06.837459 [DEBUG] mod_opus.c:613 Opus encoder: set audio bandwidth to [NARROWBAND] based on maxplaybackrate value found in SDP or local config [8000Hz] >> 2017-09-25 22:01:06.837459 [DEBUG] switch_core_io.c:780 Engaging Read Buffer at 320 bytes vs 15 >> [...CREEPY VOICE STARTS HERE AND ENDS WITH THE BELOW MESSAGE...] >> 2017-09-25 22:01:16.017463 [DEBUG] switch_core_media_bug.c:1198 Removing BUG from sofia/internalopus/2223 at sip.example.com >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From josedavid at zennio.com Wed Sep 27 12:22:27 2017 From: josedavid at zennio.com (Jose David Jurado Alonso) Date: Wed, 27 Sep 2017 14:22:27 +0200 Subject: [Freeswitch-users] Only call the last registered account when the same account is on multiple devices Message-ID: Hi, I have the same SIP account (1008) registered from several mobile SIP clients and when I call from another number (1000) the call only arrive to one of the SIP clients and not to all. How can I make the call to all SIP clients? I need this: The call came to all clients and when one response (pick up the call), it was cut into the other clients. -------------- next part -------------- An HTML attachment was scrubbed... URL: From josedavid at zennio.com Wed Sep 27 12:37:40 2017 From: josedavid at zennio.com (Jose David Jurado Alonso) Date: Wed, 27 Sep 2017 14:37:40 +0200 Subject: [Freeswitch-users] Only call the last registered account when the same account is on multiple devices In-Reply-To: References: Message-ID: More info. I tried this but no result: Multiple Registrations Call one extension and ring several phones You must enable multiple registrations in *conf/sip_profiles/internal-ipv6.xml* and *conf/sip_profiles/internal.xml* (enabling the setting in conf/autoload_configs/switch.conf.xml had no effect). Valid values for this parameter are "contact", "true", "false". value="true" is the most common use. Setting this value to "contact" will remove the old registration based on sip_user, sip_host and contact field as opposed to the call_id. José David Jurado Alonso *Área de Desarrollo de Software de Alto Nivel* *Dpto. Ingeniería* [image: Descripción: cid:image010.jpg at 01D27631.DF7E30F0] Zennio Avance y Tecnología, S.L. C/ Rio Jarama,132. Nave P-8.11 45007 - Toledo (Spain) T: +34 925 232 002 www.zennio.com [image: Descripción: Descripción: C:\Users\jjmanjarres\Desktop\Z zennio.jpg] [image: Descripción: Descripción: C:\Users\jjmanjarres\Desktop\twitter.jpg] [image: Descripción: Descripción: C:\Users\jjmanjarres\Desktop\descarga.jpg] [image: Descripción: Descripción: C:\Users\jjmanjarres\Desktop\linkedin.png] [image: Descripción: Descripción: C:\Users\jjmanjarres\Desktop\youtube.jpg] Zennio Avance y Tecnología S.L le informa de los siguientes extremos: Los datos por usted suministrados pasarán a formar parte de un fichero cuyo responsable es Zennio Avance y Tecnología S.L. Dicho fichero se encuentra legalmente inscrito en el Registro General de Protección de Datos de la Agencia Española de Protección de Datos. Los datos por usted suministrados serán empleados con fines de gestión, Zennio Avance y Tecnología S.L ha adoptado las medidas de seguridad exigidas en función del nivel de los datos suministrados, instalando las medidas técnicas y organizativas necesarias, habida cuenta del estado de la tecnología, a fin de evitar su pérdida, alteración, uso inadecuado o accesos no autorizados a los mismos. Para el ejercicio de sus derechos de acceso, rectificación, cancelación y oposición deberá dirigirse a la dirección del Responsable de Fichero Zennio Avance y Tecnología S.L C/ Rio Jarama, 132. Nave P-8.11, 45007, TOLEDO o a la dirección de correo electrónico: info at zennio.com Please, consider the environment before printing this e-mail... Save energy! Por favor, piensa en el medio ambiente antes de imprimir este e-mail... ¡Ahorra energía! Disclaimer: This message and any attached files transmitted with it, is confidential, especially as regards personal data. It is intended solely for the use of the individual or entity to whom it is addressed. If you are not the intended recipient and have received this information in error or have accessed it for any reason, please notify us of this fact by email reply and then destroy or delete the message, refraining from any reproduction, use, alteration, filing or communication to third parties of this message and attached files on penalty of incurring legal responsibilities. The opinions contained in this message and the attached archives, belong exclusively to their sender and they do not represent the opinion of the company unless it is said specifically and the sender is authorized for it. The sender does not guarantee the integrity, the accuracy, the swift delivery or the security of this email transmission, and assumes no responsibility for any possible damage incurred through data capture, virus incorporation or any manipulation carried out by third parties. Advertencia legal: Este mensaje y, en su caso, los ficheros anexos son confidenciales, especialmente en lo que respecta a los datos personales, y se dirigen exclusivamente al destinatario referenciado. Si usted no lo es y lo ha recibido por error o tiene conocimiento del mismo por cualquier motivo, le rogamos que nos lo comunique por este medio y proceda a destruirlo o borrarlo, y que en todo caso se abstenga de utilizar, reproducir, alterar, archivar o comunicar a terceros el presente mensaje y ficheros anexos, todo ello bajo pena de incurrir en responsabilidades legales. Las opiniones contenidas en este mensaje y en los archivos adjuntos, pertenecen exclusivamente a su remitente y no representan la opinión de la empresa salvo que se diga expresamente y el remitente esté autorizado para ello. El emisor no garantiza la integridad, rapidez o seguridad del presente correo, ni se responsabiliza de posibles perjuicios derivados de la captura, incorporaciones de virus o cualesquiera otras manipulaciones efectuadas por terceros. 2017-09-27 14:22 GMT+02:00 Jose David Jurado Alonso : > Hi, > > I have the same SIP account (1008) registered from several mobile SIP > clients and when I call from another number (1000) the call only arrive to > one of the SIP clients and not to all. > > How can I make the call to all SIP clients? > > I need this: The call came to all clients and when one response (pick up > the call), it was cut into the other clients. > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image005.png Type: image/png Size: 1196 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... 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Name: image004.jpg Type: image/jpeg Size: 966 bytes Desc: not available URL: From josedavid at zennio.com Wed Sep 27 13:01:59 2017 From: josedavid at zennio.com (Jose David Jurado Alonso) Date: Wed, 27 Sep 2017 15:01:59 +0200 Subject: [Freeswitch-users] Only call the last registered account when the same account is on multiple devices In-Reply-To: References: Message-ID: I just solved the problem, adding the param "multiple-registers" set to true to the externals sip_profiles the call arrives to all SIP clients. /etc/freeswitch/autoload_configs/switch.conf.xml: /etc/freeswitch/sip_profiles/external-ipv6.xml: /etc/freeswitch/sip_profiles/internal-ipv6.xml: /etc/freeswitch/sip_profiles/internal.xml: /etc/freeswitch/sip_profiles/external.xml: José David Jurado Alonso *Área de Desarrollo de Software de Alto Nivel* *Dpto. Ingeniería* [image: Descripción: cid:image010.jpg at 01D27631.DF7E30F0] Zennio Avance y Tecnología, S.L. C/ Rio Jarama,132. Nave P-8.11 45007 - Toledo (Spain) T: +34 925 232 002 www.zennio.com [image: Descripción: Descripción: C:\Users\jjmanjarres\Desktop\Z zennio.jpg] [image: Descripción: Descripción: C:\Users\jjmanjarres\Desktop\twitter.jpg] [image: Descripción: Descripción: C:\Users\jjmanjarres\Desktop\descarga.jpg] [image: Descripción: Descripción: C:\Users\jjmanjarres\Desktop\linkedin.png] [image: Descripción: Descripción: C:\Users\jjmanjarres\Desktop\youtube.jpg] Zennio Avance y Tecnología S.L le informa de los siguientes extremos: Los datos por usted suministrados pasarán a formar parte de un fichero cuyo responsable es Zennio Avance y Tecnología S.L. Dicho fichero se encuentra legalmente inscrito en el Registro General de Protección de Datos de la Agencia Española de Protección de Datos. Los datos por usted suministrados serán empleados con fines de gestión, Zennio Avance y Tecnología S.L ha adoptado las medidas de seguridad exigidas en función del nivel de los datos suministrados, instalando las medidas técnicas y organizativas necesarias, habida cuenta del estado de la tecnología, a fin de evitar su pérdida, alteración, uso inadecuado o accesos no autorizados a los mismos. Para el ejercicio de sus derechos de acceso, rectificación, cancelación y oposición deberá dirigirse a la dirección del Responsable de Fichero Zennio Avance y Tecnología S.L C/ Rio Jarama, 132. Nave P-8.11, 45007, TOLEDO o a la dirección de correo electrónico: info at zennio.com Please, consider the environment before printing this e-mail... Save energy! Por favor, piensa en el medio ambiente antes de imprimir este e-mail... ¡Ahorra energía! Disclaimer: This message and any attached files transmitted with it, is confidential, especially as regards personal data. It is intended solely for the use of the individual or entity to whom it is addressed. If you are not the intended recipient and have received this information in error or have accessed it for any reason, please notify us of this fact by email reply and then destroy or delete the message, refraining from any reproduction, use, alteration, filing or communication to third parties of this message and attached files on penalty of incurring legal responsibilities. The opinions contained in this message and the attached archives, belong exclusively to their sender and they do not represent the opinion of the company unless it is said specifically and the sender is authorized for it. The sender does not guarantee the integrity, the accuracy, the swift delivery or the security of this email transmission, and assumes no responsibility for any possible damage incurred through data capture, virus incorporation or any manipulation carried out by third parties. Advertencia legal: Este mensaje y, en su caso, los ficheros anexos son confidenciales, especialmente en lo que respecta a los datos personales, y se dirigen exclusivamente al destinatario referenciado. Si usted no lo es y lo ha recibido por error o tiene conocimiento del mismo por cualquier motivo, le rogamos que nos lo comunique por este medio y proceda a destruirlo o borrarlo, y que en todo caso se abstenga de utilizar, reproducir, alterar, archivar o comunicar a terceros el presente mensaje y ficheros anexos, todo ello bajo pena de incurrir en responsabilidades legales. Las opiniones contenidas en este mensaje y en los archivos adjuntos, pertenecen exclusivamente a su remitente y no representan la opinión de la empresa salvo que se diga expresamente y el remitente esté autorizado para ello. El emisor no garantiza la integridad, rapidez o seguridad del presente correo, ni se responsabiliza de posibles perjuicios derivados de la captura, incorporaciones de virus o cualesquiera otras manipulaciones efectuadas por terceros. 2017-09-27 14:22 GMT+02:00 Jose David Jurado Alonso : > Hi, > > I have the same SIP account (1008) registered from several mobile SIP > clients and when I call from another number (1000) the call only arrive to > one of the SIP clients and not to all. > > How can I make the call to all SIP clients? > > I need this: The call came to all clients and when one response (pick up > the call), it was cut into the other clients. > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image002.jpg Type: image/jpeg Size: 1024 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... 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Name: image005.png Type: image/png Size: 1196 bytes Desc: not available URL: From bipin at xbipin.com Wed Sep 27 13:13:02 2017 From: bipin at xbipin.com (Bipin Patel) Date: Wed, 27 Sep 2017 17:13:02 +0400 Subject: [Freeswitch-users] Only call the last registered account when the same account is on multiple devices In-Reply-To: References: Message-ID: <09e6f7da-184a-edcd-5ebd-d412e8c20b51@xbipin.com> hi, u can use sofia_contact in the bridge statement then all the phones will ring that are registered with same id Regards, Bipin ------------------------------------------------------------------------ -------- Original Message -------- Subject: Re: [Freeswitch-users] Only call the last registered account when the same account is on multiple devices From: Jose David Jurado Alonso To: FreeSWITCH Users Help Date: 9/27/2017, 5:01:59 PM > I just solved the problem, adding the param "multiple-registers" set > to true to the externals sip_profiles the call arrives to all SIP > clients. > > > /etc/freeswitch/autoload_configs/switch.conf.xml: name="multiple-registrations" value="true"/> > /etc/freeswitch/sip_profiles/external-ipv6.xml: name="multiple-registrations" value="true"/> > /etc/freeswitch/sip_profiles/internal-ipv6.xml:   name="multiple-registrations" value="true"/> > /etc/freeswitch/sip_profiles/internal.xml:       name="multiple-registrations" value="true"/> > /etc/freeswitch/sip_profiles/external.xml:     name="multiple-registrations" value="true"/> > > > José David Jurado Alonso > > > /Área de Desarrollo de Software de Alto Nivel/ > > *Dpto. Ingeniería* > > Descripción: cid:image010.jpg at 01D27631.DF7E30F0 > > Zennio Avance y Tecnología, S.L. > > C/ Rio Jarama,132. Nave P-8.11 > > 45007 - Toledo (Spain) > > T: +34 925 232 002 > > www.zennio.com > > Descripción: Descripción: C:\Users\jjmanjarres\Desktop\Z zennio.jpg > Descripción: Descripción: > C:\Users\jjmanjarres\Desktop\twitter.jpg > Descripción: Descripción: > C:\Users\jjmanjarres\Desktop\descarga.jpg > Descripción: Descripción: > C:\Users\jjmanjarres\Desktop\linkedin.png > Descripción: Descripción: > C:\Users\jjmanjarres\Desktop\youtube.jpg > > > Zennio Avance y Tecnología S.L le informa de los siguientes extremos: > Los datos por usted suministrados pasarán a formar parte de un fichero > cuyo responsable es Zennio Avance y Tecnología S.L. Dicho fichero se > encuentra legalmente inscrito en el Registro General de Protección de > Datos de la Agencia Española de Protección de Datos. Los datos por > usted suministrados serán empleados con fines de gestión, Zennio > Avance y Tecnología S.L ha adoptado las medidas de seguridad exigidas > en función del nivel de los datos suministrados, instalando las > medidas técnicas y organizativas necesarias, habida cuenta del estado > de la tecnología, a fin de evitar su pérdida, alteración, uso > inadecuado o accesos no autorizados a los mismos. > Para el ejercicio de sus derechos de acceso, rectificación, > cancelación y oposición deberá dirigirse a la dirección del > Responsable de Fichero Zennio Avance y Tecnología S.L C/ Rio Jarama, > 132. Nave P-8.11, 45007, TOLEDO o a la dirección de correo > electrónico: info at zennio.com > > Please, consider the environment before printing this e-mail... Save > energy! > > Por favor, piensa en el medio ambiente antes de imprimir este > e-mail... ¡Ahorra energía! > > Disclaimer: > This message and any attached files transmitted with it, is > confidential, especially as regards personal data. It is intended > solely for the use of the individual or entity to whom it is > addressed. If you are not the intended recipient and have received > this information in error or have accessed it for any reason, please > notify us of this fact by email reply and then destroy or delete the > message, refraining from any reproduction, use, alteration, filing or > communication to third parties of this message and attached files on > penalty of incurring legal responsibilities. The opinions contained in > this message and the attached archives, belong exclusively to their > sender and they do not represent the opinion of the company unless it > is said specifically and the sender is authorized for it. The sender > does not guarantee the integrity, the accuracy, the swift delivery or > the security of this email transmission, and assumes no responsibility > for any possible damage incurred through data capture, virus > incorporation or any manipulation carried out by third parties. > > Advertencia legal: > Este mensaje y, en su caso, los ficheros anexos son confidenciales, > especialmente en lo que respecta a los datos personales, y se dirigen > exclusivamente al destinatario referenciado. Si usted no lo es y lo ha > recibido por error o tiene conocimiento del mismo por cualquier > motivo, le rogamos que nos lo comunique por este medio y proceda a > destruirlo o borrarlo, y que en todo caso se abstenga de utilizar, > reproducir, alterar, archivar o comunicar a terceros el presente > mensaje y ficheros anexos, todo ello bajo pena de incurrir en > responsabilidades legales. Las opiniones contenidas en este mensaje y > en los archivos adjuntos, pertenecen exclusivamente a su remitente y > no representan la opinión de la empresa salvo que se diga expresamente > y el remitente esté autorizado para ello. El emisor no garantiza la > integridad, rapidez o seguridad del presente correo, ni se > responsabiliza de posibles perjuicios derivados de la captura, > incorporaciones de virus o cualesquiera otras manipulaciones > efectuadas por terceros. > > > 2017-09-27 14:22 GMT+02:00 Jose David Jurado Alonso > >: > > Hi, > > I have the same SIP account (1008) registered from several mobile > SIP clients and when I call from another number (1000) the call > only arrive to one of the SIP clients and not to all. > > How can I make the call to all SIP clients? > > I need this: The call came to all clients and when one response > (pick up the call), it was cut into the other clients. > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... 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Name: image006.jpg Type: image/jpeg Size: 1135 bytes Desc: not available URL: From bipin at xbipin.com Wed Sep 27 13:14:36 2017 From: bipin at xbipin.com (Bipin Patel) Date: Wed, 27 Sep 2017 17:14:36 +0400 Subject: [Freeswitch-users] cli command to list profiles Message-ID: <3f09f7fd-35b7-cfa8-ba9d-78994d05dfb8@xbipin.com> hi, is there any cli command to list names of all sip profiles as at times clients use different names other than internal and external -- Regards, Bipin ------------------------------------------------------------------------ -------------- next part -------------- An HTML attachment was scrubbed... URL: From vma at vallimamod.org Wed Sep 27 14:07:32 2017 From: vma at vallimamod.org (Vallimamod Abdullah) Date: Wed, 27 Sep 2017 16:07:32 +0200 Subject: [Freeswitch-users] cli command to list profiles In-Reply-To: <3f09f7fd-35b7-cfa8-ba9d-78994d05dfb8@xbipin.com> References: <3f09f7fd-35b7-cfa8-ba9d-78994d05dfb8@xbipin.com> Message-ID: Hi, Have you tried 'sofia status' ? It wil give you all active profiles and gateways. Best Regards, -- Vallimamod Abdullah SIP Solutions vma at sipsolutions.fr . > On 27 Sep 2017, at 15:14, Bipin Patel wrote: > > hi, > > is there any cli command to list names of all sip profiles as at times clients use different names other than internal and external > > > -- > Regards, > Bipin > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From bipin at xbipin.com Wed Sep 27 14:41:57 2017 From: bipin at xbipin.com (Bipin Patel) Date: Wed, 27 Sep 2017 18:41:57 +0400 Subject: [Freeswitch-users] cli command to list profiles In-Reply-To: References: <3f09f7fd-35b7-cfa8-ba9d-78994d05dfb8@xbipin.com> Message-ID: hi, yes that i know but i want something that outputs profile names only coz i developed a CDR portal so adding more things to it so can make it open source which would work like a drop in php portal that works out of the box and to see active registrations etc i need to know the profile names so then can sofia status profile internal reg Regards, Bipin ------------------------------------------------------------------------ -------- Original Message -------- Subject: Re: [Freeswitch-users] cli command to list profiles From: Vallimamod Abdullah To: FreeSWITCH Users Help Date: 9/27/2017, 6:07:32 PM > Hi, > > Have you tried 'sofia status' ? > It wil give you all active profiles and gateways. > > Best Regards, > -- > Vallimamod Abdullah > SIP Solutions > vma at sipsolutions.fr > . > >> On 27 Sep 2017, at 15:14, Bipin Patel > > wrote: >> >> hi, >> >> is there any cli command to list names of all sip profiles as at >> times clients use different names other than internal and external >> >> >> -- >> Regards, >> Bipin >> >> >> ------------------------------------------------------------------------ >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From sales at diamondcard.us Mon Sep 25 15:06:05 2017 From: sales at diamondcard.us (Diamondcard Support) Date: Mon, 25 Sep 2017 18:06:05 +0300 Subject: [Freeswitch-users] How can I use an external database to create new users? In-Reply-To: <2b6527f0-b831-fb2b-4a8e-47b43cdb4c87@level5.de> References: <2b6527f0-b831-fb2b-4a8e-47b43cdb4c87@level5.de> Message-ID: On 09/25/2017 01:44 PM, Thorsten Göllner wrote: > Hi, > > did you take a look at existing (Web)Guis like FusionPBX? > > https://www.fusionpbx.com/ It's definitely a GUI. But so buggy. Some things we have to do direct in Freeswitch and other changes in Fusion to get it working right. Not so much of a community in the forums either. But it does get the job done eventually. S > > Thorsten > > Am 25.09.2017 um 10:28 schrieb Jose David Jurado Alonso: >> Hi, >> >> I'm thinking to create a web application to create, edit and delete >> users in a simpler way than creating XML by hand but I don't find >> information about it. >> >> I've seen something about the module "mod_xml_curl" but I'm not sure >> how it interacts with the database nor if it's necessary some >> additional module more. >> >> I would like to use MySQL or postgreSQL, does anyone know how you can >> do it or some tutorial or reference application? >> >> Thanks, >> >> José D. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From gmina at connectfirst.com Mon Sep 25 21:17:21 2017 From: gmina at connectfirst.com (Geoff Mina) Date: Mon, 25 Sep 2017 15:17:21 -0600 Subject: [Freeswitch-users] Multiple FS Servers - Sofia Shared Database - Behind Kamailio Message-ID: Greetings, I am hoping this is a relatively simple question - but I can't find a lot of specific information on what I am attempting to do - although I have found enough to believe that this configuration is possible. I have 2 FS hosts (FreeSWITCH Version 1.6.19~64bit) behind a Kamailio server. Kamailio is routing REGISTER and INVITE requests to both using the Dispatcher module. I have a single MySQL server shared by both FS hosts and mod_sofia is properly using the shared MySQL server via ODBC on the back-end. I have two softphones registering through Kamailio and all is good. I can see both registrations in the "sip_registrations" table. The problem comes in when I try to call one of the extensions. If the INVITE arrives on the opposite server to the last REGISTER for that extension - the FS server can't see the registered user since the "hostname" is different now. I can experiment with it by using "sofia_contact 1000" on both servers - and only the last server that accepted the REGISTER sees the user, the other returns "error/user_not_registered". Can anyone point me in the right direction to understand what might need to be changed to allow both FS servers to share the same records in "sip_registrations"? Thanks, Geoff -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmina at connectfirst.com Mon Sep 25 21:53:13 2017 From: gmina at connectfirst.com (Geoff Mina) Date: Mon, 25 Sep 2017 15:53:13 -0600 Subject: [Freeswitch-users] Multiple FS Servers - Sofia Shared Database - Behind Kamailio In-Reply-To: References: Message-ID: Sorry for the noise - I figured out the issue. The default configuration had $${domain} defaulting to the IPv4 address - once I updated the domain to the common hostname that I was registering to (kamailio DNS) everything started working properly. *GEOFF MINA*Chief Executive Officer Connect First / Contact Center Solutions, Built Better. 3101 Iris Ave #200, Boulder, CO 80301 720.335.5924 <%28720%29%20335-5924> gmina at connectfirst.com / www.connectfirst.com [image: https://docs.google.com/uc?export=download&id=0B5b6KnVfm9lJTlFrQzRVUjJ2ZVE&revid=0B5b6KnVfm9lJUXpUMTFEbGJvaktwN1p5ejM3YTFkdWVWNzBzPQ] This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. On Mon, Sep 25, 2017 at 3:17 PM, Geoff Mina wrote: > Greetings, > I am hoping this is a relatively simple question - but I can't find a lot > of specific information on what I am attempting to do - although I have > found enough to believe that this configuration is possible. > > I have 2 FS hosts (FreeSWITCH Version 1.6.19~64bit) behind a Kamailio > server. Kamailio is routing REGISTER and INVITE requests to both using the > Dispatcher module. > > I have a single MySQL server shared by both FS hosts and mod_sofia is > properly using the shared MySQL server via ODBC on the back-end. > > I have two softphones registering through Kamailio and all is good. I can > see both registrations in the "sip_registrations" table. > > The problem comes in when I try to call one of the extensions. If the > INVITE arrives on the opposite server to the last REGISTER for that > extension - the FS server can't see the registered user since the > "hostname" is different now. > > I can experiment with it by using "sofia_contact 1000" on both servers - > and only the last server that accepted the REGISTER sees the user, the > other returns "error/user_not_registered". > > Can anyone point me in the right direction to understand what might need > to be changed to allow both FS servers to share the same records in > "sip_registrations"? > > Thanks, > Geoff > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image002.png Type: image/png Size: 4537 bytes Desc: not available URL: From vikas452 at gmail.com Mon Sep 25 10:33:59 2017 From: vikas452 at gmail.com (vikas sharma) Date: Mon, 25 Sep 2017 16:03:59 +0530 Subject: [Freeswitch-users] RTP Loss with Freeswitch in Default Mode Message-ID: Hi I am making a linphone to linphone call using Freeswitch server in Defualt mode but i am facing huge RTP loss and the video call is getting stuck in between.I have also found that there is a difference between the RTP packets received on A Leg and sent from B-leg and vice versa. Can somebody help me to understand what exactly happens with the RTP packets on server and why there is a difference in the RTP count?? How can we improve the video call quality in moderate network conditions. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Wed Sep 27 18:08:09 2017 From: david.villasmil.work at gmail.com (David Villasmil) Date: Wed, 27 Sep 2017 20:08:09 +0200 Subject: [Freeswitch-users] cli command to list profiles In-Reply-To: References: <3f09f7fd-35b7-cfa8-ba9d-78994d05dfb8@xbipin.com> Message-ID: fs_cli -x "sofia status" | grep "profile.*@" | awk '{print $1}' a little convoluted ;) ᐧ Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 On Wed, Sep 27, 2017 at 4:41 PM, Bipin Patel wrote: > hi, > > yes that i know but i want something that outputs profile names only coz i > developed a CDR portal so adding more things to it so can make it open > source which would work like a drop in php portal that works out of the box > and to see active registrations etc i need to know the profile names so > then can sofia status profile internal reg > > > Regards, > Bipin > > > ------------------------------ > -------- Original Message -------- > Subject: Re: [Freeswitch-users] cli command to list profiles > From: Vallimamod Abdullah > To: FreeSWITCH Users Help > > Date: 9/27/2017, 6:07:32 PM > > Hi, > > Have you tried 'sofia status' ? > It wil give you all active profiles and gateways. > > Best Regards, > -- > Vallimamod Abdullah > SIP Solutions > vma at sipsolutions.fr > . > > On 27 Sep 2017, at 15:14, Bipin Patel wrote: > > hi, > > is there any cli command to list names of all sip profiles as at times > clients use different names other than internal and external > > > -- > Regards, > Bipin > > > ------------------------------ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bipin at xbipin.com Wed Sep 27 19:41:23 2017 From: bipin at xbipin.com (Bipin Patel) Date: Wed, 27 Sep 2017 23:41:23 +0400 Subject: [Freeswitch-users] cli command to list profiles In-Reply-To: References: <3f09f7fd-35b7-cfa8-ba9d-78994d05dfb8@xbipin.com> Message-ID: <15ec4dba6b8.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com> I hope this works when called from a php script. I was hoping there would be any command similar to show xxx as xml or json as that's easier to parse but all day searching and couldn't find anything. Thanks will try it tomorrow On September 27, 2017 10:10:56 PM David Villasmil wrote: > fs_cli -x "sofia status" | grep "profile.*@" | awk '{print $1}' > > a little convoluted ;) > ᐧ > > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 > > On Wed, Sep 27, 2017 at 4:41 PM, Bipin Patel wrote: > >> hi, >> >> yes that i know but i want something that outputs profile names only coz i >> developed a CDR portal so adding more things to it so can make it open >> source which would work like a drop in php portal that works out of the box >> and to see active registrations etc i need to know the profile names so >> then can sofia status profile internal reg >> >> >> Regards, >> Bipin >> >> >> ------------------------------ >> -------- Original Message -------- >> Subject: Re: [Freeswitch-users] cli command to list profiles >> From: Vallimamod Abdullah >> To: FreeSWITCH Users Help >> >> Date: 9/27/2017, 6:07:32 PM >> >> Hi, >> >> Have you tried 'sofia status' ? >> It wil give you all active profiles and gateways. >> >> Best Regards, >> -- >> Vallimamod Abdullah >> SIP Solutions >> vma at sipsolutions.fr >> . >> >> On 27 Sep 2017, at 15:14, Bipin Patel wrote: >> >> hi, >> >> is there any cli command to list names of all sip profiles as at times >> clients use different names other than internal and external >> >> >> -- >> Regards, >> Bipin >> >> >> ------------------------------ >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting >> Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >> >> Official FreeSWITCH >> Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com >> >> FreeSWITCH-users mailing >> listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > ---------- > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From steveayre at gmail.com Wed Sep 27 22:43:38 2017 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 27 Sep 2017 23:43:38 +0100 Subject: [Freeswitch-users] cli command to list profiles In-Reply-To: <15ec4dba6b8.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com> References: <3f09f7fd-35b7-cfa8-ba9d-78994d05dfb8@xbipin.com> <15ec4dba6b8.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com> Message-ID: Try "sofia xmlstatus", xpath to the profile names would be "//profiles/profile/name/text()" On 27 September 2017 at 20:41, Bipin Patel wrote: > I hope this works when called from a php script. I was hoping there would > be any command similar to show xxx as xml or json as that's easier to parse > but all day searching and couldn't find anything. > > Thanks will try it tomorrow > > On September 27, 2017 10:10:56 PM David Villasmil < > david.villasmil.work at gmail.com> wrote: > >> fs_cli -x "sofia status" | grep "profile.*@" | awk '{print $1}' >> >> a little convoluted ;) >> ᐧ >> >> Regards, >> >> David Villasmil >> email: david.villasmil.work at gmail.com >> phone: +34669448337 <+34%20669%2044%2083%2037> >> >> On Wed, Sep 27, 2017 at 4:41 PM, Bipin Patel wrote: >> >>> hi, >>> >>> yes that i know but i want something that outputs profile names only coz >>> i developed a CDR portal so adding more things to it so can make it open >>> source which would work like a drop in php portal that works out of the box >>> and to see active registrations etc i need to know the profile names so >>> then can sofia status profile internal reg >>> >>> >>> Regards, >>> Bipin >>> >>> >>> ------------------------------ >>> -------- Original Message -------- >>> Subject: Re: [Freeswitch-users] cli command to list profiles >>> From: Vallimamod Abdullah >>> To: FreeSWITCH Users Help >>> >>> Date: 9/27/2017, 6:07:32 PM >>> >>> Hi, >>> >>> Have you tried 'sofia status' ? >>> It wil give you all active profiles and gateways. >>> >>> Best Regards, >>> -- >>> Vallimamod Abdullah >>> SIP Solutions >>> vma at sipsolutions.fr >>> . >>> >>> On 27 Sep 2017, at 15:14, Bipin Patel wrote: >>> >>> hi, >>> >>> is there any cli command to list names of all sip profiles as at times >>> clients use different names other than internal and external >>> >>> >>> -- >>> Regards, >>> Bipin >>> >>> >>> ------------------------------ >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com >>> >>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >>> >>> >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Wed Sep 27 22:59:37 2017 From: david.villasmil.work at gmail.com (David Villasmil) Date: Thu, 28 Sep 2017 00:59:37 +0200 Subject: [Freeswitch-users] How can I use an external database to create new users? In-Reply-To: References: <2b6527f0-b831-fb2b-4a8e-47b43cdb4c87@level5.de> Message-ID: As has been pointed out, what you want, can't be done out of the box. One way to do it is with mod_xml_curl, works perfectly. Try it. On Sep 27, 2017 17:05, "Diamondcard Support" wrote: > On 09/25/2017 01:44 PM, Thorsten Göllner wrote: > > Hi, > > > > did you take a look at existing (Web)Guis like FusionPBX? > > > > https://www.fusionpbx.com/ > > It's definitely a GUI. But so buggy. Some things we have to do direct in > Freeswitch and other changes in Fusion to get it working right. Not so > much of a community in the forums either. But it does get the job done > eventually. > > S > > > > > Thorsten > > > > Am 25.09.2017 um 10:28 schrieb Jose David Jurado Alonso: > >> Hi, > >> > >> I'm thinking to create a web application to create, edit and delete > >> users in a simpler way than creating XML by hand but I don't find > >> information about it. > >> > >> I've seen something about the module "mod_xml_curl" but I'm not sure > >> how it interacts with the database nor if it's necessary some > >> additional module more. > >> > >> I would like to use MySQL or postgreSQL, does anyone know how you can > >> do it or some tutorial or reference application? > >> > >> Thanks, > >> > >> José D. > >> > >> > >> ____________________________________________________________ > _____________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://confluence.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/ > options/freeswitch-users > >> http://www.freeswitch.org > > > > > > > > ____________________________________________________________ > _____________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From chandranraviram at gmail.com Thu Sep 28 07:02:59 2017 From: chandranraviram at gmail.com (Raviram Chandran) Date: Thu, 28 Sep 2017 12:32:59 +0530 Subject: [Freeswitch-users] How to overwrite Q850 errors code in freeswitch? Message-ID: Since we have a errors code in PSTN call (IP -PSTN call) which we wanton our telephony server to pass those errors code directly to mobile apps. When subscriber's balance is lower than some predefined value during call setup, Warning header with SIP 180/183 response for low balance alert goes to calling Party (mobile client) and the call continues. If heart-beat with calling dialog fails or maximum time limit for the call is reached, Warning header with BYE message goes to calling party (mobile client) and the call is dropped. Other than this, following failure scenarios may hit during a call 1. Authentication failed 2. Account temporarily deactivated 3. Zero or inadequate balance while making call 4. Account already in use by configured number of users 5. Rate not defined 6. Called destination blacklisted 7. Any other application failure For all such failure scenarios, IN will send 408 SIP responses to client. This response will contain a Warning header specifying cause of failure and appropriate text message (configurable at IN). Format of Warning Header is as follows Warning: Cause_Code APP_Name "Error Text", e.g., Presently error codes are configured at network provider end which we need to pass from our telephony server to mobile apps -------------- next part -------------- An HTML attachment was scrubbed... URL: From bipin at xbipin.com Thu Sep 28 07:10:46 2017 From: bipin at xbipin.com (Bipin Patel) Date: Thu, 28 Sep 2017 11:10:46 +0400 Subject: [Freeswitch-users] cli command to list profiles In-Reply-To: References: <3f09f7fd-35b7-cfa8-ba9d-78994d05dfb8@xbipin.com> <15ec4dba6b8.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com> Message-ID: hi, both methods work, brilliant, thanks Regards, Bipin ------------------------------------------------------------------------ -------- Original Message -------- Subject: Re: [Freeswitch-users] cli command to list profiles From: Steven Ayre To: FreeSWITCH Users Help Date: 9/28/2017, 2:43:38 AM > Try "sofia xmlstatus", xpath to the profile names would be > "//profiles/profile/name/text()" > > On 27 September 2017 at 20:41, Bipin Patel > wrote: > > I hope this works when called from a php script. I was hoping > there would be any command similar to show xxx as xml or json as > that's easier to parse but all day searching and couldn't find > anything. > > Thanks will try it tomorrow > > On September 27, 2017 10:10:56 PM David Villasmil > > wrote: > >>  fs_cli -x "sofia status" | grep "profile.*@" | awk '{print $1}' >> >> a little convoluted ;) >> ᐧ >> >> Regards, >> >> David Villasmil >> email: david.villasmil.work at gmail.com >> >> phone: +34669448337 >> >> On Wed, Sep 27, 2017 at 4:41 PM, Bipin Patel > > wrote: >> >> hi, >> >> yes that i know but i want something that outputs profile >> names only coz i developed a CDR portal so adding more things >> to it so can make it open source which would work like a drop >> in php portal that works out of the box and to see active >> registrations etc i need to know the profile names so then >> can sofia status profile internal reg >> >> >> Regards, >> Bipin >> >> >> ------------------------------------------------------------------------ >> -------- Original Message -------- >> Subject: Re: [Freeswitch-users] cli command to list profiles >> From: Vallimamod Abdullah >> >> To: FreeSWITCH Users Help >> >> >> Date: 9/27/2017, 6:07:32 PM >>> Hi, >>> >>> Have you tried 'sofia status' ? >>> It wil give you all active profiles and gateways. >>> >>> Best Regards, >>> -- >>> Vallimamod Abdullah >>> SIP Solutions >>> vma at sipsolutions.fr >>> . >>> >>>> On 27 Sep 2017, at 15:14, Bipin Patel >>> > wrote: >>>> >>>> hi, >>>> >>>> is there any cli command to list names of all sip profiles >>>> as at times clients use different names other than internal >>>> and external >>>> >>>> >>>> -- >>>> Regards, >>>> Bipin >>>> >>>> >>>> ------------------------------------------------------------------------ >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> >>>> http://www.freeswitch.org >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >>> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From shaun.stokes at itec-support.co.uk Thu Sep 28 12:04:07 2017 From: shaun.stokes at itec-support.co.uk (Shaun Stokes) Date: Thu, 28 Sep 2017 12:04:07 +0000 Subject: [Freeswitch-users] Bug - LUA session:recordFile when using video phones Message-ID: <6FD2F8B5BB72834E9939AEDF9FB802A901E86A1595@mbx-01.sysconfig.co.uk> Hi All, FreeSWITCH version 1.6.19 We are using LUA “session:recordFile” to record greetings which works perfectly on non-video devices, however video phones (such as Polycom VVX500\VVX600, Yealink T49G etc) are not supported unless video is disabled on the device thus preventing it from attempting to negotiate video in the SDP message. We have H264 enabled in FS gobal_codec_prefs, disabling H264 also resolves the problem. From the point we begin the recording there is a 10 second delay before the recording starts on video devices, recording starts after the error “Unable to establish inbound video stream” occurs. 2017-09-28 13:01:39.885386 [NOTICE] switch_core_io.c:1202 Activating write resampler 2017-09-28 13:01:39.985376 [DEBUG] switch_cpp.cpp:895 getDigits dtmf_buf: 2017-09-28 13:01:49.985371 [ERR] switch_ivr_play_say.c:496 Unable to establish inbound video stream 2017-09-28 13:01:49.985371 [DEBUG] switch_ivr_play_say.c:560 Raw Codec Activated, ready to waste resources! 2017-09-28 13:01:49.985371 [DEBUG] switch_ivr_play_say.c:674 Raw Codec Activated We’ve tried to work around the issue by adding this to our dialplan before the call is answered but the same issue occurs: Looks like this is a known issue: https://groups.google.com/forum/#!topic/2600hz-users/61XyiXya5Ik https://freeswitch.org/jira/si/jira.issueviews:issue-html/FS-9632/FS-9632.html The JIRA “FS-9632” was marked as resolved but doesn’t appear to apply to LUA when using “session:recordFile”. https://freeswitch.org/jira/browse/FS-9632 Perhaps we’ve missed something, does anyone have any ideas? Thanks, Shaun ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: From mirkobrankovic at gmail.com Thu Sep 28 13:33:14 2017 From: mirkobrankovic at gmail.com (Mirko Brankovic) Date: Thu, 28 Sep 2017 15:33:14 +0200 Subject: [Freeswitch-users] ICE/DTLS handshake In-Reply-To: References: Message-ID: Looks like the problem lies in fact that both call legs/channels in refer call scenario are in my case outgoing legs, and then they are sending CONTROLED stun username requests, so I don't have CONTROLING side. while at the same time Video stun is negotiated immediately and correctly. Log shows it I guess. https://pastebin.freeswitch.org/view/33d30c7e On Sat, Sep 23, 2017 at 3:12 PM, Mirko Brankovic wrote: > Thanks Mike, > all clients are webrtc clients behind same freeswitch. > At the same time video rtp/rtcp dtls is instant, but audio is waiting for > something, my best guess is for rtcp from client to confirm correct ip.port > or to do auto correct. > Call scenario is A call B and B transfers(refer) call to C. > Also I see in those 5 seconds that first client, A sends total of 6 stun > username requests and that C answers to them all at the same time, after 5s. > Can this be rtcp problem. > I was thinking to go through video ice thread and compare it to audio to > see how that one works instantly. > thanks, > Mirko > > > > On Sep 21, 2017 19:55, "Michael Jerris" wrote: > > its not going to negotiate until we get the stun responses. If we are > not, you should look if the client is sending them and something is > blocking, or why the client is waiting to send them. Sounds broken on > client side from the description. > > On Sep 21, 2017, at 7:53 AM, Mirko Brankovic > wrote: > > HI, > Has anyone experienced DTLS handshake takes 5s to get to SETUP state: > >> 2017-09-21 11:40:40.021178 [INFO] switch_rtp.c:3515 Changing audio DTLS >> state from OFF to HANDSHAKE >> 2017-09-21 11:40:45.319457 [INFO] switch_rtp.c:3172 Changing audio DTLS >> state from HANDSHAKE to SETUP > > > In network dump i see that answering side is not sending STUN for this 5s > and then suddenly answers last 5 STUNs from A side. > > Has anyone encountered this kind of problem ? > > I have a pcap if necessary... > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > -- Regards, Mirko -------------- next part -------------- An HTML attachment was scrubbed... URL: From mirkobrankovic at gmail.com Thu Sep 28 13:46:39 2017 From: mirkobrankovic at gmail.com (Mirko Brankovic) Date: Thu, 28 Sep 2017 15:46:39 +0200 Subject: [Freeswitch-users] ICE/DTLS handshake In-Reply-To: References: Message-ID: actually the answering leg responds with CONTROLING request On Thu, Sep 28, 2017 at 3:33 PM, Mirko Brankovic wrote: > Looks like the problem lies in fact that both call legs/channels in refer > call scenario are in my case outgoing legs, and then they are sending > CONTROLED stun username requests, so I don't have CONTROLING side. > > while at the same time Video stun is negotiated immediately and correctly. > Log shows it I guess. > https://pastebin.freeswitch.org/view/33d30c7e > > On Sat, Sep 23, 2017 at 3:12 PM, Mirko Brankovic > wrote: > >> Thanks Mike, >> all clients are webrtc clients behind same freeswitch. >> At the same time video rtp/rtcp dtls is instant, but audio is waiting for >> something, my best guess is for rtcp from client to confirm correct ip.port >> or to do auto correct. >> Call scenario is A call B and B transfers(refer) call to C. >> Also I see in those 5 seconds that first client, A sends total of 6 stun >> username requests and that C answers to them all at the same time, after 5s. >> Can this be rtcp problem. >> I was thinking to go through video ice thread and compare it to audio to >> see how that one works instantly. >> thanks, >> Mirko >> >> >> >> On Sep 21, 2017 19:55, "Michael Jerris" wrote: >> >> its not going to negotiate until we get the stun responses. If we are >> not, you should look if the client is sending them and something is >> blocking, or why the client is waiting to send them. Sounds broken on >> client side from the description. >> >> On Sep 21, 2017, at 7:53 AM, Mirko Brankovic >> wrote: >> >> HI, >> Has anyone experienced DTLS handshake takes 5s to get to SETUP state: >> >>> 2017-09-21 11:40:40.021178 [INFO] switch_rtp.c:3515 Changing audio DTLS >>> state from OFF to HANDSHAKE >>> 2017-09-21 11:40:45.319457 [INFO] switch_rtp.c:3172 Changing audio DTLS >>> state from HANDSHAKE to SETUP >> >> >> In network dump i see that answering side is not sending STUN for this 5s >> and then suddenly answers last 5 STUNs from A side. >> >> Has anyone encountered this kind of problem ? >> >> I have a pcap if necessary... >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> > > > -- > Regards, > Mirko > -- Regards, Mirko -------------- next part -------------- An HTML attachment was scrubbed... URL: From olegstolyar at gmail.com Fri Sep 29 13:44:38 2017 From: olegstolyar at gmail.com (Oleg Stolyar) Date: Fri, 29 Sep 2017 06:44:38 -0700 Subject: [Freeswitch-users] FreeSWITCH 1.8 Message-ID: Hi guys, sorry if I missed an announcement, but has version 1.8 been officially released? -------------- next part -------------- An HTML attachment was scrubbed... URL: From jungleboogie0 at gmail.com Fri Sep 29 19:39:41 2017 From: jungleboogie0 at gmail.com (jungle Boogie) Date: Fri, 29 Sep 2017 12:39:41 -0700 Subject: [Freeswitch-users] FreeSWITCH 1.8 In-Reply-To: References: Message-ID: On 29 September 2017 at 06:44, Oleg Stolyar wrote: > Hi guys, > > sorry if I missed an announcement, but has version 1.8 been officially > released? > It doesn't look like it. Look on the lower right, under the books and you'll see 1.6.19 is the current version. https://freeswitch.org From jungleboogie0 at gmail.com Fri Sep 29 19:49:28 2017 From: jungleboogie0 at gmail.com (jungle Boogie) Date: Fri, 29 Sep 2017 12:49:28 -0700 Subject: [Freeswitch-users] Bug - LUA session:recordFile when using video phones In-Reply-To: <6FD2F8B5BB72834E9939AEDF9FB802A901E86A1595@mbx-01.sysconfig.co.uk> References: <6FD2F8B5BB72834E9939AEDF9FB802A901E86A1595@mbx-01.sysconfig.co.uk> Message-ID: On 28 September 2017 at 05:04, Shaun Stokes wrote: > Hi All, > > FreeSWITCH version 1.6.19 > > Looks like this is a known issue: > > https://groups.google.com/forum/#!topic/2600hz-users/61XyiXya5Ik > > https://freeswitch.org/jira/si/jira.issueviews:issue-html/FS-9632/FS-9632.html > > > > The JIRA “FS-9632” was marked as resolved but doesn’t appear to apply to LUA > when using “session:recordFile”. > > https://freeswitch.org/jira/browse/FS-9632 > > > > Perhaps we’ve missed something, does anyone have any ideas? > > Well that bug has a fix version of a release that hasn't happened yet, so I don't know if that means it's not in your 1.6.19 build. Here's the commit: https://freeswitch.org/stash/projects/FS/repos/freeswitch/commits/efc2ed2a49372e1160b8cdd84872e7876ef01779 At worse, you can make a bug and reference the one you found above. > > Thanks, > > Shaun > > From mike at jerris.com Sat Sep 30 00:07:39 2017 From: mike at jerris.com (Michael Jerris) Date: Sat, 30 Sep 2017 00:07:39 +0000 Subject: [Freeswitch-users] Bug - LUA session:recordFile when using video phones In-Reply-To: References: <6FD2F8B5BB72834E9939AEDF9FB802A901E86A1595@mbx-01.sysconfig.co.uk> Message-ID: This is fixed in 1.6 as well.. it’s jira wasn’t updated, i have on my backlog to go back and fix these in jira to properly reflect the right fix version On Fri, Sep 29, 2017 at 3:50 PM jungle Boogie wrote: > On 28 September 2017 at 05:04, Shaun Stokes > wrote: > > Hi All, > > > > FreeSWITCH version 1.6.19 > > > > Looks like this is a known issue: > > > > https://groups.google.com/forum/#!topic/2600hz-users/61XyiXya5Ik > > > > > https://freeswitch.org/jira/si/jira.issueviews:issue-html/FS-9632/FS-9632.html > > > > > > > > The JIRA “FS-9632” was marked as resolved but doesn’t appear to apply to > LUA > > when using “session:recordFile”. > > > > https://freeswitch.org/jira/browse/FS-9632 > > > > > > > > Perhaps we’ve missed something, does anyone have any ideas? > > > > > > Well that bug has a fix version of a release that hasn't happened yet, > so I don't know if that means it's not in your 1.6.19 build. > Here's the commit: > > https://freeswitch.org/stash/projects/FS/repos/freeswitch/commits/efc2ed2a49372e1160b8cdd84872e7876ef01779 > > At worse, you can make a bug and reference the one you found above. > > > > > Thanks, > > > > Shaun > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From olegstolyar at gmail.com Sat Sep 30 00:24:15 2017 From: olegstolyar at gmail.com (Oleg Stolyar) Date: Fri, 29 Sep 2017 17:24:15 -0700 Subject: [Freeswitch-users] FreeSWITCH 1.8 In-Reply-To: References: Message-ID: Thanks Jungle! On Fri, Sep 29, 2017 at 12:39 PM, jungle Boogie wrote: > On 29 September 2017 at 06:44, Oleg Stolyar wrote: > > Hi guys, > > > > sorry if I missed an announcement, but has version 1.8 been officially > > released? > > > > It doesn't look like it. Look on the lower right, under the books and > you'll see 1.6.19 is the current version. > > https://freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From andrew at convergedgroup.net Thu Sep 28 14:24:00 2017 From: andrew at convergedgroup.net (Andrew Colin) Date: Thu, 28 Sep 2017 16:24:00 +0200 (SAST) Subject: [Freeswitch-users] Cisco 8845 Message-ID: <030601d33865$159699c0$40c3cd40$@convergedgroup.net> Hi Guys Has anyone got a cisco 8845 IP phone working on freeswitch? -------------- next part -------------- An HTML attachment was scrubbed... URL: