From marcel.haldemann at convercom.ch Mon May 1 18:09:16 2017 From: marcel.haldemann at convercom.ch (Marcel Haldemann) Date: Mon, 1 May 2017 14:09:16 +0000 Subject: [Freeswitch-users] Freeswitch and IoT In-Reply-To: References: Message-ID: Hi Giovanni, Thanks for ur reply. Here is what im currently thinking of to use FreeSwitch for (currently it?s just an idea): I would like to use FreeSwitch for IoT (for example to run or check something against a dialplan if sensor or ?alarm event? values arrive via http/mqtt/amqp or similar protocols). For this purpose I would like to create a channel without a sip call (I would write wrappers for all the necessary protocols, get the info on all protocols and set the variables in the same way no matter what protocol they come from). The variables I would set would be things such as location and name of the sensor and the value (Values coming from mqtt, amqp or http). Then I would like to run it against a dialplan with all the possibilities as it would be a call (for sure I would have to check some variables). (I would use mod_xml_curl to get the dialplan in redis) If necessary i would like to add an a-leg with bridge and maybe play some files an maybe get some dtmfs with lua (ivr). If necessary I would like to add a b-leg with bridge. I now asked me whether it?s possible to create a channel out of nothing, set some variables and let it run against the dialplan context. Is this possible in some way ? It must not be with event_socket I also could use some other mod. But maybe it?s necessary to write a mod_endpoint_mqtt or similar for this ? Regards, Marcel Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Giovanni Maruzzelli Gesendet: Samstag, 29. April 2017 19:46 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] Freeswitch and IoT Hello Marcel, can you elaborate? Don't be so terse, there can be a lot of meanings in your question, and lot of different answers. Please specify a clear use case, etc etc -giovanni On 29 April 2017 at 19:31, Marcel Haldemann > wrote: Hi Guys, is it possible to create a channel using event socket (without calling a number) ? LG, Marcel _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170501/528a855d/attachment.html From sdevoy at bizfocused.com Mon May 1 18:31:23 2017 From: sdevoy at bizfocused.com (Sean Devoy) Date: Mon, 1 May 2017 14:31:23 +0000 Subject: [Freeswitch-users] Help - [INCOMPATIBLE_DESTINATION] Message-ID: Can anyone see why this is "[INCOMPATIBLE_DESTINATION]" today. NOTHING changed over the weekend. d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:20.980418 [DEBUG] switch_core_session.c:1351 Send signal sofia/external/sip:203 at 198.17.240.238:5062 [BREAK] d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:20.980418 [DEBUG] switch_core_state_machine.c:415 (sofia/external/sip:203 at 198.17.240.238:5062) Running State Change CS_INIT d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:20.980418 [DEBUG] switch_core_state_machine.c:455 (sofia/external/sip:203 at 198.17.240.238:5062) State INIT d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:20.980418 [DEBUG] mod_sofia.c:87 sofia/external/sip:203 at 198.17.240.238:5062 SOFIA INIT d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:20.980418 [DEBUG] sofia_glue.c:2716 sip:203 at 198.17.240.238:5062 Setting proxy route to sofia/external/sip:203 at 198.17.240.238:5062 d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:20.980418 [DEBUG] sofia_glue.c:2744 sofia/external/sip:203 at 198.17.240.238:5062 sending invite version: 1.2.22 git 65fed13 2014-03-09 21:21:37Z 64bit d12984f7-12ec-430f-99a7-43202963672a Local SDP: d12984f7-12ec-430f-99a7-43202963672a v=0 d12984f7-12ec-430f-99a7-43202963672a o=FreeSWITCH 1493622368 1493622369 IN IP4 66.241.102.41 d12984f7-12ec-430f-99a7-43202963672a s=FreeSWITCH d12984f7-12ec-430f-99a7-43202963672a c=IN IP4 66.241.102.41 d12984f7-12ec-430f-99a7-43202963672a t=0 0 d12984f7-12ec-430f-99a7-43202963672a m=audio 23472 RTP/AVP 0 13 d12984f7-12ec-430f-99a7-43202963672a a=ptime:20 d12984f7-12ec-430f-99a7-43202963672a a=sendrecv d12984f7-12ec-430f-99a7-43202963672a d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:20.980418 [DEBUG] switch_core_session.c:1016 Send signal sofia/external/sip:203 at 198.17.240.238:5062 [BREAK] d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:20.980418 [DEBUG] mod_sofia.c:127 (sofia/external/sip:203 at 198.17.240.238:5062) State Change CS_INIT -> CS_ROUTING d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:20.980418 [DEBUG] switch_core_session.c:1351 Send signal sofia/external/sip:203 at 198.17.240.238:5062 [BREAK] d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:20.980418 [DEBUG] switch_core_state_machine.c:455 (sofia/external/sip:203 at 198.17.240.238:5062) State INIT going to sleep d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:20.980418 [DEBUG] switch_core_state_machine.c:415 (sofia/external/sip:203 at 198.17.240.238:5062) Running State Change CS_ROUTING d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:20.980418 [DEBUG] sofia.c:5815 Channel sofia/external/sip:203 at 198.17.240.238:5062 entering state [calling][0] d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:20.980418 [DEBUG] switch_core_state_machine.c:471 (sofia/external/sip:203 at 198.17.240.238:5062) State ROUTING d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:20.980418 [DEBUG] mod_sofia.c:150 sofia/external/sip:203 at 198.17.240.238:5062 SOFIA ROUTING d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:20.980418 [DEBUG] switch_ivr_originate.c:67 (sofia/external/sip:203 at 198.17.240.238:5062) State Change CS_ROUTING -> CS_CONSUME_MEDIA d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:20.980418 [DEBUG] switch_core_session.c:1351 Send signal sofia/external/sip:203 at 198.17.240.238:5062 [BREAK] d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:20.980418 [DEBUG] switch_core_state_machine.c:471 (sofia/external/sip:203 at 198.17.240.238:5062) State ROUTING going to sleep d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:20.980418 [DEBUG] switch_core_state_machine.c:415 (sofia/external/sip:203 at 198.17.240.238:5062) Running State Change CS_CONSUME_MEDIA d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:20.980418 [DEBUG] switch_core_state_machine.c:490 (sofia/external/sip:203 at 198.17.240.238:5062) State CONSUME_MEDIA d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:20.980418 [DEBUG] switch_core_state_machine.c:490 (sofia/external/sip:203 at 198.17.240.238:5062) State CONSUME_MEDIA going to sleep d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:21.040403 [DEBUG] switch_core_session.c:1016 Send signal sofia/external/sip:203 at 198.17.240.238:5062 [BREAK] d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:21.040403 [DEBUG] switch_core_session.c:1016 Send signal sofia/external/sip:203 at 198.17.240.238:5062 [BREAK] d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:21.040403 [DEBUG] switch_core_session.c:1016 Send signal sofia/external/sip:203 at 198.17.240.238:5062 [BREAK] d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:21.040403 [DEBUG] sofia.c:5815 Channel sofia/external/sip:203 at 198.17.240.238:5062 entering state [terminated][488] d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:21.040403 [NOTICE] sofia.c:6659 Hangup sofia/external/sip:203 at 198.17.240.238:5062 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:21.040403 [DEBUG] switch_channel.c:3187 Send signal sofia/external/sip:203 at 198.17.240.238:5062 [KILL] d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:21.040403 [DEBUG] switch_core_session.c:1351 Send signal sofia/external/sip:203 at 198.17.240.238:5062 [BREAK] d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:21.040403 [DEBUG] switch_core_state_machine.c:415 (sofia/external/sip:203 at 198.17.240.238:5062) Running State Change CS_HANGUP d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:21.040403 [DEBUG] switch_core_state_machine.c:678 (sofia/external/sip:203 at 198.17.240.238:5062) Callstate Change DOWN -> HANGUP d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:21.040403 [DEBUG] switch_core_state_machine.c:680 (sofia/external/sip:203 at 198.17.240.238:5062) State HANGUP d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:21.040403 [DEBUG] mod_sofia.c:506 Channel sofia/external/sip:203 at 198.17.240.238:5062 hanging up, cause: INCOMPATIBLE_DESTINATION d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:21.040403 [DEBUG] switch_core_state_machine.c:48 sofia/external/sip:203 at 198.17.240.238:5062 Standard HANGUP, cause: INCOMPATIBLE_DESTINATION d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:21.040403 [DEBUG] switch_core_state_machine.c:680 (sofia/external/sip:203 at 198.17.240.238:5062) State HANGUP going to sleep d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:21.040403 [DEBUG] switch_core_state_machine.c:447 (sofia/external/sip:203 at 198.17.240.238:5062) State Change CS_HANGUP -> CS_REPORTING d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:21.040403 [DEBUG] switch_core_session.c:1351 Send signal sofia/external/sip:203 at 198.17.240.238:5062 [BREAK] d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:21.040403 [DEBUG] switch_core_state_machine.c:415 (sofia/external/sip:203 at 198.17.240.238:5062) Running State Change CS_REPORTING d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:21.040403 [DEBUG] switch_core_state_machine.c:764 (sofia/external/sip:203 at 198.17.240.238:5062) State REPORTING d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:21.040403 [DEBUG] switch_core_state_machine.c:92 sofia/external/sip:203 at 198.17.240.238:5062 Standard REPORTING, cause: INCOMPATIBLE_DESTINATION d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:21.040403 [DEBUG] switch_core_state_machine.c:764 (sofia/external/sip:203 at 198.17.240.238:5062) State REPORTING going to sleep d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:21.040403 [DEBUG] switch_core_state_machine.c:441 (sofia/external/sip:203 at 198.17.240.238:5062) State Change CS_REPORTING -> CS_DESTROY d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:21.040403 [DEBUG] switch_core_session.c:1351 Send signal sofia/external/sip:203 at 198.17.240.238:5062 [BREAK] d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:21.040403 [DEBUG] switch_core_session.c:1559 Session 223 (sofia/external/sip:203 at 198.17.240.238:5062) Locked, Waiting on external entities -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170501/cc2ce6a7/attachment-0001.html From krice at freeswitch.org Mon May 1 18:36:13 2017 From: krice at freeswitch.org (Ken Rice) Date: Mon, 1 May 2017 09:36:13 -0500 Subject: [Freeswitch-users] FS core dumps In-Reply-To: References: Message-ID: <0f4101d2c288$491c2dd0$db548970$@freeswitch.org> Segfaults are bugs Bugs get reported to Jira See https://freeswitch.org/confluence/display/FREESWITCH/Reporting+Bugs+to+JIRA From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rahul MathuR Sent: Sunday, April 30, 2017 6:40 AM To: FreeSWITCH Users Help Cc: RM Subject: [Freeswitch-users] FS core dumps Hello guys, I'm trying to Auth over RADIUS from inside Lua script - session:execute("auth_function","in "..cli..", in "..dnid..", out AUTH_RESULT"); But FS keeps on dumping core files - SIGSEGV inside "extract_out_variable" inside mod_rad_auth.so My guess is that I have not defined AUTH_RESULT anywhere in the script. But then again I haven't find any example on this either. Any help would be much appreciated. -- Warm Regds. MathuRahul -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170501/a18a41cd/attachment.html From gb at cm.nl Mon May 1 18:38:51 2017 From: gb at cm.nl (Grant Bagdasarian) Date: Mon, 1 May 2017 14:38:51 +0000 Subject: [Freeswitch-users] mod_distributor from db Message-ID: Hello, Is it possible to load the gateways of a sip profile from a database and the distributor config also from a database, using built in modules/functions? Regards, Grant -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170501/3586a81d/attachment.html From anthony.minessale at gmail.com Mon May 1 20:17:34 2017 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 1 May 2017 11:17:34 -0500 Subject: [Freeswitch-users] mod_distributor from db In-Reply-To: References: Message-ID: mod_xml_curl could. On Mon, May 1, 2017 at 9:38 AM, Grant Bagdasarian wrote: > Hello, > > > > Is it possible to load the gateways of a sip profile from a database and > the distributor config also from a database, using built in > modules/functions? > > > > Regards, > > > > Grant > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170501/8ea916c3/attachment.html From colton.conor at gmail.com Mon May 1 23:03:30 2017 From: colton.conor at gmail.com (Colton Conor) Date: Mon, 1 May 2017 14:03:30 -0500 Subject: [Freeswitch-users] OPUS Codec Message-ID: Does freeswitch support transcoding from OPUS to G711? How well does this work in theory? Having full HD come in via OPUS from the phone, and then transcoding to crappy G711. Does it sound funny? We are thinking that OPUS has superior audio quality, and can withstand up to 30 percent packet loss without any audio distorition. However, I would say the majority of our users calls as still to the PSTN which only speaks G711. How much processing power would it take to transcode from OPUS to G711? How well is the OPUS codec implemented on newer IP phones? I hear the Yealink supports OPUS with their new S line of phones, but I also heard they pulled support for it? Polycom seems to only support it on the VVX 500 and VVX 600, but you have to disable video according to the 5.4 release notes. OPUS must be processor intensive I assume if it can't run OPUS and video at the same time. Have any idea if the newer VVX, like the 411, 501, and 601 have this same limitation? I know they have faster processors and ram. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170501/1f370f58/attachment.html From cjbujold at accra.ca Mon May 1 23:13:52 2017 From: cjbujold at accra.ca (Charles Bujold) Date: Mon, 1 May 2017 16:13:52 -0300 Subject: [Freeswitch-users] Outbound calls being dropped after update to Version 1.6.17 -34-0fc0946 64bit Message-ID: <001d01d2c2af$12244d40$366ce7c0$@accra.ca> Hi, Getting a strange issue, all incoming calls work great, but outgoing calls are getting dropped. Initially though that it was a NAT issue but verified the NAT and it is working properly. The proper external IP is being used and the gateway is registering with the service provider without issue. Checked the call flow with sngrep and wireshark and found some strange issues. 1. From my end, making the outbound call I see the call being connected and I receive a 200 OK that the call is connected and I have a connection. It will last about 1.5 minutes while at the same time I see 5-6 INVITE being sent from my Freeswitch to my service provider and then a BYE that it is hanging up. 2. From my service provider, they see the call established and then: SIP/SDP 82 Request: INVITE sip: telephonnumber at IPxxxxxx:5060; transport=udp, in-dialog The question is what is the "in-dialog" it seems to be causing the call to not be recognize that it is active and INVITE being generated and not being answered which causes the Bye Command to occur. Not certain how to fix. Any suggestions would be appreciated. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170501/eaae0840/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 23262 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170501/eaae0840/attachment-0002.png -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 29178 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170501/eaae0840/attachment-0003.png From colin.morelli at gmail.com Mon May 1 23:22:53 2017 From: colin.morelli at gmail.com (Colin Morelli) Date: Mon, 1 May 2017 15:22:53 -0400 Subject: [Freeswitch-users] Outbound calls being dropped after update to Version 1.6.17 -34-0fc0946 64bit In-Reply-To: <001d01d2c2af$12244d40$366ce7c0$@accra.ca> References: <001d01d2c2af$12244d40$366ce7c0$@accra.ca> Message-ID: I have no idea if this is actually the case or not (logs should probably confirm this) but it's possible that the subsequent re-invites being sent by FS are session timer refresh requests. If your gateway advertises support for session timers, but doesn't respond to re-invites, that would be a problem with your gateway. Of course you could disable session timers on FS if that is the case. Best, Colin On Mon, May 1, 2017 at 3:13 PM, Charles Bujold wrote: > Hi, > > > > Getting a strange issue, all incoming calls work great, but outgoing calls > are getting dropped. > > > > Initially though that it was a NAT issue but verified the NAT and it is > working properly. The proper external IP is being used and the gateway is > registering with the service provider without issue. > > > > Checked the call flow with sngrep and wireshark and found some strange > issues. > > > > 1. From my end, making the outbound call I see the call being > connected and I receive a 200 OK that the call is connected and I have a > connection. It will last about 1.5 minutes while at the same time I see > 5-6 INVITE being sent from my Freeswitch to my service provider and then a > BYE that it is hanging up. > > > > 1. From my service provider, they see the call established and > then: SIP/SDP 82 Request: INVITE sip: telephonnumber at IPxxxxxx:5060; > transport=udp, in-dialog > > > > The question is what is the ?in-dialog? it seems to be causing the call to > not be recognize that it is active and INVITE being generated and not being > answered which causes the Bye Command to occur. > > > > Not certain how to fix. Any suggestions would be appreciated. > > > > > > [image: cid:image002.png at 01D2C295.2EF109F0] > > > > > > [image: cid:image003.png at 01D2C294.73F37890] > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170501/14ad5bc4/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image002.png Type: image/png Size: 29178 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170501/14ad5bc4/attachment-0002.png -------------- next part -------------- A non-text attachment was scrubbed... Name: image003.png Type: image/png Size: 23262 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170501/14ad5bc4/attachment-0003.png From grcamauer at gmail.com Mon May 1 23:52:38 2017 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Mon, 1 May 2017 16:52:38 -0300 Subject: [Freeswitch-users] Help - [INCOMPATIBLE_DESTINATION] In-Reply-To: References: Message-ID: Maybe the destination doesn?t support the offered codec anymore? In any case, version 1.2.22 is way too old. You should update to 1.6. Regards, Guillermo On Mon, May 1, 2017 at 11:31 AM, Sean Devoy wrote: > Can anyone see why this is ?[INCOMPATIBLE_DESTINATION]? today. > > NOTHING changed over the weekend. > > > > d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:20.980418 [DEBUG] > switch_core_session.c:1351 Send signal sofia/external/sip:203 at 198.17. > 240.238:5062 [BREAK] > > d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:20.980418 [DEBUG] > switch_core_state_machine.c:415 (sofia/external/sip:203 at 198. > 17.240.238:5062) Running State Change CS_INIT > > d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:20.980418 [DEBUG] > switch_core_state_machine.c:455 (sofia/external/sip:203 at 198. > 17.240.238:5062) State INIT > > d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:20.980418 [DEBUG] > mod_sofia.c:87 sofia/external/sip:203 at 198.17.240.238:5062 SOFIA INIT > > d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:20.980418 [DEBUG] > sofia_glue.c:2716 sip:203 at 198.17.240.238:5062 Setting proxy route to > sofia/external/sip:203 at 198.17.240.238:5062 > > d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:20.980418 [DEBUG] > sofia_glue.c:2744 sofia/external/sip:203 at 198.17.240.238:5062 sending > invite version: 1.2.22 git 65fed13 2014-03-09 21:21:37Z 64bit > > d12984f7-12ec-430f-99a7-43202963672a Local SDP: > > d12984f7-12ec-430f-99a7-43202963672a v=0 > > d12984f7-12ec-430f-99a7-43202963672a o=FreeSWITCH 1493622368 1493622369 > IN IP4 66.241.102.41 > > d12984f7-12ec-430f-99a7-43202963672a s=FreeSWITCH > > d12984f7-12ec-430f-99a7-43202963672a c=IN IP4 66.241.102.41 > > d12984f7-12ec-430f-99a7-43202963672a t=0 0 > > d12984f7-12ec-430f-99a7-43202963672a m=audio 23472 RTP/AVP 0 13 > > d12984f7-12ec-430f-99a7-43202963672a a=ptime:20 > > d12984f7-12ec-430f-99a7-43202963672a a=sendrecv > > d12984f7-12ec-430f-99a7-43202963672a > > d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:20.980418 [DEBUG] > switch_core_session.c:1016 Send signal sofia/external/sip:203 at 198.17. > 240.238:5062 [BREAK] > > d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:20.980418 [DEBUG] > mod_sofia.c:127 (sofia/external/sip:203 at 198.17.240.238:5062) State Change > CS_INIT -> CS_ROUTING > > d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:20.980418 [DEBUG] > switch_core_session.c:1351 Send signal sofia/external/sip:203 at 198.17. > 240.238:5062 [BREAK] > > d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:20.980418 [DEBUG] > switch_core_state_machine.c:455 (sofia/external/sip:203 at 198. > 17.240.238:5062) State INIT going to sleep > > d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:20.980418 [DEBUG] > switch_core_state_machine.c:415 (sofia/external/sip:203 at 198. > 17.240.238:5062) Running State Change CS_ROUTING > > d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:20.980418 [DEBUG] > sofia.c:5815 Channel sofia/external/sip:203 at 198.17.240.238:5062 entering > state [calling][0] > > d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:20.980418 [DEBUG] > switch_core_state_machine.c:471 (sofia/external/sip:203 at 198. > 17.240.238:5062) State ROUTING > > d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:20.980418 [DEBUG] > mod_sofia.c:150 sofia/external/sip:203 at 198.17.240.238:5062 SOFIA ROUTING > > d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:20.980418 [DEBUG] > switch_ivr_originate.c:67 (sofia/external/sip:203 at 198.17.240.238:5062) > State Change CS_ROUTING -> CS_CONSUME_MEDIA > > d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:20.980418 [DEBUG] > switch_core_session.c:1351 Send signal sofia/external/sip:203 at 198.17. > 240.238:5062 [BREAK] > > d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:20.980418 [DEBUG] > switch_core_state_machine.c:471 (sofia/external/sip:203 at 198. > 17.240.238:5062) State ROUTING going to sleep > > d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:20.980418 [DEBUG] > switch_core_state_machine.c:415 (sofia/external/sip:203 at 198. > 17.240.238:5062) Running State Change CS_CONSUME_MEDIA > > d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:20.980418 [DEBUG] > switch_core_state_machine.c:490 (sofia/external/sip:203 at 198. > 17.240.238:5062) State CONSUME_MEDIA > > d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:20.980418 [DEBUG] > switch_core_state_machine.c:490 (sofia/external/sip:203 at 198. > 17.240.238:5062) State CONSUME_MEDIA going to sleep > > d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:21.040403 [DEBUG] > switch_core_session.c:1016 Send signal sofia/external/sip:203 at 198.17. > 240.238:5062 [BREAK] > > d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:21.040403 [DEBUG] > switch_core_session.c:1016 Send signal sofia/external/sip:203 at 198.17. > 240.238:5062 [BREAK] > > d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:21.040403 [DEBUG] > switch_core_session.c:1016 Send signal sofia/external/sip:203 at 198.17. > 240.238:5062 [BREAK] > > d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:21.040403 [DEBUG] > sofia.c:5815 Channel sofia/external/sip:203 at 198.17.240.238:5062 entering > state [terminated][488] > > d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:21.040403 [NOTICE] > sofia.c:6659 Hangup sofia/external/sip:203 at 198.17.240.238:5062 > [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] > > d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:21.040403 [DEBUG] > switch_channel.c:3187 Send signal sofia/external/sip:203 at 198.17. > 240.238:5062 [KILL] > > d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:21.040403 [DEBUG] > switch_core_session.c:1351 Send signal sofia/external/sip:203 at 198.17. > 240.238:5062 [BREAK] > > d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:21.040403 [DEBUG] > switch_core_state_machine.c:415 (sofia/external/sip:203 at 198. > 17.240.238:5062) Running State Change CS_HANGUP > > d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:21.040403 [DEBUG] > switch_core_state_machine.c:678 (sofia/external/sip:203 at 198. > 17.240.238:5062) Callstate Change DOWN -> HANGUP > > d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:21.040403 [DEBUG] > switch_core_state_machine.c:680 (sofia/external/sip:203 at 198. > 17.240.238:5062) State HANGUP > > d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:21.040403 [DEBUG] > mod_sofia.c:506 Channel sofia/external/sip:203 at 198.17.240.238:5062 > hanging up, cause: INCOMPATIBLE_DESTINATION > > d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:21.040403 [DEBUG] > switch_core_state_machine.c:48 sofia/external/sip:203 at 198.17.240.238:5062 > Standard HANGUP, cause: INCOMPATIBLE_DESTINATION > > d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:21.040403 [DEBUG] > switch_core_state_machine.c:680 (sofia/external/sip:203 at 198. > 17.240.238:5062) State HANGUP going to sleep > > d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:21.040403 [DEBUG] > switch_core_state_machine.c:447 (sofia/external/sip:203 at 198. > 17.240.238:5062) State Change CS_HANGUP -> CS_REPORTING > > d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:21.040403 [DEBUG] > switch_core_session.c:1351 Send signal sofia/external/sip:203 at 198.17. > 240.238:5062 [BREAK] > > d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:21.040403 [DEBUG] > switch_core_state_machine.c:415 (sofia/external/sip:203 at 198. > 17.240.238:5062) Running State Change CS_REPORTING > > d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:21.040403 [DEBUG] > switch_core_state_machine.c:764 (sofia/external/sip:203 at 198. > 17.240.238:5062) State REPORTING > > d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:21.040403 [DEBUG] > switch_core_state_machine.c:92 sofia/external/sip:203 at 198.17.240.238:5062 > Standard REPORTING, cause: INCOMPATIBLE_DESTINATION > > d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:21.040403 [DEBUG] > switch_core_state_machine.c:764 (sofia/external/sip:203 at 198. > 17.240.238:5062) State REPORTING going to sleep > > d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:21.040403 [DEBUG] > switch_core_state_machine.c:441 (sofia/external/sip:203 at 198. > 17.240.238:5062) State Change CS_REPORTING -> CS_DESTROY > > d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:21.040403 [DEBUG] > switch_core_session.c:1351 Send signal sofia/external/sip:203 at 198.17. > 240.238:5062 [BREAK] > > d12984f7-12ec-430f-99a7-43202963672a 2017-05-01 09:37:21.040403 [DEBUG] > switch_core_session.c:1559 Session 223 (sofia/external/sip:203 at 198. > 17.240.238:5062) Locked, Waiting on external entities > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Guillermo Ruiz Camauer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170501/cdd8d559/attachment-0001.html From rbetancor at gmail.com Mon May 1 23:54:35 2017 From: rbetancor at gmail.com (=?UTF-8?Q?Ra=C3=BAl_Alexis_Betancor_Santana?=) Date: Mon, 1 May 2017 20:54:35 +0100 Subject: [Freeswitch-users] OPUS Codec In-Reply-To: References: Message-ID: 30% of packet loss ... are you mad? ... there is no way any codec could recover from that. Most of codecs, you will notice whenever you have more than 1-3% packet loss ... 30% ? ... really? ... should sound like and old synthetizer 2017-05-01 20:03 GMT+01:00 Colton Conor : > Does freeswitch support transcoding from OPUS to G711? How well does this > work in theory? Having full HD come in via OPUS from the phone, and then > transcoding to crappy G711. Does it sound funny? > > We are thinking that OPUS has superior audio quality, and can withstand up > to 30 percent packet loss without any audio distorition. However, I would > say the majority of our users calls as still to the PSTN which only speaks > G711. > > How much processing power would it take to transcode from OPUS to G711? > > How well is the OPUS codec implemented on newer IP phones? > > I hear the Yealink supports OPUS with their new S line of phones, but I > also heard they pulled support for it? > > Polycom seems to only support it on the VVX 500 and VVX 600, but you have > to disable video according to the 5.4 release notes. OPUS must be processor > intensive I assume if it can't run OPUS and video at the same time. Have > any idea if the newer VVX, like the 411, 501, and 601 have this same > limitation? I know they have faster processors and ram. > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170501/a48cd8e8/attachment.html From colin.morelli at gmail.com Tue May 2 00:07:43 2017 From: colin.morelli at gmail.com (Colin Morelli) Date: Mon, 1 May 2017 16:07:43 -0400 Subject: [Freeswitch-users] OPUS Codec In-Reply-To: References: Message-ID: Raul, I'm sure the OP was referring to 30% packet loss with FEC, which opus can turn back into a usable audio stream. You'll absolutely notice artifacts in the audio, and you'd clearly identify something is wrong, but it's a heck of a lot better than 30% packet loss without FEC. Colton, Freeswitch supports opus, yes. I'd be curious to know why you want to transcode between opus and G.711, unless your endpoints don't support G.711 though. Yes, FEC can help you with packet loss on one end, but if your packet loss is on the G.711 end, then you're SOL anyway, so it seems pointless to do this unless you can use opus all the way though. As to how it will sound if doing the conversion, well, you're transcoding from one lossy codec to another lossy codec. The end result will be audio that will not sound as good as if it were just encoded in G.711 to begin with. Remember, the nature of lossy codecs is that they're, well, lossy. You can't make a G.711 stream sound better by re-encoding it to opus. Once it's encoded in G.711, that information is lost. As for CPU usage, opus is not particularly cheap. Transcoding + resampling (if required), will considerably limit the number of concurrent calls your FS instance can handle. Actual results are going to depend on a number of factors that make it infeasible to talk about it. But again, you probably don't want to be transcoding between these two codecs unless you absolutely have two (i.e. you can't get the two endpoints to otherwise agree on a codec). You're not going to gain any real advantage from doing this. Best, Colin On Mon, May 1, 2017 at 3:54 PM, Ra?l Alexis Betancor Santana < rbetancor at gmail.com> wrote: > 30% of packet loss ... are you mad? ... there is no way any codec could > recover from that. Most of codecs, you will notice whenever you have more > than 1-3% packet loss ... 30% ? ... really? ... should sound like and old > synthetizer > > 2017-05-01 20:03 GMT+01:00 Colton Conor : > >> Does freeswitch support transcoding from OPUS to G711? How well does this >> work in theory? Having full HD come in via OPUS from the phone, and then >> transcoding to crappy G711. Does it sound funny? >> >> We are thinking that OPUS has superior audio quality, and can withstand >> up to 30 percent packet loss without any audio distorition. However, I >> would say the majority of our users calls as still to the PSTN which only >> speaks G711. >> >> How much processing power would it take to transcode from OPUS to G711? >> >> How well is the OPUS codec implemented on newer IP phones? >> >> I hear the Yealink supports OPUS with their new S line of phones, but I >> also heard they pulled support for it? >> >> Polycom seems to only support it on the VVX 500 and VVX 600, but you have >> to disable video according to the 5.4 release notes. OPUS must be processor >> intensive I assume if it can't run OPUS and video at the same time. Have >> any idea if the newer VVX, like the 411, 501, and 601 have this same >> limitation? I know they have faster processors and ram. >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170501/f6996870/attachment.html From mike at jerris.com Tue May 2 00:11:28 2017 From: mike at jerris.com (Michael Jerris) Date: Mon, 1 May 2017 16:11:28 -0400 Subject: [Freeswitch-users] Outbound calls being dropped after update to Version 1.6.17 -34-0fc0946 64bit In-Reply-To: <001d01d2c2af$12244d40$366ce7c0$@accra.ca> References: <001d01d2c2af$12244d40$366ce7c0$@accra.ca> Message-ID: you can?t tell just from the data provided, but these issues are usually nat related. Inspect the IP addrs and ports in the sip packet and on the wire and see if all looks right. > On May 1, 2017, at 3:13 PM, Charles Bujold wrote: > > Hi, > > Getting a strange issue, all incoming calls work great, but outgoing calls are getting dropped. > > Initially though that it was a NAT issue but verified the NAT and it is working properly. The proper external IP is being used and the gateway is registering with the service provider without issue. > > Checked the call flow with sngrep and wireshark and found some strange issues. > > From my end, making the outbound call I see the call being connected and I receive a 200 OK that the call is connected and I have a connection. It will last about 1.5 minutes while at the same time I see 5-6 INVITE being sent from my Freeswitch to my service provider and then a BYE that it is hanging up. > > From my service provider, they see the call established and then: SIP/SDP 82 Request: INVITE sip: telephonnumber at IPxxxxxx:5060; transport=udp, in-dialog > > The question is what is the ?in-dialog? it seems to be causing the call to not be recognize that it is active and INVITE being generated and not being answered which causes the Bye Command to occur. > > Not certain how to fix. Any suggestions would be appreciated. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170501/24655075/attachment-0001.html From mike at jerris.com Tue May 2 00:13:14 2017 From: mike at jerris.com (Michael Jerris) Date: Mon, 1 May 2017 16:13:14 -0400 Subject: [Freeswitch-users] OPUS Codec In-Reply-To: References: Message-ID: <0AB37249-513F-477D-B90D-4C0748DE0CBC@jerris.com> I disagree? plenty of situations where you have a known quality leg to a provider over g711, and a uncontrolled internet leg to customer that you want to use opus/fec for. > On May 1, 2017, at 4:07 PM, Colin Morelli wrote: > > Raul, I'm sure the OP was referring to 30% packet loss with FEC, which opus can turn back into a usable audio stream. You'll absolutely notice artifacts in the audio, and you'd clearly identify something is wrong, but it's a heck of a lot better than 30% packet loss without FEC. > > Colton, > > Freeswitch supports opus, yes. I'd be curious to know why you want to transcode between opus and G.711, unless your endpoints don't support G.711 though. Yes, FEC can help you with packet loss on one end, but if your packet loss is on the G.711 end, then you're SOL anyway, so it seems pointless to do this unless you can use opus all the way though. > > As to how it will sound if doing the conversion, well, you're transcoding from one lossy codec to another lossy codec. The end result will be audio that will not sound as good as if it were just encoded in G.711 to begin with. Remember, the nature of lossy codecs is that they're, well, lossy. You can't make a G.711 stream sound better by re-encoding it to opus. Once it's encoded in G.711, that information is lost. > > As for CPU usage, opus is not particularly cheap. Transcoding + resampling (if required), will considerably limit the number of concurrent calls your FS instance can handle. Actual results are going to depend on a number of factors that make it infeasible to talk about it. But again, you probably don't want to be transcoding between these two codecs unless you absolutely have two (i.e. you can't get the two endpoints to otherwise agree on a codec). You're not going to gain any real advantage from doing this. > > Best, > Colin > > On Mon, May 1, 2017 at 3:54 PM, Ra?l Alexis Betancor Santana > wrote: > 30% of packet loss ... are you mad? ... there is no way any codec could recover from that. Most of codecs, you will notice whenever you have more than 1-3% packet loss ... 30% ? ... really? ... should sound like and old synthetizer > > 2017-05-01 20:03 GMT+01:00 Colton Conor >: > Does freeswitch support transcoding from OPUS to G711? How well does this work in theory? Having full HD come in via OPUS from the phone, and then transcoding to crappy G711. Does it sound funny? > > We are thinking that OPUS has superior audio quality, and can withstand up to 30 percent packet loss without any audio distorition. However, I would say the majority of our users calls as still to the PSTN which only speaks G711. > > How much processing power would it take to transcode from OPUS to G711? > > How well is the OPUS codec implemented on newer IP phones? > > I hear the Yealink supports OPUS with their new S line of phones, but I also heard they pulled support for it? > > Polycom seems to only support it on the VVX 500 and VVX 600, but you have to disable video according to the 5.4 release notes. OPUS must be processor intensive I assume if it can't run OPUS and video at the same time. Have any idea if the newer VVX, like the 411, 501, and 601 have this same limitation? I know they have faster processors and ram. > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170501/809f8126/attachment.html From colin.morelli at gmail.com Tue May 2 00:19:24 2017 From: colin.morelli at gmail.com (Colin Morelli) Date: Mon, 1 May 2017 16:19:24 -0400 Subject: [Freeswitch-users] OPUS Codec In-Reply-To: <0AB37249-513F-477D-B90D-4C0748DE0CBC@jerris.com> References: <0AB37249-513F-477D-B90D-4C0748DE0CBC@jerris.com> Message-ID: That's fair, if the primary focus of using the codec is purely to overcome a packet loss issue; however, consistent 30% packet loss to your endpoints seems like a fairly extreme scenario. I suppose if you anticipate a high enough level of packet loss on the average call to justify the slightly degraded audio quality that will be introduced by transcoding, then it may be worthwhile. I'll stand by the rest of my points though! Best, Colin On Mon, May 1, 2017 at 4:13 PM, Michael Jerris wrote: > I disagree? plenty of situations where you have a known quality leg to a > provider over g711, and a uncontrolled internet leg to customer that you > want to use opus/fec for. > > On May 1, 2017, at 4:07 PM, Colin Morelli wrote: > > Raul, I'm sure the OP was referring to 30% packet loss with FEC, which > opus can turn back into a usable audio stream. You'll absolutely notice > artifacts in the audio, and you'd clearly identify something is wrong, but > it's a heck of a lot better than 30% packet loss without FEC. > > Colton, > > Freeswitch supports opus, yes. I'd be curious to know why you want to > transcode between opus and G.711, unless your endpoints don't support G.711 > though. Yes, FEC can help you with packet loss on one end, but if your > packet loss is on the G.711 end, then you're SOL anyway, so it seems > pointless to do this unless you can use opus all the way though. > > As to how it will sound if doing the conversion, well, you're transcoding > from one lossy codec to another lossy codec. The end result will be audio > that will not sound as good as if it were just encoded in G.711 to begin > with. Remember, the nature of lossy codecs is that they're, well, lossy. > You can't make a G.711 stream sound better by re-encoding it to opus. Once > it's encoded in G.711, that information is lost. > > As for CPU usage, opus is not particularly cheap. Transcoding + resampling > (if required), will considerably limit the number of concurrent calls your > FS instance can handle. Actual results are going to depend on a number of > factors that make it infeasible to talk about it. But again, you probably > don't want to be transcoding between these two codecs unless you absolutely > have two (i.e. you can't get the two endpoints to otherwise agree on a > codec). You're not going to gain any real advantage from doing this. > > Best, > Colin > > On Mon, May 1, 2017 at 3:54 PM, Ra?l Alexis Betancor Santana < > rbetancor at gmail.com> wrote: > >> 30% of packet loss ... are you mad? ... there is no way any codec could >> recover from that. Most of codecs, you will notice whenever you have more >> than 1-3% packet loss ... 30% ? ... really? ... should sound like and old >> synthetizer >> >> 2017-05-01 20:03 GMT+01:00 Colton Conor : >> >>> Does freeswitch support transcoding from OPUS to G711? How well does >>> this work in theory? Having full HD come in via OPUS from the phone, and >>> then transcoding to crappy G711. Does it sound funny? >>> >>> We are thinking that OPUS has superior audio quality, and can withstand >>> up to 30 percent packet loss without any audio distorition. However, I >>> would say the majority of our users calls as still to the PSTN which only >>> speaks G711. >>> >>> How much processing power would it take to transcode from OPUS to G711? >>> >>> How well is the OPUS codec implemented on newer IP phones? >>> >>> I hear the Yealink supports OPUS with their new S line of phones, but I >>> also heard they pulled support for it? >>> >>> Polycom seems to only support it on the VVX 500 and VVX 600, but you >>> have to disable video according to the 5.4 release notes. OPUS must be >>> processor intensive I assume if it can't run OPUS and video at the same >>> time. Have any idea if the newer VVX, like the 411, 501, and 601 have this >>> same limitation? I know they have faster processors and ram. >>> >>> >>> >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170501/cd8cf78f/attachment-0001.html From colton.conor at gmail.com Tue May 2 01:21:51 2017 From: colton.conor at gmail.com (Colton Conor) Date: Mon, 1 May 2017 16:21:51 -0500 Subject: [Freeswitch-users] OPUS Codec In-Reply-To: References: <0AB37249-513F-477D-B90D-4C0748DE0CBC@jerris.com> Message-ID: I am just judging the 30 percent packet loss with FEC. Example are on the OPUS website https://opus-codec.org/examples/ Sounds pretty good to me. I would be terrified to know what G711 sounds like without FEC having evien 10 percent packet loss. I think there are many cases where one leg of the audio is uncontrolled, and the other leg of the audio via G711 is stable. For example, a freeswitch server sitting in a data center with redundant gigabit internet connections connected to a origination/termination carrier via G711. Then clients using OTT internet connections like their home cable connection, LTE, or wifi. The clients are the leg I want in OPUS to communicate back to the freeswitch server. Yes, the clients are able to speak G711 too, but I want OPUS for the FEC and bandwidth control. Are you saying transcoding from OPUS client to G711 is going to sound worse than just talking G711 all the way through? So what can a new Dell server with most recent gen intel xeon processors handle as far as transcoding goes? On Mon, May 1, 2017 at 3:19 PM, Colin Morelli wrote: > That's fair, if the primary focus of using the codec is purely to overcome > a packet loss issue; however, consistent 30% packet loss to your endpoints > seems like a fairly extreme scenario. > > I suppose if you anticipate a high enough level of packet loss on the > average call to justify the slightly degraded audio quality that will be > introduced by transcoding, then it may be worthwhile. > > I'll stand by the rest of my points though! > > Best, > Colin > > On Mon, May 1, 2017 at 4:13 PM, Michael Jerris wrote: > >> I disagree? plenty of situations where you have a known quality leg to a >> provider over g711, and a uncontrolled internet leg to customer that you >> want to use opus/fec for. >> >> On May 1, 2017, at 4:07 PM, Colin Morelli >> wrote: >> >> Raul, I'm sure the OP was referring to 30% packet loss with FEC, which >> opus can turn back into a usable audio stream. You'll absolutely notice >> artifacts in the audio, and you'd clearly identify something is wrong, but >> it's a heck of a lot better than 30% packet loss without FEC. >> >> Colton, >> >> Freeswitch supports opus, yes. I'd be curious to know why you want to >> transcode between opus and G.711, unless your endpoints don't support G.711 >> though. Yes, FEC can help you with packet loss on one end, but if your >> packet loss is on the G.711 end, then you're SOL anyway, so it seems >> pointless to do this unless you can use opus all the way though. >> >> As to how it will sound if doing the conversion, well, you're transcoding >> from one lossy codec to another lossy codec. The end result will be audio >> that will not sound as good as if it were just encoded in G.711 to begin >> with. Remember, the nature of lossy codecs is that they're, well, lossy. >> You can't make a G.711 stream sound better by re-encoding it to opus. Once >> it's encoded in G.711, that information is lost. >> >> As for CPU usage, opus is not particularly cheap. Transcoding + >> resampling (if required), will considerably limit the number of concurrent >> calls your FS instance can handle. Actual results are going to depend on a >> number of factors that make it infeasible to talk about it. But again, you >> probably don't want to be transcoding between these two codecs unless you >> absolutely have two (i.e. you can't get the two endpoints to otherwise >> agree on a codec). You're not going to gain any real advantage from doing >> this. >> >> Best, >> Colin >> >> On Mon, May 1, 2017 at 3:54 PM, Ra?l Alexis Betancor Santana < >> rbetancor at gmail.com> wrote: >> >>> 30% of packet loss ... are you mad? ... there is no way any codec could >>> recover from that. Most of codecs, you will notice whenever you have more >>> than 1-3% packet loss ... 30% ? ... really? ... should sound like and old >>> synthetizer >>> >>> 2017-05-01 20:03 GMT+01:00 Colton Conor : >>> >>>> Does freeswitch support transcoding from OPUS to G711? How well does >>>> this work in theory? Having full HD come in via OPUS from the phone, and >>>> then transcoding to crappy G711. Does it sound funny? >>>> >>>> We are thinking that OPUS has superior audio quality, and can withstand >>>> up to 30 percent packet loss without any audio distorition. However, I >>>> would say the majority of our users calls as still to the PSTN which only >>>> speaks G711. >>>> >>>> How much processing power would it take to transcode from OPUS to G711? >>>> >>>> How well is the OPUS codec implemented on newer IP phones? >>>> >>>> I hear the Yealink supports OPUS with their new S line of phones, but I >>>> also heard they pulled support for it? >>>> >>>> Polycom seems to only support it on the VVX 500 and VVX 600, but you >>>> have to disable video according to the 5.4 release notes. OPUS must be >>>> processor intensive I assume if it can't run OPUS and video at the same >>>> time. Have any idea if the newer VVX, like the 411, 501, and 601 have this >>>> same limitation? I know they have faster processors and ram. >>>> >>>> >>>> >>>> >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-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 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170501/46626e35/attachment.html From colin.morelli at gmail.com Tue May 2 01:39:51 2017 From: colin.morelli at gmail.com (Colin Morelli) Date: Mon, 1 May 2017 17:39:51 -0400 Subject: [Freeswitch-users] OPUS Codec In-Reply-To: References: <0AB37249-513F-477D-B90D-4C0748DE0CBC@jerris.com> Message-ID: I agree, and was wrong before to suggest that there's no real advantage. FEC can be helpful to combat packet loss on unpredictable networks. Although again, I'll stand by my statement that consistent 10% packet loss would be pretty significant on any modern internet connection. Even consistent 2% packet loss would be cause for some concern. Intermittent packet loss issues, sure; but sustained 10% packet loss would be fairly alarming to hear about on any large scale. At that level, it's likely the user has problems with a number of services (even TCP ones, as they're likely to be experiencing slower speeds). On a consumer internet connection, your more frequent issues are almost certainly going to be latency and jitter, and generally your first line of defense (at least for jitter) would be the endpoints' jitter buffers. Although FEC can also help if you have to start dropping packets from the jitter buffer. Again, I'm not trying to claim it can't or won't happen. It can, and opus will help when it does. I'm just saying it probably makes sense to actually confirm this is the issue (if you're currently experiencing problems). Maybe my experiences have been different. *Are you saying transcoding from OPUS client to G711 is going to sound worse than just talking G711 all the way through? * Yes. Both codecs are lossy. Any time you encode to a lossy codec, you lose data. In this case, you're doing it twice. The originating device will be encoding raw audio to G.711, and FS will decode the G.711 and re-encode as opus. The degradation on the transcoding won't be huge, and it many cases it may not even really be noticeable (you'd have to test this in your own setup to see), but it will exist. I won't try to answer the question of how many transcoding channels any particular server can handle though. Perhaps the FS guys will have a decent answer, but I think they'll probably give similar feedback - you just need to test and find out. There are so many factors that come into play with performance that it becomes painfully difficult to try to guess. Best, Colin On Mon, May 1, 2017 at 5:21 PM, Colton Conor wrote: > I am just judging the 30 percent packet loss with FEC. Example are on the > OPUS website https://opus-codec.org/examples/ Sounds pretty good to me. I > would be terrified to know what G711 sounds like without FEC having evien > 10 percent packet loss. > > I think there are many cases where one leg of the audio is uncontrolled, > and the other leg of the audio via G711 is stable. For example, a > freeswitch server sitting in a data center with redundant gigabit internet > connections connected to a origination/termination carrier via G711. > > Then clients using OTT internet connections like their home cable > connection, LTE, or wifi. The clients are the leg I want in OPUS to > communicate back to the freeswitch server. Yes, the clients are able to > speak G711 too, but I want OPUS for the FEC and bandwidth control. > > Are you saying transcoding from OPUS client to G711 is going to sound > worse than just talking G711 all the way through? > > So what can a new Dell server with most recent gen intel xeon processors > handle as far as transcoding goes? > > On Mon, May 1, 2017 at 3:19 PM, Colin Morelli > wrote: > >> That's fair, if the primary focus of using the codec is purely to >> overcome a packet loss issue; however, consistent 30% packet loss to your >> endpoints seems like a fairly extreme scenario. >> >> I suppose if you anticipate a high enough level of packet loss on the >> average call to justify the slightly degraded audio quality that will be >> introduced by transcoding, then it may be worthwhile. >> >> I'll stand by the rest of my points though! >> >> Best, >> Colin >> >> On Mon, May 1, 2017 at 4:13 PM, Michael Jerris wrote: >> >>> I disagree? plenty of situations where you have a known quality leg to a >>> provider over g711, and a uncontrolled internet leg to customer that you >>> want to use opus/fec for. >>> >>> On May 1, 2017, at 4:07 PM, Colin Morelli >>> wrote: >>> >>> Raul, I'm sure the OP was referring to 30% packet loss with FEC, which >>> opus can turn back into a usable audio stream. You'll absolutely notice >>> artifacts in the audio, and you'd clearly identify something is wrong, but >>> it's a heck of a lot better than 30% packet loss without FEC. >>> >>> Colton, >>> >>> Freeswitch supports opus, yes. I'd be curious to know why you want to >>> transcode between opus and G.711, unless your endpoints don't support G.711 >>> though. Yes, FEC can help you with packet loss on one end, but if your >>> packet loss is on the G.711 end, then you're SOL anyway, so it seems >>> pointless to do this unless you can use opus all the way though. >>> >>> As to how it will sound if doing the conversion, well, you're >>> transcoding from one lossy codec to another lossy codec. The end result >>> will be audio that will not sound as good as if it were just encoded in >>> G.711 to begin with. Remember, the nature of lossy codecs is that they're, >>> well, lossy. You can't make a G.711 stream sound better by re-encoding it >>> to opus. Once it's encoded in G.711, that information is lost. >>> >>> As for CPU usage, opus is not particularly cheap. Transcoding + >>> resampling (if required), will considerably limit the number of concurrent >>> calls your FS instance can handle. Actual results are going to depend on a >>> number of factors that make it infeasible to talk about it. But again, you >>> probably don't want to be transcoding between these two codecs unless you >>> absolutely have two (i.e. you can't get the two endpoints to otherwise >>> agree on a codec). You're not going to gain any real advantage from doing >>> this. >>> >>> Best, >>> Colin >>> >>> On Mon, May 1, 2017 at 3:54 PM, Ra?l Alexis Betancor Santana < >>> rbetancor at gmail.com> wrote: >>> >>>> 30% of packet loss ... are you mad? ... there is no way any codec could >>>> recover from that. Most of codecs, you will notice whenever you have more >>>> than 1-3% packet loss ... 30% ? ... really? ... should sound like and old >>>> synthetizer >>>> >>>> 2017-05-01 20:03 GMT+01:00 Colton Conor : >>>> >>>>> Does freeswitch support transcoding from OPUS to G711? How well does >>>>> this work in theory? Having full HD come in via OPUS from the phone, and >>>>> then transcoding to crappy G711. Does it sound funny? >>>>> >>>>> We are thinking that OPUS has superior audio quality, and can >>>>> withstand up to 30 percent packet loss without any audio distorition. >>>>> However, I would say the majority of our users calls as still to the PSTN >>>>> which only speaks G711. >>>>> >>>>> How much processing power would it take to transcode from OPUS to G711? >>>>> >>>>> How well is the OPUS codec implemented on newer IP phones? >>>>> >>>>> I hear the Yealink supports OPUS with their new S line of phones, but >>>>> I also heard they pulled support for it? >>>>> >>>>> Polycom seems to only support it on the VVX 500 and VVX 600, but you >>>>> have to disable video according to the 5.4 release notes. OPUS must be >>>>> processor intensive I assume if it can't run OPUS and video at the same >>>>> time. Have any idea if the newer VVX, like the 411, 501, and 601 have this >>>>> same limitation? I know they have faster processors and ram. >>>>> >>>>> >>>>> >>>>> >>>>> ____________________________________________________________ >>>>> _____________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-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 >>> >>> >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170501/7fab8b5a/attachment-0001.html From sdevoy at bizfocused.com Tue May 2 02:08:44 2017 From: sdevoy at bizfocused.com (Sean Devoy) Date: Mon, 1 May 2017 22:08:44 +0000 Subject: [Freeswitch-users] Upgrade path? Message-ID: Hi, I can finally upgrade my config to latest stable!!! I have version 1.2.22 (really). Can I just upgrade to the latest build? Have the config files and folders all changed? I am running multi-tenant if that matters? Recommendations? Sean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170501/0383acf6/attachment.html From s.safarov at gmail.com Tue May 2 05:19:23 2017 From: s.safarov at gmail.com (Sergey Safarov) Date: Tue, 02 May 2017 01:19:23 +0000 Subject: [Freeswitch-users] Outbound calls being dropped after update to Version 1.6.17 -34-0fc0946 64bit In-Reply-To: References: <001d01d2c2af$12244d40$366ce7c0$@accra.ca> Message-ID: I can confirm that issue related to fragmented UDP packets from carrier equipment. Try swith to TCP transport. ??, 1 ??? 2017, 23:12 Michael Jerris : > you can?t tell just from the data provided, but these issues are usually > nat related. Inspect the IP addrs and ports in the sip packet and on the > wire and see if all looks right. > > On May 1, 2017, at 3:13 PM, Charles Bujold wrote: > > Hi, > > Getting a strange issue, all incoming calls work great, but outgoing calls > are getting dropped. > > Initially though that it was a NAT issue but verified the NAT and it is > working properly. The proper external IP is being used and the gateway is > registering with the service provider without issue. > > Checked the call flow with sngrep and wireshark and found some strange > issues. > > > 1. From my end, making the outbound call I see the call being > connected and I receive a 200 OK that the call is connected and I have a > connection. It will last about 1.5 minutes while at the same time I see > 5-6 INVITE being sent from my Freeswitch to my service provider and then a > BYE that it is hanging up. > > > > 1. From my service provider, they see the call established and > then: SIP/SDP 82 Request: INVITE sip: telephonnumber at IPxxxxxx:5060; > transport=udp, in-dialog > > > The question is what is the ?in-dialog? it seems to be causing the call to > not be recognize that it is active and INVITE being generated and not being > answered which causes the Bye Command to occur. > > Not certain how to fix. Any suggestions would be appreciated. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170502/dd91cab7/attachment.html From sdevoy at bizfocused.com Tue May 2 06:39:59 2017 From: sdevoy at bizfocused.com (Sean Devoy) Date: Tue, 2 May 2017 02:39:59 +0000 Subject: [Freeswitch-users] Looking to hire a consultant Message-ID: Hi, We want to hire someone for a few hours to get a configuration working with FS and Cisco SPA504G phones. We have all the basics working great for FS on SPA phones. We need a single DID number to ring on 3 line buttons on 2 different phones. If a call is placed on hold on phone 1, it must be picked up on phone 2. The customer claims the staff is too stupid to use call parking. I think this really needs to be someone who knows both FS and Cisco SPA phones. There are just too many possible parameter combos for us to work it out. Contact me directly if you can handle the task and have availability soon. Thanks, Sean SDevoy(-at-)bizfocused.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170502/d9441dab/attachment.html From gmaruzz at gmail.com Tue May 2 07:15:23 2017 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Tue, 2 May 2017 05:15:23 +0200 Subject: [Freeswitch-users] Looking to hire a consultant In-Reply-To: References: Message-ID: Please email consulting at freeswitch.org -giovanni On 2 May 2017 at 04:39, Sean Devoy wrote: > Hi, > > > > We want to hire someone for a few hours to get a configuration working > with FS and Cisco SPA504G phones. > > > > We have all the basics working great for FS on SPA phones. We need a > single DID number to ring on 3 line buttons on 2 different phones. If a > call is placed on hold on phone 1, it must be picked up on phone 2. The > customer claims the staff is too stupid to use call parking. > > > > I think this really needs to be someone who knows both FS and Cisco SPA > phones. There are just too many possible parameter combos for us to work it > out. > > > > Contact me directly if you can handle the task and have availability soon. > > > > Thanks, > > Sean > > SDevoy(-at-)bizfocused.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 > -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170502/db8a18fa/attachment-0001.html From sdevoy at bizfocused.com Tue May 2 08:23:58 2017 From: sdevoy at bizfocused.com (Sean Devoy) Date: Tue, 2 May 2017 04:23:58 +0000 Subject: [Freeswitch-users] Looking to hire a consultant In-Reply-To: References: Message-ID: Can?t afford them. I emailed them at least once before. Perhaps I misunderstood, but I came away thinking anything less than $1000 wasn?t worth their time or at least worth the overhead they charge for ?planning?. This is more like a $400 to $500 job max, I can?t spend $300 ?planning.? They wanted hours (2 min I think) of conf calls and planning at a pretty high rate just to talk about what I wanted to do. There would be more money spent ?planning the project? than just doing the work. I am all for specifications and project planning for something that will take more than 4 or 5 hours or is in some way vague, but that is not required on simple jobs. They are just priced out of my reach, I wish they were not but I don?t have hundreds of users to pass this on to. I would rather send the FS team the money than anyone else, but not at that premium. The project is pretty simple: * Install a new FS Stable build on a new Debian 8.7 server * Move/Convert my current XML config over (about 30 extensions) * Help config Cisco SPA phones for SCA. That should not cost $250+ to plan. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Giovanni Maruzzelli Sent: Monday, May 1, 2017 11:15 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Looking to hire a consultant Please email consulting at freeswitch.org -giovanni On 2 May 2017 at 04:39, Sean Devoy > wrote: Hi, We want to hire someone for a few hours to get a configuration working with FS and Cisco SPA504G phones. We have all the basics working great for FS on SPA phones. We need a single DID number to ring on 3 line buttons on 2 different phones. If a call is placed on hold on phone 1, it must be picked up on phone 2. The customer claims the staff is too stupid to use call parking. I think this really needs to be someone who knows both FS and Cisco SPA phones. There are just too many possible parameter combos for us to work it out. Contact me directly if you can handle the task and have availability soon. Thanks, Sean SDevoy(-at-)bizfocused.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 -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170502/5ead0c7b/attachment.html From jaybinks at gmail.com Tue May 2 08:38:42 2017 From: jaybinks at gmail.com (jay binks) Date: Tue, 2 May 2017 14:38:42 +1000 Subject: [Freeswitch-users] Looking to hire a consultant In-Reply-To: References: Message-ID: You are probably looking at a day or 2's worth of work there, maybe more if you consider the risk of things blowing out or providing you support and bug fixes for bits that were missed or not quite working right. Maybe your better off posting your budget to the mailing list and seeing if someone is willing to help you out. Kinda like a bounty ? On 2 May 2017 at 14:23, Sean Devoy wrote: > Can?t afford them. I emailed them at least once before. Perhaps I > misunderstood, but I came away thinking anything less than $1000 wasn?t > worth their time or at least worth the overhead they charge for > ?planning?. This is more like a $400 to $500 job max, I can?t spend $300 > ?planning.? > > > > They wanted hours (2 min I think) of conf calls and planning at a pretty > high rate just to talk about what I wanted to do. There would be more money > spent ?planning the project? than just doing the work. I am all for > specifications and project planning for something that will take more than > 4 or 5 hours or is in some way vague, but that is not required on simple > jobs. They are just priced out of my reach, I wish they were not but I > don?t have hundreds of users to pass this on to. I would rather send the > FS team the money than anyone else, but not at that premium. > > > > The project is pretty simple: > > - Install a new FS Stable build on a new Debian 8.7 server > - Move/Convert my current XML config over (about 30 extensions) > - Help config Cisco SPA phones for SCA. > > > > That should not cost $250+ to plan. > > > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Giovanni > Maruzzelli > *Sent:* Monday, May 1, 2017 11:15 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Looking to hire a consultant > > > > Please email consulting at freeswitch.org > > -giovanni > > > > On 2 May 2017 at 04:39, Sean Devoy wrote: > > Hi, > > > > We want to hire someone for a few hours to get a configuration working > with FS and Cisco SPA504G phones. > > > > We have all the basics working great for FS on SPA phones. We need a > single DID number to ring on 3 line buttons on 2 different phones. If a > call is placed on hold on phone 1, it must be picked up on phone 2. The > customer claims the staff is too stupid to use call parking. > > > > I think this really needs to be someone who knows both FS and Cisco SPA > phones. There are just too many possible parameter combos for us to work it > out. > > > > Contact me directly if you can handle the task and have availability soon. > > > > Thanks, > > Sean > > SDevoy(-at-)bizfocused.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 > > > > > -- > > > Sincerely, > > Giovanni Maruzzelli > OpenTelecom.IT > cell: +39 347 266 56 18 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170502/44b964df/attachment-0001.html From d at d-man.org Tue May 2 08:41:52 2017 From: d at d-man.org (Darren) Date: Tue, 2 May 2017 04:41:52 +0000 Subject: [Freeswitch-users] Looking to hire a consultant In-Reply-To: References: Message-ID: <04533639-FB2B-4C7A-A64F-2374DE844588@d-man.org> I agree with Jay. Your $ might be something someone will take up, but your estimate (or use of the words ?simple?) are way off. If I had to guess: ? Install a new FS Stable build on a new Debian 8.7 server > 30 minutes if everything goes perfectly, 4-8 hours if anything doesn?t. ? Move/Convert my current XML config over (about 30 extensions) > 60-90 minutes figuring out what the heck your current XML even does. Maybe it can just be moved. If not and has to be refactored for some reason (or voicemails retained, media files retained, etc.) then add some time here. I?d guess 2-5 hours. ? Help config Cisco SPA phones for SCA. > Don?t know what this means, manually configure things/etc.? yikes x 30 phones. Conceptually it?s simple. In practice, it?s work. From: on behalf of jay binks Reply-To: FreeSWITCH Users Help Date: Monday, May 1, 2017 at 9:38 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Looking to hire a consultant You are probably looking at a day or 2's worth of work there, maybe more if you consider the risk of things blowing out or providing you support and bug fixes for bits that were missed or not quite working right. Maybe your better off posting your budget to the mailing list and seeing if someone is willing to help you out. Kinda like a bounty ? On 2 May 2017 at 14:23, Sean Devoy > wrote: Can?t afford them. I emailed them at least once before. Perhaps I misunderstood, but I came away thinking anything less than $1000 wasn?t worth their time or at least worth the overhead they charge for ?planning?. This is more like a $400 to $500 job max, I can?t spend $300 ?planning.? They wanted hours (2 min I think) of conf calls and planning at a pretty high rate just to talk about what I wanted to do. There would be more money spent ?planning the project? than just doing the work. I am all for specifications and project planning for something that will take more than 4 or 5 hours or is in some way vague, but that is not required on simple jobs. They are just priced out of my reach, I wish they were not but I don?t have hundreds of users to pass this on to. I would rather send the FS team the money than anyone else, but not at that premium. The project is pretty simple: * Install a new FS Stable build on a new Debian 8.7 server * Move/Convert my current XML config over (about 30 extensions) * Help config Cisco SPA phones for SCA. That should not cost $250+ to plan. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Giovanni Maruzzelli Sent: Monday, May 1, 2017 11:15 PM To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Looking to hire a consultant Please email consulting at freeswitch.org -giovanni On 2 May 2017 at 04:39, Sean Devoy > wrote: Hi, We want to hire someone for a few hours to get a configuration working with FS and Cisco SPA504G phones. We have all the basics working great for FS on SPA phones. We need a single DID number to ring on 3 line buttons on 2 different phones. If a call is placed on hold on phone 1, it must be picked up on phone 2. The customer claims the staff is too stupid to use call parking. I think this really needs to be someone who knows both FS and Cisco SPA phones. There are just too many possible parameter combos for us to work it out. Contact me directly if you can handle the task and have availability soon. Thanks, Sean SDevoy(-at-)bizfocused.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 -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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 Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170502/2c49b719/attachment-0001.html From shaun.stokes at itec-support.co.uk Tue May 2 11:16:18 2017 From: shaun.stokes at itec-support.co.uk (Shaun Stokes) Date: Tue, 2 May 2017 07:16:18 +0000 Subject: [Freeswitch-users] Looking to hire a consultant In-Reply-To: References: Message-ID: <6FD2F8B5BB72834E9939AEDF9FB802A901E865D636@mbx-01.sysconfig.co.uk> Have you considered using FusionPBX? It?s essentially a front end for FreeSWITCH, will be much easier to install and configure since most of the configuration is already prepared. FusionPBX have user guides on features such as call parking, the project is open source, flexible and easy to work with and everything is configurable. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sean Devoy Sent: 02 May 2017 05:24 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Looking to hire a consultant Can?t afford them. I emailed them at least once before. Perhaps I misunderstood, but I came away thinking anything less than $1000 wasn?t worth their time or at least worth the overhead they charge for ?planning?. This is more like a $400 to $500 job max, I can?t spend $300 ?planning.? They wanted hours (2 min I think) of conf calls and planning at a pretty high rate just to talk about what I wanted to do. There would be more money spent ?planning the project? than just doing the work. I am all for specifications and project planning for something that will take more than 4 or 5 hours or is in some way vague, but that is not required on simple jobs. They are just priced out of my reach, I wish they were not but I don?t have hundreds of users to pass this on to. I would rather send the FS team the money than anyone else, but not at that premium. The project is pretty simple: * Install a new FS Stable build on a new Debian 8.7 server * Move/Convert my current XML config over (about 30 extensions) * Help config Cisco SPA phones for SCA. That should not cost $250+ to plan. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Giovanni Maruzzelli Sent: Monday, May 1, 2017 11:15 PM To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Looking to hire a consultant Please email consulting at freeswitch.org -giovanni On 2 May 2017 at 04:39, Sean Devoy > wrote: Hi, We want to hire someone for a few hours to get a configuration working with FS and Cisco SPA504G phones. We have all the basics working great for FS on SPA phones. We need a single DID number to ring on 3 line buttons on 2 different phones. If a call is placed on hold on phone 1, it must be picked up on phone 2. The customer claims the staff is too stupid to use call parking. I think this really needs to be someone who knows both FS and Cisco SPA phones. There are just too many possible parameter combos for us to work it out. Contact me directly if you can handle the task and have availability soon. Thanks, Sean SDevoy(-at-)bizfocused.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 -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18 ______________________________________________________________________ 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170502/653c5f50/attachment.html From gb at cm.nl Tue May 2 11:53:10 2017 From: gb at cm.nl (Grant Bagdasarian) Date: Tue, 2 May 2017 07:53:10 +0000 Subject: [Freeswitch-users] mod_distributor from db In-Reply-To: References: Message-ID: <4fa3d198c0034e9cbf17f88d14cf3cc6@cm.nl> Nice! Thanks! From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: maandag 1 mei 2017 18:18 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] mod_distributor from db mod_xml_curl could. On Mon, May 1, 2017 at 9:38 AM, Grant Bagdasarian > wrote: Hello, Is it possible to load the gateways of a sip profile from a database and the distributor config also from a database, using built in modules/functions? Regards, Grant _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170502/47476a59/attachment-0001.html From devang.nathwani31589 at gmail.com Tue May 2 17:06:53 2017 From: devang.nathwani31589 at gmail.com (devang nathwani) Date: Tue, 2 May 2017 18:36:53 +0530 Subject: [Freeswitch-users] RTP handling from different freeswitch Message-ID: Hello, I have my setup at Norway and some of my customers are in USA so i have concern about audio latency. I want the configuration such a way that if USA customer is calling than the rtp should route through my USA freeswitch servers however sip should route through Norway setup. Is there any way i can achieve this? I have tried modifying rtp-ip but i am getting 'DESTINATION_OUT_OF_ORDER' 2017-05-02 08:50:42.650711 [ERR] switch_core_media.c:5843 AUDIO RTP REPORTS ERROR: [Bind Error! 65.11.17.9:25146] 2017-05-02 08:50:42.650711 [NOTICE] switch_core_media.c:5844 Hangup sofia/default/12345679 [CS_CONSUME_MEDIA] [DESTINATION_OUT_OF_ORDER] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170502/f2cf64d4/attachment.html From shaun.stokes at itec-support.co.uk Tue May 2 17:36:00 2017 From: shaun.stokes at itec-support.co.uk (Shaun Stokes) Date: Tue, 2 May 2017 13:36:00 +0000 Subject: [Freeswitch-users] RTP handling from different freeswitch In-Reply-To: References: Message-ID: <6FD2F8B5BB72834E9939AEDF9FB802A901E865D942@mbx-01.sysconfig.co.uk> Although I don?t have the answer to your question, it would make much more sense for your server in the US to handle both SIP and RTP traffic as this will reduce call setup time and points of failure. Presume the reason you want it setup that way is so you can transfer to extensions based in different regions but you could have extensions which are homed on different servers and use a LUA script to route calls between the servers. LUA script are very flexible, we do something similar for routing external numbers between multiple pools of servers. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of devang nathwani Sent: 02 May 2017 14:07 To: FreeSWITCH Users Help Subject: [Freeswitch-users] RTP handling from different freeswitch Hello, I have my setup at Norway and some of my customers are in USA so i have concern about audio latency. I want the configuration such a way that if USA customer is calling than the rtp should route through my USA freeswitch servers however sip should route through Norway setup. Is there any way i can achieve this? I have tried modifying rtp-ip but i am getting 'DESTINATION_OUT_OF_ORDER' 2017-05-02 08:50:42.650711 [ERR] switch_core_media.c:5843 AUDIO RTP REPORTS ERROR: [Bind Error! 65.11.17.9:25146] 2017-05-02 08:50:42.650711 [NOTICE] switch_core_media.c:5844 Hangup sofia/default/12345679 [CS_CONSUME_MEDIA] [DESTINATION_OUT_OF_ORDER] ______________________________________________________________________ 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170502/8f44aca5/attachment.html From snehsach at rediffmail.com Tue May 2 16:25:16 2017 From: snehsach at rediffmail.com (sachin ) Date: 2 May 2017 12:25:16 -0000 Subject: [Freeswitch-users] =?utf-8?q?Setting_Webrtc_Call_using_FreeSwitch?= =?utf-8?q?_=28Windows=29_and_SipMl5?= Message-ID: <20170502122516.17380.qmail@f4mail-235-174.rediffmail.com> Hello All,?I am trying to setup the Webrtc call between 2 SIPML5 client. I am using FreeSWITCH Version 1.6.14 on Windows. I am using internal LAN. My setup is as followsSIPML5<-----> FS v1.6.14) (Windows) <----> SIPML5?I am using Firefox 53.0 browserThe clients are getting registered over wss. I have created self signed certificates.  In var.xml I have set the codecs setting as follows  <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMU,PCMA">  <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA">?Please find attached the log.txt for detail log. When I try to make the call it gives me the error as "mod_sofia.c:2332 CODEC NEGOTIATION ERROR.  SDP:". 2017-05-02 17:51:06.337143 [INFO] mod_dialplan_xml.c:637 Processing 1001 <1001>->1002 in context default2017-05-02 17:51:06.357163 [INFO] switch_ivr_async.c:4166 Bound B-Leg: *1 execute_extension::dx XML features2017-05-02 17:51:06.357163 [INFO] switch_ivr_async.c:4166 Bound B-Leg: *2 record_session::C:/Program Files/FreeSWITCH/recordings/1001.2017-05-02-17-51-06.wav2017-05-02 17:51:06.357163 [INFO] switch_ivr_async.c:4166 Bound B-Leg: *3 execute_extension::cf XML features2017-05-02 17:51:06.366663 [INFO] switch_ivr_async.c:4166 Bound B-Leg: *4 execute_extension::att_xfer XML features2017-05-02 17:51:06.397185 [INFO] switch_core_session.c:2631 Sending early media2017-05-02 17:51:06.397185 [WARNING] switch_core_media.c:3421 NO candidate ACL defined, Defaulting to wan.auto2017-05-02 17:51:06.397185 [ERR] mod_sofia.c:2332 CODEC NEGOTIATION ERROR.  SDP:v=0o=mozilla...THIS_IS_SDPARTA-53.0 1042623354377839900 0 IN IP4 127.0.0.1s=Doubango Telecom - firefoxt=0 0a=sendrecva=fingerprint:sha-256 D2:CB:EC:EC:B9:81:9A:28:2C:24:BD:85:AF:44:89:01:33:42:3A:DC:57:C6:BD:44:AD:D5:58:94:99:CC:2F:E9a=ice-options:tricklea=msid-semantic:WMS *m=audio 62834 UDP/TLS/RTP/SAVPF 109 9 0 8 101c=IN IP4 192.168.1.3a=rtpmap:109 opus/48000/2a=fmtp:109 maxplaybackrate=48000;stereo=1;useinbandfec=1a=rtpmap:9 G722/8000/1a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-15a=candidate:0 1 UDP 2122252543 192.168.1.3 62834 typ hosta=candidate:1 1 UDP 2122187007 192.168.2.5 62835 typ hosta=candidate:0 2 UDP 2122252542 192.168.1.3 62836 typ hosta=candidate:1 2 UDP 2122187006 192.168.2.5 62837 typ hosta=end-of-candidatesa=extmap:1/sendonly urn:ietf:params:rtp-hdrext:ssrc-audio-levela=ice-pwd:e3aca7bfa237d621bbf4f3d709e0ab2ea=ice-ufrag:9b911841a=mid:sdparta_0a=msid:{fb4d37bb-a6b4-40f9-ba27-75f868ec8426} {a8ecd3ca-97c4-4bc8-91b3-98fa5634c298}a=rtcp:62836 IN IP4 192.168.1.3a=rtcp-muxa=setup:actpassa=ssrc:690260622 cname:{b99ab296-c01f-49ed-a57a-c262cb73a13c} 2017-05-02 17:51:06.397185 [NOTICE] switch_channel.c:3514 Hangup sofia/internal/1001 at 192.168.1.3 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]send 892 bytes to wss/[192.168.1.3]:62283 at 12:21:06.446220:   ------------------------------------------------------------------------   SIP/2.0 488 Not Acceptable Here   Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKaFdSEd6k7URebn5TtkvKmi9sOtWbdohr;rport=62283;received=192.168.1.3   Max-Forwards: 70   From: "1001"<sip:1001 at 192.168.1.3>;tag=ns1luRrOuQhBdWIHhFa9   To: <sip:1002 at 192.168.1.3>;tag=c264gFt1Seymr   Call-ID: 69cfc0c1-d7a1-e71d-4fb1-1855a268442c   CSeq: 12200 INVITE   User-Agent: FreeSWITCH-mod_sofia/1.6.14~64bit   Accept: application/sdp   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE   Supported: timer, path, replaces   Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer   Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"   Content-Length: 0   Remote-Party-ID: "1002" <sip:1002 at 192.168.1.3>;party=calling;privacy=off;screen=no    ------------------------------------------------------------------------2017-05-02 17:51:06.446720 [NOTICE] switch_core_session.c:1665 Session 9 (sofia/internal/1001 at 192.168.1.3) Ended2017-05-02 17:51:06.446720 [NOTICE] switch_core_session.c:1669 Close Channel sofia/internal/1001 at 192.168.1.3 [CS_DESTROY]recv 400 bytes from wss/[192.168.1.3]:62283 at 12:21:06.474240:?Please  let me know what could be the issue.?Thanks and Regards?SD  -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170502/692fa711/attachment-0001.html -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: log.txt Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170502/692fa711/attachment-0001.txt From gmaruzz at gmail.com Tue May 2 18:26:33 2017 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Tue, 2 May 2017 16:26:33 +0200 Subject: [Freeswitch-users] TLS SIP packet tracing and visualization Message-ID: Hello fellows, after some experimentation with various tools, I come out with a little shell tool that maybe can be useful to you too. It can only work with non-forward secrecy ciphers, obviously, and only if is started before the client do the initial TLS handshake (eg, just restart the client). Forward secrecy cannot be decrypted after fact, so don't waste effort. An example of ciphers that can be decrypted are the "AES256-SHA" openssl cipher group. You can use ssldump to check what cipher is used by serverhello. Enjoy, make it better, and share it :) #!/bin/bash # brought to you by Giovanni Maruzzelli # SERVERIP="192.168.1.150" SERVERPORT="5061" PRIVKEY="/etc/certs/privkey.pem" STDERR2DEVNULL=" 2>/dev/null " REGEX="notyet" if [ -z "$1" ]; then REGEX="\\\.*" else REGEX="$1" fi FILTER="ssl.app_data and sip matches" FILTER2="$FILTER \"$REGEX\"" FILTER3="'$FILTER2'" ARGUMENT="-i 1 -Y $FILTER3 -E header=y -T fields -e frame.number -e frame.time -e frame.time_delta_displayed -e ip.src -e ip.dst -e sip.Status-Line -e sip.Request-Line -e sip.msg_hdr -l -d tcp.port\=\=5061,sip -o \"ssl.keys_list: $SERVERIP,$SERVERPORT,sip,$PRIVKEY\" $STDERR2DEVNULL | sed -u 's/\t/\n/g' | sed -u '/^$/d' | sed -u 's/^[0-9]*$/\n==&==============================/g'" echo "" echo "NB: if it do not works, edit script so that STDERR2DEVNULL=\" \" and try again" echo "" echo "NB: remember to quote and escape match patterns, using triple slash" echo " eg, for matching 1010 at pbx.example.com, use \"1010 at pbx.example.com \"" echo " eg, for matching anything, use \"\\\\\\.*\"" echo " eg, for matching *98, use \"\\\\\\*98\"" echo "USAGE: $0 \"\\\\\\*98 at pbx.example.com\"" echo "" case "$1" in -help|--help|?) exit 0 ;; *) echo "THIS TIME WE'RE DOING:" echo "tshark $ARGUMENT" echo "" bash -c "tshark $ARGUMENT" ;; esac -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170502/67b516b7/attachment.html From gmaruzz at gmail.com Tue May 2 18:52:14 2017 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Tue, 2 May 2017 16:52:14 +0200 Subject: [Freeswitch-users] TLS SIP packet tracing and visualization In-Reply-To: References: Message-ID: For a cut and paste ready version, that has the correct carriage returns (mangled by mail), check it in FreeSWITCH documentation: https://freeswitch.org/confluence/display/FREESWITCH/Packet+Capture#PacketCapture-TLSwithsharka -giovanni On 2 May 2017 at 16:26, Giovanni Maruzzelli wrote: > Hello fellows, > > after some experimentation with various tools, I come out with a little > shell tool that maybe can be useful to you too. > > It can only work with non-forward secrecy ciphers, obviously, and only if > is started before the client do the initial TLS handshake (eg, just restart > the client). Forward secrecy cannot be decrypted after fact, so don't waste > effort. > > An example of ciphers that can be decrypted are the "AES256-SHA" openssl > cipher group. You can use ssldump to check what cipher is used by > serverhello. > > Enjoy, make it better, and share it :) > > > #!/bin/bash > # brought to you by Giovanni Maruzzelli > # > SERVERIP="192.168.1.150" > SERVERPORT="5061" > PRIVKEY="/etc/certs/privkey.pem" > STDERR2DEVNULL=" 2>/dev/null " > REGEX="notyet" > > if [ -z "$1" ]; then > REGEX="\\\.*" > else > REGEX="$1" > fi > FILTER="ssl.app_data and sip matches" > FILTER2="$FILTER \"$REGEX\"" > FILTER3="'$FILTER2'" > ARGUMENT="-i 1 -Y $FILTER3 -E header=y -T fields -e frame.number -e > frame.time -e frame.time_delta_displayed -e ip.src -e ip.dst -e > sip.Status-Line -e sip.Request-Line -e sip.msg_hdr -l -d > tcp.port\=\=5061,sip -o \"ssl.keys_list: $SERVERIP,$SERVERPORT,sip,$PRIVKEY\" > $STDERR2DEVNULL | sed -u 's/\t/\n/g' | sed -u '/^$/d' | sed -u > 's/^[0-9]*$/\n==&==============================/g'" > > echo "" > echo "NB: if it do not works, edit script so that STDERR2DEVNULL=\" \" and > try again" > echo "" > echo "NB: remember to quote and escape match patterns, using triple slash" > echo " eg, for matching 1010 at pbx.example.com, use \" > 1010 at pbx.example.com\"" > echo " eg, for matching anything, use \"\\\\\\.*\"" > echo " eg, for matching *98, use \"\\\\\\*98\"" > echo "USAGE: $0 \"\\\\\\*98 at pbx.example.com\"" > echo "" > > > case "$1" in > -help|--help|?) > exit 0 > ;; > *) > echo "THIS TIME WE'RE DOING:" > echo "tshark $ARGUMENT" > echo "" > bash -c "tshark $ARGUMENT" > ;; > esac > > > > -- > > Sincerely, > > Giovanni Maruzzelli > OpenTelecom.IT > cell: +39 347 266 56 18 > -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170502/d32a25a9/attachment.html From mike at jerris.com Tue May 2 19:03:14 2017 From: mike at jerris.com (Michael Jerris) Date: Tue, 2 May 2017 11:03:14 -0400 Subject: [Freeswitch-users] Outbound calls being dropped after update to Version 1.6.17 -34-0fc0946 64bit In-Reply-To: References: <001d01d2c2af$12244d40$366ce7c0$@accra.ca> Message-ID: You can not possibly confirm that from the data provided. What you stated is worth a try, but there was not enough information to even indicate that is the issue. > On May 1, 2017, at 9:19 PM, Sergey Safarov wrote: > > I can confirm that issue related to fragmented UDP packets from carrier equipment. > Try swith to TCP transport. > > > ??, 1 ??? 2017, 23:12 Michael Jerris >: > you can?t tell just from the data provided, but these issues are usually nat related. Inspect the IP addrs and ports in the sip packet and on the wire and see if all looks right. > >> On May 1, 2017, at 3:13 PM, Charles Bujold > wrote: >> >> Hi, >> >> Getting a strange issue, all incoming calls work great, but outgoing calls are getting dropped. >> >> Initially though that it was a NAT issue but verified the NAT and it is working properly. The proper external IP is being used and the gateway is registering with the service provider without issue. >> >> Checked the call flow with sngrep and wireshark and found some strange issues. >> >> From my end, making the outbound call I see the call being connected and I receive a 200 OK that the call is connected and I have a connection. It will last about 1.5 minutes while at the same time I see 5-6 INVITE being sent from my Freeswitch to my service provider and then a BYE that it is hanging up. >> >> From my service provider, they see the call established and then: SIP/SDP 82 Request: INVITE sip: telephonnumber at IPxxxxxx:5060; transport=udp, in-dialog >> >> The question is what is the ?in-dialog? it seems to be causing the call to not be recognize that it is active and INVITE being generated and not being answered which causes the Bye Command to occur. >> >> Not certain how to fix. Any suggestions would be appreciated. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170502/88572904/attachment.html From tg-maillistings at level5.de Tue May 2 19:55:48 2017 From: tg-maillistings at level5.de (=?UTF-8?Q?Thorsten_G=c3=b6llner?=) Date: Tue, 2 May 2017 17:55:48 +0200 Subject: [Freeswitch-users] Chat message to unregistered user currently in conference room Message-ID: <29cc048c-9c26-a439-dff5-d32f6b4423de@level5.de> Hi, I tested the chat command via cli to send a *registered* sip user a message. Works fine. (as described here: https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_chat) Is it possible to send a user, who is currently in a conference room on my freeswitch server, a chat message if this user is *not* registered? In the cli I receive "Sent" but the message does not reach the client (check via sip debg log on the client). Thanks in advance, Thorsten From giacomo.vacca at gmail.com Wed May 3 10:59:20 2017 From: giacomo.vacca at gmail.com (Giacomo Vacca) Date: Wed, 03 May 2017 06:59:20 +0000 Subject: [Freeswitch-users] OPUS Codec In-Reply-To: References: <0AB37249-513F-477D-B90D-4C0748DE0CBC@jerris.com> Message-ID: Instead of trying to go HD with Opus, when you know there will be transcoding with G.711, you may consider using Opus at 8 KHz (assuming your devices can). This is supported by FS and in particular advisable if the clients are mobile devices. It will also reduce the bandwidth usage. With that packet loss, FEC becomes crucial. We documented various considerations in this area from a previous project: https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+And+The+Opus+Audio+Codec For what concerns the estimated capacity, I have the same feeling as Colin above and you'll have to do your load testing. Best Regards, Giacomo On 1 May 2017 at 23:39, Colin Morelli wrote: > I agree, and was wrong before to suggest that there's no real advantage. > FEC can be helpful to combat packet loss on unpredictable networks. > Although again, I'll stand by my statement that consistent 10% packet loss > would be pretty significant on any modern internet connection. Even > consistent 2% packet loss would be cause for some concern. Intermittent > packet loss issues, sure; but sustained 10% packet loss would be fairly > alarming to hear about on any large scale. At that level, it's likely the > user has problems with a number of services (even TCP ones, as they're > likely to be experiencing slower speeds). On a consumer internet > connection, your more frequent issues are almost certainly going to be > latency and jitter, and generally your first line of defense (at least for > jitter) would be the endpoints' jitter buffers. Although FEC can also help > if you have to start dropping packets from the jitter buffer. > > Again, I'm not trying to claim it can't or won't happen. It can, and opus > will help when it does. I'm just saying it probably makes sense to actually > confirm this is the issue (if you're currently experiencing problems). > Maybe my experiences have been different. > > *Are you saying transcoding from OPUS client to G711 is going to sound > worse than just talking G711 all the way through? * > > Yes. Both codecs are lossy. Any time you encode to a lossy codec, you lose > data. In this case, you're doing it twice. The originating device will be > encoding raw audio to G.711, and FS will decode the G.711 and re-encode as > opus. The degradation on the transcoding won't be huge, and it many cases > it may not even really be noticeable (you'd have to test this in your own > setup to see), but it will exist. > > I won't try to answer the question of how many transcoding channels any > particular server can handle though. Perhaps the FS guys will have a decent > answer, but I think they'll probably give similar feedback - you just need > to test and find out. There are so many factors that come into play with > performance that it becomes painfully difficult to try to guess. > > Best, > Colin > > On Mon, May 1, 2017 at 5:21 PM, Colton Conor > wrote: > >> I am just judging the 30 percent packet loss with FEC. Example are on the >> OPUS website https://opus-codec.org/examples/ Sounds pretty good to me. >> I would be terrified to know what G711 sounds like without FEC having evien >> 10 percent packet loss. >> >> I think there are many cases where one leg of the audio is uncontrolled, >> and the other leg of the audio via G711 is stable. For example, a >> freeswitch server sitting in a data center with redundant gigabit internet >> connections connected to a origination/termination carrier via G711. >> >> Then clients using OTT internet connections like their home cable >> connection, LTE, or wifi. The clients are the leg I want in OPUS to >> communicate back to the freeswitch server. Yes, the clients are able to >> speak G711 too, but I want OPUS for the FEC and bandwidth control. >> >> Are you saying transcoding from OPUS client to G711 is going to sound >> worse than just talking G711 all the way through? >> >> So what can a new Dell server with most recent gen intel xeon processors >> handle as far as transcoding goes? >> >> On Mon, May 1, 2017 at 3:19 PM, Colin Morelli >> wrote: >> >>> That's fair, if the primary focus of using the codec is purely to >>> overcome a packet loss issue; however, consistent 30% packet loss to your >>> endpoints seems like a fairly extreme scenario. >>> >>> I suppose if you anticipate a high enough level of packet loss on the >>> average call to justify the slightly degraded audio quality that will be >>> introduced by transcoding, then it may be worthwhile. >>> >>> I'll stand by the rest of my points though! >>> >>> Best, >>> Colin >>> >>> On Mon, May 1, 2017 at 4:13 PM, Michael Jerris wrote: >>> >>>> I disagree? plenty of situations where you have a known quality leg to >>>> a provider over g711, and a uncontrolled internet leg to customer that you >>>> want to use opus/fec for. >>>> >>>> On May 1, 2017, at 4:07 PM, Colin Morelli >>>> wrote: >>>> >>>> Raul, I'm sure the OP was referring to 30% packet loss with FEC, which >>>> opus can turn back into a usable audio stream. You'll absolutely notice >>>> artifacts in the audio, and you'd clearly identify something is wrong, but >>>> it's a heck of a lot better than 30% packet loss without FEC. >>>> >>>> Colton, >>>> >>>> Freeswitch supports opus, yes. I'd be curious to know why you want to >>>> transcode between opus and G.711, unless your endpoints don't support G.711 >>>> though. Yes, FEC can help you with packet loss on one end, but if your >>>> packet loss is on the G.711 end, then you're SOL anyway, so it seems >>>> pointless to do this unless you can use opus all the way though. >>>> >>>> As to how it will sound if doing the conversion, well, you're >>>> transcoding from one lossy codec to another lossy codec. The end result >>>> will be audio that will not sound as good as if it were just encoded in >>>> G.711 to begin with. Remember, the nature of lossy codecs is that they're, >>>> well, lossy. You can't make a G.711 stream sound better by re-encoding it >>>> to opus. Once it's encoded in G.711, that information is lost. >>>> >>>> As for CPU usage, opus is not particularly cheap. Transcoding + >>>> resampling (if required), will considerably limit the number of concurrent >>>> calls your FS instance can handle. Actual results are going to depend on a >>>> number of factors that make it infeasible to talk about it. But again, you >>>> probably don't want to be transcoding between these two codecs unless you >>>> absolutely have two (i.e. you can't get the two endpoints to otherwise >>>> agree on a codec). You're not going to gain any real advantage from doing >>>> this. >>>> >>>> Best, >>>> Colin >>>> >>>> On Mon, May 1, 2017 at 3:54 PM, Ra?l Alexis Betancor Santana < >>>> rbetancor at gmail.com> wrote: >>>> >>>>> 30% of packet loss ... are you mad? ... there is no way any codec >>>>> could recover from that. Most of codecs, you will notice whenever you have >>>>> more than 1-3% packet loss ... 30% ? ... really? ... should sound like and >>>>> old synthetizer >>>>> >>>>> 2017-05-01 20:03 GMT+01:00 Colton Conor : >>>>> >>>>>> Does freeswitch support transcoding from OPUS to G711? How well does >>>>>> this work in theory? Having full HD come in via OPUS from the phone, and >>>>>> then transcoding to crappy G711. Does it sound funny? >>>>>> >>>>>> We are thinking that OPUS has superior audio quality, and can >>>>>> withstand up to 30 percent packet loss without any audio distorition. >>>>>> However, I would say the majority of our users calls as still to the PSTN >>>>>> which only speaks G711. >>>>>> >>>>>> How much processing power would it take to transcode from OPUS to >>>>>> G711? >>>>>> >>>>>> How well is the OPUS codec implemented on newer IP phones? >>>>>> >>>>>> I hear the Yealink supports OPUS with their new S line of phones, but >>>>>> I also heard they pulled support for it? >>>>>> >>>>>> Polycom seems to only support it on the VVX 500 and VVX 600, but you >>>>>> have to disable video according to the 5.4 release notes. OPUS must be >>>>>> processor intensive I assume if it can't run OPUS and video at the same >>>>>> time. Have any idea if the newer VVX, like the 411, 501, and 601 have this >>>>>> same limitation? I know they have faster processors and ram. >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> >>>>>> >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> >>>>>> >>>>>> consulting at freeswitch.org >>>>>> >>>>>> >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> >>>>>> >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> http://confluence.freeswitch.org >>>>>> >>>>>> >>>>>> http://www.cluecon.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> >>>>>> >>>>>> FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170503/f6de3065/attachment-0001.html From fs.user at fordior.net Wed May 3 11:16:38 2017 From: fs.user at fordior.net (EL) Date: Wed, 3 May 2017 09:16:38 +0200 Subject: [Freeswitch-users] OPUS Codec In-Reply-To: References: Message-ID: <20170503071638.GA14349@mail.marktcontact.com> Yealink is supporting OPUS on several other models since firmware V81: Quote: "We will support opus on the standard V81 of SIP-T40P/T23P/T23G/T2 1(P) E2/T19(P) E2." Source: http://forum.yealink.com/forum/showthread.php?tid=29650&pid=51262&mode=threaded I can confirm OPUS implementation on the 'T21P E2' model. -- EL From hardikitpl at gmail.com Wed May 3 14:17:41 2017 From: hardikitpl at gmail.com (Hardik Patel) Date: Wed, 3 May 2017 15:47:41 +0530 Subject: [Freeswitch-users] Fax receive with INCOMPATIBLE_DESTINATION Message-ID: Hello, I am using opensips as entry point using dispatcher to opensips( 127.0.0.1), i am routing call to freeswitch server (127.0.0.2). Now I am trying to send fax using span_dsp modules, my issue is when i receive a fax perfectly but in fs_cli i receive 'INCOMPATIBLE_DESTINATION' ,i need to 'NORMAL_CLEARING'. 127.0.0.3 => carrier/provider IP 98989898 => Fax number and here i am attaching sip trace. -- Hardik Patel iNextrix Technologies Pvt Ltd -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170503/02b80d4b/attachment.html -------------- next part -------------- 2017/05/03 10:43:20.691002 127.0.0.2:7777 -> 127.0.0.1:5060 INVITE sip:98989898 at 127.0.0.3 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.2:7777;rport;branch=z9hG4bKH9jeDKvgKFaZe Max-Forwards: 70 From: "" ;tag=U10cU9N04jr2H To: Call-ID: 66fc7129-aa7f-1235-d7a5-623562383663 CSeq: 106564436 INVITE Contact: User-Agent: ASTPP Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, NOTIFY Supported: path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 224 Remote-Party-ID: ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1493777736 1493777737 IN IP4 127.0.0.2 s=FreeSWITCH c=IN IP4 127.0.0.2 t=0 0 m=audio 23264 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 2017/05/03 10:43:20.692476 127.0.0.1:5060 -> 127.0.0.2:7777 SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 127.0.0.2:7777;received=127.0.0.2;rport=7777;branch=z9hG4bKH9jeDKvgKFaZe From: "" ;tag=U10cU9N04jr2H To: Call-ID: 66fc7129-aa7f-1235-d7a5-623562383663 CSeq: 106564436 INVITE server: SBC02 Content-Length: 0 2017/05/03 10:43:21.864131 127.0.0.1:5060 -> 127.0.0.2:7777 SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 127.0.0.2:7777;received=127.0.0.2;rport=7777;branch=z9hG4bKH9jeDKvgKFaZe From: "" ;tag=U10cU9N04jr2H To: ;tag=gK00fb6796 Call-ID: 66fc7129-aa7f-1235-d7a5-623562383663 CSeq: 106564436 INVITE Record-Route: Contact: Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH Content-Length: 242 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 489775 683282 IN IP4 127.0.0.3 s=SIP Media Capabilities c=IN IP4 74.201.159.142 t=0 0 m=audio 18542 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:20 2017/05/03 10:43:22.221180 127.0.0.1:5060 -> 127.0.0.2:7777 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 127.0.0.2:7777;received=127.0.0.2;rport=7777;branch=z9hG4bKH9jeDKvgKFaZe From: "" ;tag=U10cU9N04jr2H To: ;tag=gK00fb6796 Call-ID: 66fc7129-aa7f-1235-d7a5-623562383663 CSeq: 106564436 INVITE Record-Route: Contact: Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH Content-Length: 242 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 489775 683282 IN IP4 127.0.0.3 s=SIP Media Capabilities c=IN IP4 74.201.159.142 t=0 0 m=audio 18542 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:20 2017/05/03 10:43:39.460967 127.0.0.1:5060 -> 127.0.0.2:7777 SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.2:7777;received=127.0.0.2;rport=7777;branch=z9hG4bKH9jeDKvgKFaZe From: "" ;tag=U10cU9N04jr2H To: ;tag=gK00fb6796 Call-ID: 66fc7129-aa7f-1235-d7a5-623562383663 CSeq: 106564436 INVITE Record-Route: Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Contact: Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH Supported: timer,replaces Session-Expires: 1800;refresher=uas Content-Length: 242 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 489775 683282 IN IP4 127.0.0.3 s=SIP Media Capabilities c=IN IP4 74.201.159.142 t=0 0 m=audio 18542 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:20 2017/05/03 10:43:39.461906 127.0.0.2:7777 -> 127.0.0.1:5060 ACK sip:98989898 at 127.0.0.3:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.2:7777;rport;branch=z9hG4bKjjc7eeDmgr0Ha Route: Max-Forwards: 70 From: "" ;tag=U10cU9N04jr2H To: ;tag=gK00fb6796 Call-ID: 66fc7129-aa7f-1235-d7a5-623562383663 CSeq: 106564436 ACK Contact: Content-Length: 0 2017/05/03 10:43:41.242553 127.0.0.1:5060 -> 127.0.0.2:7777 INVITE sip:gw+Orange at 127.0.0.2:7777;transport=udp;gw=Orange SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bKd71d.003aca31.0 Via: SIP/2.0/UDP 127.0.0.3:5060;branch=z9hG4bK00B482bf1060047fa3d From: ;tag=gK00fb6796 To: ;tag=U10cU9N04jr2H Call-ID: 66fc7129-aa7f-1235-d7a5-623562383663 CSeq: 45709 INVITE Max-Forwards: 30 Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Contact: Supported: timer,replaces Session-Expires: 1800;refresher=uac Min-SE: 90 Content-Length: 292 Content-Disposition: session; handling=required Content-Type: application/sdp P-hint: rr-enforced X-AUTH-IP: 127.0.0.3 P-Accountcode: P-effective_caller_id_name: P-effective_caller_id_number: v=0 o=Sonus_UAC 489775 683283 IN IP4 127.0.0.3 s=SIP Media Capabilities c=IN IP4 74.201.159.142 t=0 0 m=audio 18542 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:20 2017/05/03 10:43:41.242863 127.0.0.2:7777 -> 127.0.0.1:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bKd71d.003aca31.0 Via: SIP/2.0/UDP 127.0.0.3:5060;branch=z9hG4bK00B482bf1060047fa3d From: ;tag=gK00fb6796 To: ;tag=U10cU9N04jr2H Call-ID: 66fc7129-aa7f-1235-d7a5-623562383663 CSeq: 45709 INVITE User-Agent: ASTPP Content-Length: 0 2017/05/03 10:43:41.245834 127.0.0.2:7777 -> 127.0.0.1:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bKd71d.003aca31.0 Via: SIP/2.0/UDP 127.0.0.3:5060;branch=z9hG4bK00B482bf1060047fa3d From: ;tag=gK00fb6796 To: ;tag=U10cU9N04jr2H Call-ID: 66fc7129-aa7f-1235-d7a5-623562383663 CSeq: 45709 INVITE Contact: User-Agent: ASTPP Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, NOTIFY Supported: path, replaces Content-Type: application/sdp Content-Length: 235 v=0 o=FreeSWITCH 1493777736 1493777738 IN IP4 127.0.0.2 s=FreeSWITCH c=IN IP4 127.0.0.2 t=0 0 m=audio 23264 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv 2017/05/03 10:43:41.330553 127.0.0.1:5060 -> 127.0.0.2:7777 ACK sip:gw+Orange at 127.0.0.2:7777;transport=udp;gw=Orange SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bKd71d.003aca31.2 Via: SIP/2.0/UDP 127.0.0.3:5060;branch=z9hG4bK00B483689250047fa3d From: ;tag=gK00fb6796 To: ;tag=U10cU9N04jr2H Call-ID: 66fc7129-aa7f-1235-d7a5-623562383663 CSeq: 45709 ACK Max-Forwards: 30 Content-Length: 0 P-hint: rr-enforced X-AUTH-IP: 127.0.0.3 P-Accountcode: P-effective_caller_id_name: P-effective_caller_id_number: 2017/05/03 10:43:41.330924 127.0.0.1:5060 -> 127.0.0.2:7777 INVITE sip:gw+Orange at 127.0.0.2:7777;transport=udp;gw=Orange SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK671d.b93a7e91.0 Via: SIP/2.0/UDP 127.0.0.3:5060;branch=z9hG4bK00B483753df0047fa3d From: ;tag=gK00fb6796 To: ;tag=U10cU9N04jr2H Call-ID: 66fc7129-aa7f-1235-d7a5-623562383663 CSeq: 45710 INVITE Max-Forwards: 30 Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Contact: Supported: timer,replaces Session-Expires: 1800;refresher=uac Min-SE: 90 Content-Length: 242 Content-Disposition: session; handling=required Content-Type: application/sdp P-hint: rr-enforced X-AUTH-IP: 127.0.0.3 P-Accountcode: P-effective_caller_id_name: P-effective_caller_id_number: v=0 o=Sonus_UAC 489775 683284 IN IP4 127.0.0.3 s=SIP Media Capabilities c=IN IP4 74.201.159.142 t=0 0 m=audio 18542 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:20 2017/05/03 10:43:41.331071 127.0.0.2:7777 -> 127.0.0.1:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK671d.b93a7e91.0 Via: SIP/2.0/UDP 127.0.0.3:5060;branch=z9hG4bK00B483753df0047fa3d From: ;tag=gK00fb6796 To: ;tag=U10cU9N04jr2H Call-ID: 66fc7129-aa7f-1235-d7a5-623562383663 CSeq: 45710 INVITE User-Agent: ASTPP Content-Length: 0 2017/05/03 10:43:41.345784 127.0.0.2:7777 -> 127.0.0.1:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK671d.b93a7e91.0 Via: SIP/2.0/UDP 127.0.0.3:5060;branch=z9hG4bK00B483753df0047fa3d From: ;tag=gK00fb6796 To: ;tag=U10cU9N04jr2H Call-ID: 66fc7129-aa7f-1235-d7a5-623562383663 CSeq: 45710 INVITE Contact: User-Agent: ASTPP Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, NOTIFY Supported: path, replaces Content-Type: application/sdp Content-Length: 235 v=0 o=FreeSWITCH 1493777736 1493777739 IN IP4 127.0.0.2 s=FreeSWITCH c=IN IP4 127.0.0.2 t=0 0 m=audio 23264 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv 2017/05/03 10:43:41.429819 127.0.0.1:5060 -> 127.0.0.2:7777 ACK sip:gw+Orange at 127.0.0.2:7777;transport=udp;gw=Orange SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK671d.b93a7e91.2 Via: SIP/2.0/UDP 127.0.0.3:5060;branch=z9hG4bK00B483fb2700047fa3d From: ;tag=gK00fb6796 To: ;tag=U10cU9N04jr2H Call-ID: 66fc7129-aa7f-1235-d7a5-623562383663 CSeq: 45710 ACK Max-Forwards: 30 Content-Length: 0 P-hint: rr-enforced X-AUTH-IP: 127.0.0.3 P-Accountcode: P-effective_caller_id_name: P-effective_caller_id_number: 2017/05/03 10:43:41.745470 127.0.0.2:7777 -> 127.0.0.1:5060 INVITE sip:98989898 at 127.0.0.3:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.2:7777;rport;branch=z9hG4bKKU5Zg9XQD1p4N Route: Max-Forwards: 70 From: "" ;tag=U10cU9N04jr2H To: ;tag=gK00fb6796 Call-ID: 66fc7129-aa7f-1235-d7a5-623562383663 CSeq: 106564437 INVITE Contact: User-Agent: ASTPP Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, NOTIFY Supported: path, replaces Content-Type: application/sdp Content-Length: 320 Remote-Party-ID: ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1493777736 1493777740 IN IP4 127.0.0.2 s=FreeSWITCH c=IN IP4 127.0.0.2 t=0 0 m=image 23264 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxFillBitRemoval a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:2000 a=T38FaxMaxDatagram:400 a=T38FaxUdpEC:t38UDPRedundancy 2017/05/03 10:43:41.745989 127.0.0.1:5060 -> 127.0.0.2:7777 SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 127.0.0.2:7777;received=127.0.0.2;rport=7777;branch=z9hG4bKKU5Zg9XQD1p4N From: "" ;tag=U10cU9N04jr2H To: ;tag=gK00fb6796 Call-ID: 66fc7129-aa7f-1235-d7a5-623562383663 CSeq: 106564437 INVITE server: SBC02 Content-Length: 0 2017/05/03 10:43:42.062994 127.0.0.1:5060 -> 127.0.0.2:7777 SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.2:7777;received=127.0.0.2;rport=7777;branch=z9hG4bKKU5Zg9XQD1p4N From: "" ;tag=U10cU9N04jr2H To: ;tag=gK00fb6796 Call-ID: 66fc7129-aa7f-1235-d7a5-623562383663 CSeq: 106564437 INVITE Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Contact: Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH Supported: timer,replaces Session-Expires: 1800;refresher=uas Content-Length: 310 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 489775 683285 IN IP4 127.0.0.3 s=SIP Media Capabilities c=IN IP4 74.201.159.142 t=0 0 m=image 18542 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:262 a=T38FaxMaxDatagram:176 a=T38FaxUdpEC:t38UDPRedundancy a=sendrecv 2017/05/03 10:43:42.079909 127.0.0.2:7777 -> 127.0.0.1:5060 ACK sip:98989898 at 127.0.0.3:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.2:7777;rport;branch=z9hG4bKm4yrj4eUaaDQH Route: Max-Forwards: 70 From: "" ;tag=U10cU9N04jr2H To: ;tag=gK00fb6796 Call-ID: 66fc7129-aa7f-1235-d7a5-623562383663 CSeq: 106564437 ACK Contact: Content-Length: 0 2017/05/03 10:45:10.563547 127.0.0.1:5060 -> 127.0.0.2:7777 INVITE sip:gw+Orange at 127.0.0.2:7777;transport=udp;gw=Orange SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK771d.5b43c791.0 Via: SIP/2.0/UDP 127.0.0.3:5060;branch=z9hG4bK00B6b46e6dd0047fa3d From: ;tag=gK00fb6796 To: ;tag=U10cU9N04jr2H Call-ID: 66fc7129-aa7f-1235-d7a5-623562383663 CSeq: 45711 INVITE Max-Forwards: 30 Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Contact: Supported: timer,replaces Session-Expires: 1800;refresher=uac Min-SE: 90 Content-Length: 237 Content-Disposition: session; handling=required Content-Type: application/sdp P-hint: rr-enforced X-AUTH-IP: 127.0.0.3 P-Accountcode: P-effective_caller_id_name: P-effective_caller_id_number: v=0 o=Sonus_UAC 489775 683286 IN IP4 127.0.0.3 s=SIP Media Capabilities c=IN IP4 74.201.159.142 t=0 0 m=audio 18542 RTP/AVP 8 0 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=sendrecv a=silenceSupp:off - - - - a=maxptime:20 2017/05/03 10:45:10.563788 127.0.0.2:7777 -> 127.0.0.1:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK771d.5b43c791.0 Via: SIP/2.0/UDP 127.0.0.3:5060;branch=z9hG4bK00B6b46e6dd0047fa3d From: ;tag=gK00fb6796 To: ;tag=U10cU9N04jr2H Call-ID: 66fc7129-aa7f-1235-d7a5-623562383663 CSeq: 45711 INVITE User-Agent: ASTPP Content-Length: 0 2017/05/03 10:45:10.564095 127.0.0.2:7777 -> 127.0.0.1:5060 SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK771d.5b43c791.0 Via: SIP/2.0/UDP 127.0.0.3:5060;branch=z9hG4bK00B6b46e6dd0047fa3d From: ;tag=gK00fb6796 To: ;tag=U10cU9N04jr2H Call-ID: 66fc7129-aa7f-1235-d7a5-623562383663 CSeq: 45711 INVITE User-Agent: ASTPP Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, NOTIFY Supported: path, replaces Content-Length: 0 2017/05/03 10:45:10.565095 127.0.0.1:5060 -> 127.0.0.2:7777 ACK sip:gw+Orange at 127.0.0.2:7777;transport=udp;gw=Orange SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK771d.5b43c791.0 From: ;tag=gK00fb6796 Call-ID: 66fc7129-aa7f-1235-d7a5-623562383663 To: ;tag=U10cU9N04jr2H CSeq: 45711 ACK Max-Forwards: 70 User-Agent: SIP Proxy Content-Length: 0 2017/05/03 10:45:10.566229 127.0.0.2:7777 -> 127.0.0.1:5060 BYE sip:98989898 at 127.0.0.3:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.2:7777;rport;branch=z9hG4bKNDrHmZZy7j39c Route: Max-Forwards: 70 From: "" ;tag=U10cU9N04jr2H To: ;tag=gK00fb6796 Call-ID: 66fc7129-aa7f-1235-d7a5-623562383663 CSeq: 106564438 BYE User-Agent: ASTPP Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, NOTIFY Supported: path, replaces Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" Content-Length: 0 2017/05/03 10:45:10.651303 127.0.0.1:5060 -> 127.0.0.2:7777 SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.2:7777;received=127.0.0.2;rport=7777;branch=z9hG4bKNDrHmZZy7j39c From: "" ;tag=U10cU9N04jr2H To: ;tag=gK00fb6796 Call-ID: 66fc7129-aa7f-1235-d7a5-623562383663 CSeq: 106564438 BYE Content-Length: 0 From mylists at polite.se Wed May 3 14:41:37 2017 From: mylists at polite.se (Oivvio Polite) Date: Wed, 3 May 2017 12:41:37 +0200 Subject: [Freeswitch-users] Options for initiating call from browser (not WebRTC) Message-ID: <20170503104137.GA9671@blomma.liberationtech.net> I have a simple call recording solution set up with a Twilio-like PaaS. The user interacts with a web app where she enters a destination number. My web app then instructs the PaaS to 1. Call the users phone 2. Wait for the user to answer 3. Start a recording. 4. Connect the destination number. The resulting wav is available for the user to download in the web app. I now want to replace the PaaS with FreeSwitch. I've done a lot of experimenting with FS during the last year, and managed to get callrecording of PSTN and WebRTC calls to work. With those calls origination was from PSTN or WebRTC. Now I want to do origination with a HTTP request. When leafing through the FreeSwitch book from PACKT I see that I have multiple options for doing this. 1. fs_cli Have the browser talk to a separate web app (written in node or python or whatever) that the initiates the call with fs_cli. 2. mod_httapi Have the browser talk to a webapp that talks to FS via mod_httapi. In this scenario I can't find any info on how to originate calls in mod_httapi though. 3. mod_event_socket Have a the browser talk to a webapp that talks to FS via mod_event_socket. In all of these scenarios I figure that I'll have the webapp running on the same box as FS and that I'll do all authentication/authorization of the user in the webapp. But which option should I go with? regards Oivvio From tayeb.meftah at gmail.com Wed May 3 15:36:26 2017 From: tayeb.meftah at gmail.com (tayeb.meftah at gmail.com) Date: Wed, 3 May 2017 12:36:26 +0100 Subject: [Freeswitch-users] Options for initiating call from browser (not WebRTC) In-Reply-To: <20170503104137.GA9671@blomma.liberationtech.net> References: <20170503104137.GA9671@blomma.liberationtech.net> Message-ID: <32197E08-0F58-4F57-B8F1-8B6D1E94A2C2@gmail.com> mod xml rpc or php esl Envoy? de mon iPhone > Le 3 mai 2017 ? 11:41, Oivvio Polite a ?crit : > > I have a simple call recording solution set up with a Twilio-like PaaS. > > The user interacts with a web app where she enters a destination number. > My web app then instructs the PaaS to > > 1. Call the users phone > 2. Wait for the user to answer > 3. Start a recording. > 4. Connect the destination number. > > The resulting wav is available for the user to download in the web app. > > I now want to replace the PaaS with FreeSwitch. I've done a lot of > experimenting with FS during the last year, and managed to get > callrecording of PSTN and WebRTC calls to work. > > With those calls origination was from PSTN or WebRTC. Now I want to do > origination with a HTTP request. When leafing through the FreeSwitch > book from PACKT I see that I have multiple options for doing this. > > 1. fs_cli > > Have the browser talk to a separate web app (written in node or python > or whatever) that the initiates the call with fs_cli. > > > 2. mod_httapi > > Have the browser talk to a webapp that talks to FS via mod_httapi. In > this scenario I can't find any info on how to originate calls in > mod_httapi though. > > > 3. mod_event_socket > > Have a the browser talk to a webapp that talks to FS via > mod_event_socket. > > > In all of these scenarios I figure that I'll have the webapp running on > the same box as FS and that I'll do all authentication/authorization of > the user in the webapp. > > But which option should I go with? > > > regards Oivvio > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170503/12d22c35/attachment.html From xxxman2008 at 126.com Wed May 3 16:02:30 2017 From: xxxman2008 at 126.com (Raymond) Date: Wed, 3 May 2017 20:02:30 +0800 (CST) Subject: [Freeswitch-users] RTP handling from different freeswitch In-Reply-To: References: Message-ID: <6e766e3d.b752.15bce30b51e.Coremail.xxxman2008@126.com> Hi,all I really want to tell you an Chinese story ,but my english is poor. Anyway , I just want to say , we can resolve "difficult" things in a simple way. devang , the key of your question is , how to define your "USA customer" and "Norway customer"? IP address? or Call-in Number? if you use "ip address" , Many DNS Service provider will glad to help you redirect traffic to "optimized destination". I.E ,your USA customer get DNS resolve result to usa server ip ,and your Norway customer get Norway server ip. it's called ?intelligent dns service?. That's why we use "Domain Name" in internet. And the reason of 'DESTINATION_OUT_OF_ORDER' is so clear. Freeswtich already told you ,"[Bind Error! 65.11.17.9:25146]" . Maybe , 65.11.17.9 is not your server ip ,maybe the port 25146 has been used. and maybe ...... ,plz check it out yourself . Raymond ? 2017-05-02 21:06:53?"devang nathwani" ??? Hello, I have my setup at Norway and some of my customers are in USA so i have concern about audio latency. I want the configuration such a way that if USA customer is calling than the rtp should route through my USA freeswitch servers however sip should route through Norway setup. Is there any way i can achieve this? I have tried modifying rtp-ip but i am getting 'DESTINATION_OUT_OF_ORDER' 2017-05-02 08:50:42.650711 [ERR] switch_core_media.c:5843 AUDIO RTP REPORTS ERROR: [Bind Error! 65.11.17.9:25146] 2017-05-02 08:50:42.650711 [NOTICE] switch_core_media.c:5844 Hangup sofia/default/12345679 [CS_CONSUME_MEDIA] [DESTINATION_OUT_OF_ORDER] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170503/3479d9eb/attachment.html From mathias at celea-consulting.fr Wed May 3 16:11:44 2017 From: mathias at celea-consulting.fr (CELEA : Mathias) Date: Wed, 3 May 2017 14:11:44 +0200 Subject: [Freeswitch-users] RTP handling from different freeswitch In-Reply-To: <6e766e3d.b752.15bce30b51e.Coremail.xxxman2008@126.com> References: <6e766e3d.b752.15bce30b51e.Coremail.xxxman2008@126.com> Message-ID: Hi, You can use a kamailio in front of your FS servers with geoip module, and load balance sip INVITE depending the ip address location. Regards Mathias Le 03/05/2017 ? 14:02, Raymond a ?crit : > Hi,all > > I really want to tell you an Chinese story ,but my english is > poor. Anyway , I just want to say , we can resolve "difficult" > things in a simple way. > > devang , the key of your question is , how to define your "USA > customer" and "Norway customer"? IP address? or Call-in Number? > > if you use "ip address" , Many DNS Service provider will glad > to help you redirect traffic to "optimized destination". I.E ,your > USA customer get DNS resolve result to usa server ip ,and your Norway > customer get Norway server ip. it's called ?intelligent dns > service?. That's why we use "Domain Name" in internet. > > And the reason of 'DESTINATION_OUT_OF_ORDER' is so clear. > Freeswtich already told you ,"[Bind Error! 65.11.17.9:25146 > ]" . Maybe , 65.11.17.9 > is not your server ip ,maybe the port 25146 > has been used. and maybe ...... ,plz check it out yourself . > > Raymond > > > > ? 2017-05-02 21:06:53?"devang nathwani" > ??? > > Hello, > > I have my setup at Norway and some of my customers are in USA so i > have concern about audio latency. > > I want the configuration such a way that if USA customer is > calling than the rtp should route through my USA freeswitch > servers however sip should route through Norway setup. > > Is there any way i can achieve this? > > I have tried modifying rtp-ip but i am getting > 'DESTINATION_OUT_OF_ORDER' > > 2017-05-02 08:50:42.650711 [ERR] switch_core_media.c:5843 AUDIO > RTP REPORTS ERROR: [Bind Error! 65.11.17.9:25146 > ] > 2017-05-02 08:50:42.650711 [NOTICE] switch_core_media.c:5844 > Hangup sofia/default/12345679 [CS_CONSUME_MEDIA] > [DESTINATION_OUT_OF_ORDER] > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- ...................................................................... Mathias WOLFF Expert T?l?com @ CELEA CONSULTING Tel : +33 9.72.13.22.62 Gsm : +33 6.79.59.43.32 Fax : +33 2.40.63.41.33 Mattermost : https://framateam.org/celea/channels/pyfreebilling site web : www.celea-consulting.fr Blog : www.blog-des-telecoms.com PyFreeBilling : www.pyfreebilling.com ...................................................................... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170503/03558a7a/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 488 bytes Desc: OpenPGP digital signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170503/03558a7a/attachment-0001.bin From dragos.oancea at vonage.com Wed May 3 16:21:53 2017 From: dragos.oancea at vonage.com (Oancea, Dragos) Date: Wed, 3 May 2017 13:21:53 +0100 Subject: [Freeswitch-users] OPUS Codec In-Reply-To: <20170503071638.GA14349@mail.marktcontact.com> References: <20170503071638.GA14349@mail.marktcontact.com> Message-ID: I've reached 2000 simultaneous calls on a Dell Blade showing 40 CPU cores in /proc/cpuinfo , transcoding opus at 8000h <-> PCMA with FS. Two years ago. If you want to encode FEC you'll need to use opus at 8000h as codec_string or tweak the config settings in the xml to set the encoder in Narrowband audio . opus at 16000h (newly added) should work with encoding FEC too. When you prepare your setup use CLI command opus_debug and netem to introduce packet loss, this way you'll be sure if the phones are sending FEC or not. Some may try to vary the bitrate to a value where the opus encoder simply does not encode FEC. Freeswitch, just like WebRTC stack sets by default the encoder in Fullband , which is CELT mode, and there's no FEC there. It's all in the doc Giacomo shown. Cheers, Dragos On Wed, May 3, 2017 at 8:16 AM, EL wrote: > > Yealink is supporting OPUS on several other models since firmware > V81: > > Quote: > "We will support opus on the standard V81 of > SIP-T40P/T23P/T23G/T2 1(P) E2/T19(P) E2." > > Source: http://forum.yealink.com/forum/showthread.php?tid= > 29650&pid=51262&mode=threaded > > I can confirm OPUS implementation on the 'T21P E2' model. > > -- > EL > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170503/b35985e2/attachment.html From cjbujold at accra.ca Wed May 3 18:54:26 2017 From: cjbujold at accra.ca (Charles Bujold) Date: Wed, 3 May 2017 11:54:26 -0300 Subject: [Freeswitch-users] Outbound calls being dropped after update to Version 1.6.17 -34-0fc0946 64bit Message-ID: <006d01d2c41d$294be0e0$7be3a2a0$@accra.ca> Still no solutions, Here is a log of a call. Any suggestion would be appreciated 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:57.156987 [DEBUG] sofia.c:7048 Channel sofia/internal/15068559226 entering state [calling][0] 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:57.156987 [DEBUG] switch_core_state_machine.c:643 (sofia/internal/15068559226) State ROUTING 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:57.156987 [DEBUG] mod_sofia.c:143 sofia/internal/15068559226 SOFIA ROUTING 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:57.156987 [DEBUG] switch_ivr_originate.c:67 (sofia/internal/15068559226) State Change CS_ROUTING -> CS_CONSUME_MEDIA 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:57.156987 [DEBUG] switch_core_state_machine.c:643 (sofia/internal/15068559226) State ROUTING going to sleep 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:57.156987 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/15068559226) Running State Change CS_CONSUME_MEDIA (Cur 2 Tot 136) 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:57.156987 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/15068559226) State CONSUME_MEDIA 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:57.156987 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/15068559226) State CONSUME_MEDIA going to sleep 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:57.236974 [DEBUG] sofia.c:7048 Channel sofia/internal/15068559226 entering state [calling][0] 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.737079 [DEBUG] sofia.c:7048 Channel sofia/internal/15068559226 entering state [proceeding][183] 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.737079 [DEBUG] sofia.c:7058 Remote SDP: 3960b42a-6c36-49c9-96a0-a844b3f0f259 v=0 3960b42a-6c36-49c9-96a0-a844b3f0f259 o=FreeSWITCH 1493799827 1493799828 IN IP4 72.55.158.152 3960b42a-6c36-49c9-96a0-a844b3f0f259 s=FreeSWITCH 3960b42a-6c36-49c9-96a0-a844b3f0f259 c=IN IP4 72.55.158.152 3960b42a-6c36-49c9-96a0-a844b3f0f259 t=0 0 3960b42a-6c36-49c9-96a0-a844b3f0f259 m=audio 22274 RTP/AVP 0 101 13 3960b42a-6c36-49c9-96a0-a844b3f0f259 a=rtpmap:0 PCMU/8000 3960b42a-6c36-49c9-96a0-a844b3f0f259 a=rtpmap:101 telephone-event/8000 3960b42a-6c36-49c9-96a0-a844b3f0f259 a=fmtp:101 0-16 3960b42a-6c36-49c9-96a0-a844b3f0f259 a=rtpmap:13 CN/8000 3960b42a-6c36-49c9-96a0-a844b3f0f259 a=ptime:20 3960b42a-6c36-49c9-96a0-a844b3f0f259 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.737079 [DEBUG] switch_core_media.c:4436 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.737079 [DEBUG] switch_core_media.c:4491 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.737079 [DEBUG] switch_core_media.c:4436 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[G722:9:8000:20:64000:1] 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.737079 [DEBUG] switch_core_media.c:4352 Set telephone-event payload to 101 at 8000 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.737079 [DEBUG] switch_core_media.c:3056 Set Codec sofia/internal/15068559226 PCMU/8000 20 ms 160 samples 64000 bits 1 channels 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.737079 [DEBUG] switch_core_codec.c:111 sofia/internal/15068559226 Original read codec set to PCMU:0 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.737079 [DEBUG] switch_core_media.c:4695 Set telephone-event payload to 101 at 8000 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.737079 [DEBUG] switch_core_media.c:4754 sofia/internal/15068559226 Set 2833 dtmf send payload to 101 recv payload to 101 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.737079 [DEBUG] switch_core_media.c:6865 AUDIO RTP [sofia/internal/15068559226] 192.168.20.161 port 29764 -> 72.55.158.152 port 22274 codec: 0 ms: 20 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.737079 [DEBUG] switch_rtp.c:4096 Starting timer [soft] 160 bytes per 20ms 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.737079 [DEBUG] switch_core_media.c:7166 sofia/internal/15068559226 Set 2833 dtmf send payload to 101 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.737079 [DEBUG] switch_core_media.c:7173 sofia/internal/15068559226 Set 2833 dtmf receive payload to 101 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.737079 [DEBUG] switch_core_media.c:7196 sofia/internal/15068559226 Set rtp dtmf delay to 40 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.737079 [DEBUG] switch_core_media.c:7202 Set comfort noise payload to 13 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.737079 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/15068559226! 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.737079 [DEBUG] switch_channel.c:3473 (sofia/internal/15068559226) Callstate Change DOWN -> EARLY cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:34:58.757038 [DEBUG] switch_ivr_originate.c:410 Setting codec string on sofia/internal/200 at 192.168.20.161 to PCMU at 8000h@20i cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:34:58.757038 [INFO] switch_ivr_originate.c:3639 Sending early media cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:34:58.757038 [DEBUG] switch_core_media.c:6865 AUDIO RTP [sofia/internal/200 at 192.168.20.161] 192.168.20.161 port 16564 -> 192.168.20.150 port 19542 codec: 0 ms: 20 cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:34:58.757038 [DEBUG] switch_rtp.c:4096 Starting timer [soft] 160 bytes per 20ms cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:34:58.757038 [DEBUG] switch_core_media.c:7166 sofia/internal/200 at 192.168.20.161 Set 2833 dtmf send payload to 101 cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:34:58.757038 [DEBUG] switch_core_media.c:7173 sofia/internal/200 at 192.168.20.161 Set 2833 dtmf receive payload to 101 cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:34:58.757038 [DEBUG] switch_core_media.c:7196 sofia/internal/200 at 192.168.20.161 Set rtp dtmf delay to 40 cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:34:58.757038 [DEBUG] mod_sofia.c:2363 Ring SDP: cc1d0c18-e576-47a6-bf4f-35ab835f920c v=0 cc1d0c18-e576-47a6-bf4f-35ab835f920c o=FreeSWITCH 1493805534 1493805535 IN IP4 192.168.20.161 cc1d0c18-e576-47a6-bf4f-35ab835f920c s=FreeSWITCH cc1d0c18-e576-47a6-bf4f-35ab835f920c c=IN IP4 192.168.20.161 cc1d0c18-e576-47a6-bf4f-35ab835f920c t=0 0 cc1d0c18-e576-47a6-bf4f-35ab835f920c m=audio 16564 RTP/AVP 0 101 cc1d0c18-e576-47a6-bf4f-35ab835f920c a=rtpmap:0 PCMU/8000 cc1d0c18-e576-47a6-bf4f-35ab835f920c a=rtpmap:101 telephone-event/8000 cc1d0c18-e576-47a6-bf4f-35ab835f920c a=fmtp:101 0-16 cc1d0c18-e576-47a6-bf4f-35ab835f920c a=ptime:20 cc1d0c18-e576-47a6-bf4f-35ab835f920c a=sendrecv cc1d0c18-e576-47a6-bf4f-35ab835f920c cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:34:58.757038 [NOTICE] mod_sofia.c:2366 Pre-Answer sofia/internal/200 at 192.168.20.161! cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:34:58.757038 [DEBUG] switch_channel.c:3473 (sofia/internal/200 at 192.168.20.161) Callstate Change RINGING -> EARLY cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:34:58.757038 [DEBUG] switch_ivr_originate.c:3690 Originate Resulted in Success: [sofia/internal/15068559226] cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:34:58.757038 [DEBUG] sofia.c:7048 Channel sofia/internal/200 at 192.168.20.161 entering state [early][183] 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.757038 [DEBUG] switch_ivr_bridge.c:1601 (sofia/internal/15068559226) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.757038 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/15068559226) Running State Change CS_EXCHANGE_MEDIA (Cur 2 Tot 136) 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.757038 [DEBUG] switch_core_state_machine.c:653 (sofia/internal/15068559226) State EXCHANGE_MEDIA 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.757038 [DEBUG] mod_sofia.c:631 SOFIA EXCHANGE_MEDIA 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.796985 [DEBUG] switch_rtp.c:7229 Correct audio ip/port confirmed. cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:34:58.877072 [DEBUG] switch_rtp.c:7229 Correct audio ip/port confirmed. 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:35:00.317073 [DEBUG] sofia.c:7048 Channel sofia/internal/15068559226 entering state [completing][200] 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:35:00.317073 [DEBUG] sofia.c:7055 Duplicate SDP 3960b42a-6c36-49c9-96a0-a844b3f0f259 v=0 3960b42a-6c36-49c9-96a0-a844b3f0f259 o=FreeSWITCH 1493799827 1493799828 IN IP4 72.55.158.152 3960b42a-6c36-49c9-96a0-a844b3f0f259 s=FreeSWITCH 3960b42a-6c36-49c9-96a0-a844b3f0f259 c=IN IP4 72.55.158.152 3960b42a-6c36-49c9-96a0-a844b3f0f259 t=0 0 3960b42a-6c36-49c9-96a0-a844b3f0f259 m=audio 22274 RTP/AVP 0 101 13 3960b42a-6c36-49c9-96a0-a844b3f0f259 a=rtpmap:0 PCMU/8000 3960b42a-6c36-49c9-96a0-a844b3f0f259 a=rtpmap:101 telephone-event/8000 3960b42a-6c36-49c9-96a0-a844b3f0f259 a=fmtp:101 0-16 3960b42a-6c36-49c9-96a0-a844b3f0f259 a=rtpmap:13 CN/8000 3960b42a-6c36-49c9-96a0-a844b3f0f259 a=ptime:20 3960b42a-6c36-49c9-96a0-a844b3f0f259 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:35:00.337073 [DEBUG] sofia.c:7048 Channel sofia/internal/15068559226 entering state [ready][200] 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:35:00.337073 [NOTICE] sofia.c:8123 Channel [sofia/internal/15068559226] has been answered 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:35:00.337073 [DEBUG] switch_channel.c:3772 (sofia/internal/15068559226) Callstate Change EARLY -> ACTIVE cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:35:00.357041 [DEBUG] switch_core_media.c:6848 Audio params are unchanged for sofia/internal/200 at 192.168.20.161. cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:35:00.357041 [DEBUG] mod_sofia.c:850 Local SDP sofia/internal/200 at 192.168.20.161: cc1d0c18-e576-47a6-bf4f-35ab835f920c v=0 cc1d0c18-e576-47a6-bf4f-35ab835f920c o=FreeSWITCH 1493805534 1493805536 IN IP4 192.168.20.161 cc1d0c18-e576-47a6-bf4f-35ab835f920c s=FreeSWITCH cc1d0c18-e576-47a6-bf4f-35ab835f920c c=IN IP4 192.168.20.161 cc1d0c18-e576-47a6-bf4f-35ab835f920c t=0 0 cc1d0c18-e576-47a6-bf4f-35ab835f920c m=audio 16564 RTP/AVP 0 101 cc1d0c18-e576-47a6-bf4f-35ab835f920c a=rtpmap:0 PCMU/8000 cc1d0c18-e576-47a6-bf4f-35ab835f920c a=rtpmap:101 telephone-event/8000 cc1d0c18-e576-47a6-bf4f-35ab835f920c a=fmtp:101 0-16 cc1d0c18-e576-47a6-bf4f-35ab835f920c a=ptime:20 cc1d0c18-e576-47a6-bf4f-35ab835f920c a=sendrecv cc1d0c18-e576-47a6-bf4f-35ab835f920c cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:35:00.357041 [NOTICE] switch_ivr_bridge.c:623 Channel [sofia/internal/200 at 192.168.20.161] has been answered cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:35:00.357041 [DEBUG] switch_channel.c:3772 (sofia/internal/200 at 192.168.20.161) Callstate Change EARLY -> ACTIVE cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:35:00.357041 [DEBUG] sofia.c:7048 Channel sofia/internal/200 at 192.168.20.161 entering state [completed][200] cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:35:00.377070 [DEBUG] sofia.c:7048 Channel sofia/internal/200 at 192.168.20.161 entering state [ready][200] cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:35:00.396999 [DEBUG] switch_rtp.c:7229 Correct audio ip/port confirmed. 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:35:00.396999 [DEBUG] switch_rtp.c:7229 Correct audio ip/port confirmed. 2017-05-03 11:35:00.396999 [INFO] sofia.c:1279 sofia/internal/15068559226 Update Callee ID to "Outbound Call" <+15068559226> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:36:02.717070 [DEBUG] sofia.c:7048 Channel sofia/internal/15068559226 entering state [calling][0] 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:36:03.757073 [DEBUG] sofia.c:7048 Channel sofia/internal/15068559226 entering state [completing][200] 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:36:03.757073 [DEBUG] sofia.c:7055 Duplicate SDP 3960b42a-6c36-49c9-96a0-a844b3f0f259 v=0 3960b42a-6c36-49c9-96a0-a844b3f0f259 o=FreeSWITCH 1493799827 1493799828 IN IP4 72.55.158.152 3960b42a-6c36-49c9-96a0-a844b3f0f259 s=FreeSWITCH 3960b42a-6c36-49c9-96a0-a844b3f0f259 c=IN IP4 72.55.158.152 3960b42a-6c36-49c9-96a0-a844b3f0f259 t=0 0 3960b42a-6c36-49c9-96a0-a844b3f0f259 m=audio 22274 RTP/AVP 0 101 13 3960b42a-6c36-49c9-96a0-a844b3f0f259 a=rtpmap:0 PCMU/8000 3960b42a-6c36-49c9-96a0-a844b3f0f259 a=rtpmap:101 telephone-event/8000 3960b42a-6c36-49c9-96a0-a844b3f0f259 a=fmtp:101 0-16 3960b42a-6c36-49c9-96a0-a844b3f0f259 a=rtpmap:13 CN/8000 3960b42a-6c36-49c9-96a0-a844b3f0f259 a=ptime:20 3960b42a-6c36-49c9-96a0-a844b3f0f259 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:36:03.777033 [DEBUG] sofia.c:7048 Channel sofia/internal/15068559226 entering state [ready][200] 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:03.737072 [DEBUG] sofia.c:7048 Channel sofia/internal/15068559226 entering state [calling][0] 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.737038 [DEBUG] sofia.c:7048 Channel sofia/internal/15068559226 entering state [terminating][408] 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.737038 [NOTICE] sofia.c:8237 Hangup sofia/internal/15068559226 [CS_EXCHANGE_MEDIA] [RECOVERY_ON_TIMER_EXPIRE] 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.737038 [DEBUG] switch_ivr_bridge.c:712 sofia/internal/15068559226 ending bridge by request from read function 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.737038 [DEBUG] switch_ivr_bridge.c:787 BRIDGE THREAD DONE [sofia/internal/15068559226] 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.737038 [DEBUG] switch_core_state_machine.c:653 (sofia/internal/15068559226) State EXCHANGE_MEDIA going to sleep 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.737038 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/15068559226) Running State Change CS_HANGUP (Cur 2 Tot 136) 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.737038 [DEBUG] switch_core_state_machine.c:850 (sofia/internal/15068559226) Callstate Change ACTIVE -> HANGUP 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.737038 [DEBUG] switch_core_state_machine.c:852 (sofia/internal/15068559226) State HANGUP 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.737038 [DEBUG] mod_sofia.c:438 Channel sofia/internal/15068559226 hanging up, cause: RECOVERY_ON_TIMER_EXPIRE 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.737038 [DEBUG] switch_core_state_machine.c:60 sofia/internal/15068559226 Standard HANGUP, cause: RECOVERY_ON_TIMER_EXPIRE 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.737038 [DEBUG] switch_core_state_machine.c:852 (sofia/internal/15068559226) State HANGUP going to sleep 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.737038 [DEBUG] switch_core_state_machine.c:619 (sofia/internal/15068559226) State Change CS_HANGUP -> CS_REPORTING 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.737038 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/15068559226) Running State Change CS_REPORTING (Cur 2 Tot 136) 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.737038 [DEBUG] switch_core_state_machine.c:938 (sofia/internal/15068559226) State REPORTING 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.737038 [DEBUG] switch_core_state_machine.c:174 sofia/internal/15068559226 Standard REPORTING, cause: RECOVERY_ON_TIMER_EXPIRE 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.737038 [DEBUG] switch_core_state_machine.c:938 (sofia/internal/15068559226) State REPORTING going to sleep 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.737038 [DEBUG] switch_core_state_machine.c:610 (sofia/internal/15068559226) State Change CS_REPORTING -> CS_DESTROY 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.737038 [DEBUG] switch_core_session.c:1664 Session 136 (sofia/internal/15068559226) Locked, Waiting on external entities cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.757070 [DEBUG] switch_ivr_bridge.c:706 sofia/internal/15068559226 ending bridge by request from write function cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.757070 [DEBUG] switch_ivr_bridge.c:787 BRIDGE THREAD DONE [sofia/internal/200 at 192.168.20.161] cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.757070 [NOTICE] switch_ivr_bridge.c:1751 Hangup sofia/internal/200 at 192.168.20.161 [CS_EXECUTE] [RECOVERY_ON_TIMER_EXPIRE] 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.757070 [NOTICE] switch_core_session.c:1682 Session 136 (sofia/internal/15068559226) Ended 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.757070 [NOTICE] switch_core_session.c:1686 Close Channel sofia/internal/15068559226 [CS_DESTROY] 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.757070 [DEBUG] switch_core_state_machine.c:741 (sofia/internal/15068559226) Running State Change CS_DESTROY (Cur 1 Tot 136) cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.757070 [DEBUG] switch_core_session.c:2814 sofia/internal/200 at 192.168.20.161 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.757070 [DEBUG] switch_core_state_machine.c:650 (sofia/internal/200 at 192.168.20.161) State EXECUTE going to sleep cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.757070 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/200 at 192.168.20.161) Running State Change CS_HANGUP (Cur 1 Tot 136) 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.757070 [DEBUG] switch_core_state_machine.c:751 (sofia/internal/15068559226) State DESTROY 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.757070 [DEBUG] mod_sofia.c:343 sofia/internal/15068559226 SOFIA DESTROY cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.757070 [DEBUG] switch_core_state_machine.c:850 (sofia/internal/200 at 192.168.20.161) Callstate Change ACTIVE -> HANGUP cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.757070 [DEBUG] switch_core_state_machine.c:852 (sofia/internal/200 at 192.168.20.161) State HANGUP cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.757070 [DEBUG] mod_sofia.c:432 sofia/internal/200 at 192.168.20.161 Overriding SIP cause 504 with 408 from the other leg cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.757070 [DEBUG] mod_sofia.c:438 Channel sofia/internal/200 at 192.168.20.161 hanging up, cause: RECOVERY_ON_TIMER_EXPIRE cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.757070 [DEBUG] mod_sofia.c:491 Sending BYE to sofia/internal/200 at 192.168.20.161 cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.757070 [DEBUG] switch_core_state_machine.c:60 sofia/internal/200 at 192.168.20.161 Standard HANGUP, cause: RECOVERY_ON_TIMER_EXPIRE cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.757070 [DEBUG] switch_core_state_machine.c:852 (sofia/internal/200 at 192.168.20.161) State HANGUP going to sleep 2017-05-03 11:37:35.757070 [DEBUG] switch_nat.c:542 unmapped public port 29764 protocol UDP to localport 29764 cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.757070 [DEBUG] switch_core_state_machine.c:619 (sofia/internal/200 at 192.168.20.161) State Change CS_HANGUP -> CS_REPORTING cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.757070 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/200 at 192.168.20.161) Running State Change CS_REPORTING (Cur 1 Tot 136) cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.757070 [DEBUG] switch_core_state_machine.c:938 (sofia/internal/200 at 192.168.20.161) State REPORTING 2017-05-03 11:37:35.757070 [DEBUG] switch_nat.c:542 unmapped public port 29765 protocol UDP to localport 29765 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.757070 [DEBUG] switch_core_state_machine.c:181 sofia/internal/15068559226 Standard DESTROY 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.757070 [DEBUG] switch_core_state_machine.c:751 (sofia/internal/15068559226) State DESTROY going to sleep cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.776979 [DEBUG] switch_core_state_machine.c:174 sofia/internal/200 at 192.168.20.161 Standard REPORTING, cause: RECOVERY_ON_TIMER_EXPIRE cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.776979 [DEBUG] switch_core_state_machine.c:938 (sofia/internal/200 at 192.168.20.161) State REPORTING going to sleep cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.776979 [DEBUG] switch_core_state_machine.c:610 (sofia/internal/200 at 192.168.20.161) State Change CS_REPORTING -> CS_DESTROY cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.776979 [DEBUG] switch_core_session.c:1664 Session 135 (sofia/internal/200 at 192.168.20.161) Locked, Waiting on external entities cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.776979 [NOTICE] switch_core_session.c:1682 Session 135 (sofia/internal/200 at 192.168.20.161) Ended cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.776979 [NOTICE] switch_core_session.c:1686 Close Channel sofia/internal/200 at 192.168.20.161 [CS_DESTROY] cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.776979 [DEBUG] switch_core_state_machine.c:741 (sofia/internal/200 at 192.168.20.161) Running State Change CS_DESTROY (Cur 0 Tot 136) cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.796986 [DEBUG] switch_core_state_machine.c:751 (sofia/internal/200 at 192.168.20.161) State DESTROY cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.796986 [DEBUG] mod_sofia.c:343 sofia/internal/200 at 192.168.20.161 SOFIA DESTROY cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.796986 [DEBUG] switch_core_state_machine.c:181 sofia/internal/200 at 192.168.20.161 Standard DESTROY cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.796986 [DEBUG] switch_core_state_machine.c:751 (sofia/internal/200 at 192.168.20.161) State DESTROY going to sleep -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170503/768803ae/attachment-0001.html From brian at freeswitch.org Wed May 3 19:05:00 2017 From: brian at freeswitch.org (Brian West) Date: Wed, 3 May 2017 10:05:00 -0500 Subject: [Freeswitch-users] Outbound calls being dropped after update to Version 1.6.17 -34-0fc0946 64bit In-Reply-To: <006d01d2c41d$294be0e0$7be3a2a0$@accra.ca> References: <006d01d2c41d$294be0e0$7be3a2a0$@accra.ca> Message-ID: This is a NAT issue, 'sofia global siptrace on' and watch the signaling. /b 2017-05-03 9:54 GMT-05:00 Charles Bujold : > Still no solutions, Here is a log > of a call. Any suggestion would be appreciated > > > > > > > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:57.156987 [DEBUG] > sofia.c:7048 Channel sofia/internal/15068559226 entering state [calling][0] > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:57.156987 [DEBUG] > switch_core_state_machine.c:643 (sofia/internal/15068559226) State ROUTING > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:57.156987 [DEBUG] > mod_sofia.c:143 sofia/internal/15068559226 SOFIA ROUTING > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:57.156987 [DEBUG] > switch_ivr_originate.c:67 (sofia/internal/15068559226) State Change > CS_ROUTING -> CS_CONSUME_MEDIA > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:57.156987 [DEBUG] > switch_core_state_machine.c:643 (sofia/internal/15068559226) State > ROUTING going to sleep > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:57.156987 [DEBUG] > switch_core_state_machine.c:584 (sofia/internal/15068559226) Running > State Change CS_CONSUME_MEDIA (Cur 2 Tot 136) > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:57.156987 [DEBUG] > switch_core_state_machine.c:662 (sofia/internal/15068559226) State > CONSUME_MEDIA > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:57.156987 [DEBUG] > switch_core_state_machine.c:662 (sofia/internal/15068559226) State > CONSUME_MEDIA going to sleep > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:57.236974 [DEBUG] > sofia.c:7048 Channel sofia/internal/15068559226 entering state [calling][0] > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.737079 [DEBUG] > sofia.c:7048 Channel sofia/internal/15068559226 entering state > [proceeding][183] > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.737079 [DEBUG] > sofia.c:7058 Remote SDP: > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 v=0 > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 o=FreeSWITCH 1493799827 1493799828 > IN IP4 72.55.158.152 > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 s=FreeSWITCH > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 c=IN IP4 72.55.158.152 > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 t=0 0 > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 m=audio 22274 RTP/AVP 0 101 13 > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 a=rtpmap:0 PCMU/8000 > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 a=rtpmap:101 telephone-event/8000 > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 a=fmtp:101 0-16 > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 a=rtpmap:13 CN/8000 > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 a=ptime:20 > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.737079 [DEBUG] > switch_core_media.c:4436 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[ > PCMU:0:8000:20:64000:1] > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.737079 [DEBUG] > switch_core_media.c:4491 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ > is saved as a match > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.737079 [DEBUG] > switch_core_media.c:4436 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[ > G722:9:8000:20:64000:1] > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.737079 [DEBUG] > switch_core_media.c:4352 Set telephone-event payload to 101 at 8000 > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.737079 [DEBUG] > switch_core_media.c:3056 Set Codec sofia/internal/15068559226 PCMU/8000 20 > ms 160 samples 64000 bits 1 channels > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.737079 [DEBUG] > switch_core_codec.c:111 sofia/internal/15068559226 Original read codec set > to PCMU:0 > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.737079 [DEBUG] > switch_core_media.c:4695 Set telephone-event payload to 101 at 8000 > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.737079 [DEBUG] > switch_core_media.c:4754 sofia/internal/15068559226 Set 2833 dtmf send > payload to 101 recv payload to 101 > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.737079 [DEBUG] > switch_core_media.c:6865 AUDIO RTP [sofia/internal/15068559226] > 192.168.20.161 port 29764 -> 72.55.158.152 port 22274 codec: 0 ms: 20 > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.737079 [DEBUG] > switch_rtp.c:4096 Starting timer [soft] 160 bytes per 20ms > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.737079 [DEBUG] > switch_core_media.c:7166 sofia/internal/15068559226 Set 2833 dtmf send > payload to 101 > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.737079 [DEBUG] > switch_core_media.c:7173 sofia/internal/15068559226 Set 2833 dtmf receive > payload to 101 > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.737079 [DEBUG] > switch_core_media.c:7196 sofia/internal/15068559226 Set rtp dtmf delay to 40 > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.737079 [DEBUG] > switch_core_media.c:7202 Set comfort noise payload to 13 > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.737079 [NOTICE] > sofia_media.c:92 Pre-Answer sofia/internal/15068559226! > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.737079 [DEBUG] > switch_channel.c:3473 (sofia/internal/15068559226) Callstate Change DOWN -> > EARLY > > cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:34:58.757038 [DEBUG] > switch_ivr_originate.c:410 Setting codec string on sofia/internal/ > 200 at 192.168.20.161 to PCMU at 8000h@20i > > cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:34:58.757038 [INFO] > switch_ivr_originate.c:3639 Sending early media > > cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:34:58.757038 [DEBUG] > switch_core_media.c:6865 AUDIO RTP [sofia/internal/200 at 192.168.20.161] > 192.168.20.161 port 16564 -> 192.168.20.150 port 19542 codec: 0 ms: 20 > > cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:34:58.757038 [DEBUG] > switch_rtp.c:4096 Starting timer [soft] 160 bytes per 20ms > > cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:34:58.757038 [DEBUG] > switch_core_media.c:7166 sofia/internal/200 at 192.168.20.161 Set 2833 dtmf > send payload to 101 > > cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:34:58.757038 [DEBUG] > switch_core_media.c:7173 sofia/internal/200 at 192.168.20.161 Set 2833 dtmf > receive payload to 101 > > cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:34:58.757038 [DEBUG] > switch_core_media.c:7196 sofia/internal/200 at 192.168.20.161 Set rtp dtmf > delay to 40 > > cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:34:58.757038 [DEBUG] > mod_sofia.c:2363 Ring SDP: > > cc1d0c18-e576-47a6-bf4f-35ab835f920c v=0 > > cc1d0c18-e576-47a6-bf4f-35ab835f920c o=FreeSWITCH 1493805534 1493805535 > IN IP4 192.168.20.161 > > cc1d0c18-e576-47a6-bf4f-35ab835f920c s=FreeSWITCH > > cc1d0c18-e576-47a6-bf4f-35ab835f920c c=IN IP4 192.168.20.161 > > cc1d0c18-e576-47a6-bf4f-35ab835f920c t=0 0 > > cc1d0c18-e576-47a6-bf4f-35ab835f920c m=audio 16564 RTP/AVP 0 101 > > cc1d0c18-e576-47a6-bf4f-35ab835f920c a=rtpmap:0 PCMU/8000 > > cc1d0c18-e576-47a6-bf4f-35ab835f920c a=rtpmap:101 telephone-event/8000 > > cc1d0c18-e576-47a6-bf4f-35ab835f920c a=fmtp:101 0-16 > > cc1d0c18-e576-47a6-bf4f-35ab835f920c a=ptime:20 > > cc1d0c18-e576-47a6-bf4f-35ab835f920c a=sendrecv > > cc1d0c18-e576-47a6-bf4f-35ab835f920c > > cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:34:58.757038 [NOTICE] > mod_sofia.c:2366 Pre-Answer sofia/internal/200 at 192.168.20.161! > > cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:34:58.757038 [DEBUG] > switch_channel.c:3473 (sofia/internal/200 at 192.168.20.161) Callstate > Change RINGING -> EARLY > > cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:34:58.757038 [DEBUG] > switch_ivr_originate.c:3690 Originate Resulted in Success: > [sofia/internal/15068559226] > > cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:34:58.757038 [DEBUG] > sofia.c:7048 Channel sofia/internal/200 at 192.168.20.161 entering state > [early][183] > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.757038 [DEBUG] > switch_ivr_bridge.c:1601 (sofia/internal/15068559226) State Change > CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.757038 [DEBUG] > switch_core_state_machine.c:584 (sofia/internal/15068559226) Running > State Change CS_EXCHANGE_MEDIA (Cur 2 Tot 136) > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.757038 [DEBUG] > switch_core_state_machine.c:653 (sofia/internal/15068559226) State > EXCHANGE_MEDIA > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.757038 [DEBUG] > mod_sofia.c:631 SOFIA EXCHANGE_MEDIA > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.796985 [DEBUG] > switch_rtp.c:7229 Correct audio ip/port confirmed. > > cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:34:58.877072 [DEBUG] > switch_rtp.c:7229 Correct audio ip/port confirmed. > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:35:00.317073 [DEBUG] > sofia.c:7048 Channel sofia/internal/15068559226 entering state > [completing][200] > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:35:00.317073 [DEBUG] > sofia.c:7055 Duplicate SDP > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 v=0 > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 o=FreeSWITCH 1493799827 1493799828 > IN IP4 72.55.158.152 > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 s=FreeSWITCH > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 c=IN IP4 72.55.158.152 > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 t=0 0 > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 m=audio 22274 RTP/AVP 0 101 13 > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 a=rtpmap:0 PCMU/8000 > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 a=rtpmap:101 telephone-event/8000 > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 a=fmtp:101 0-16 > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 a=rtpmap:13 CN/8000 > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 a=ptime:20 > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:35:00.337073 [DEBUG] > sofia.c:7048 Channel sofia/internal/15068559226 entering state [ready][200] > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:35:00.337073 [NOTICE] > sofia.c:8123 Channel [sofia/internal/15068559226] has been answered > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:35:00.337073 [DEBUG] > switch_channel.c:3772 (sofia/internal/15068559226) Callstate Change EARLY > -> ACTIVE > > cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:35:00.357041 [DEBUG] > switch_core_media.c:6848 Audio params are unchanged for sofia/internal/ > 200 at 192.168.20.161. > > cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:35:00.357041 [DEBUG] > mod_sofia.c:850 Local SDP sofia/internal/200 at 192.168.20.161: > > cc1d0c18-e576-47a6-bf4f-35ab835f920c v=0 > > cc1d0c18-e576-47a6-bf4f-35ab835f920c o=FreeSWITCH 1493805534 1493805536 > IN IP4 192.168.20.161 > > cc1d0c18-e576-47a6-bf4f-35ab835f920c s=FreeSWITCH > > cc1d0c18-e576-47a6-bf4f-35ab835f920c c=IN IP4 192.168.20.161 > > cc1d0c18-e576-47a6-bf4f-35ab835f920c t=0 0 > > cc1d0c18-e576-47a6-bf4f-35ab835f920c m=audio 16564 RTP/AVP 0 101 > > cc1d0c18-e576-47a6-bf4f-35ab835f920c a=rtpmap:0 PCMU/8000 > > cc1d0c18-e576-47a6-bf4f-35ab835f920c a=rtpmap:101 telephone-event/8000 > > cc1d0c18-e576-47a6-bf4f-35ab835f920c a=fmtp:101 0-16 > > cc1d0c18-e576-47a6-bf4f-35ab835f920c a=ptime:20 > > cc1d0c18-e576-47a6-bf4f-35ab835f920c a=sendrecv > > cc1d0c18-e576-47a6-bf4f-35ab835f920c > > cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:35:00.357041 [NOTICE] > switch_ivr_bridge.c:623 Channel [sofia/internal/200 at 192.168.20.161] has > been answered > > cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:35:00.357041 [DEBUG] > switch_channel.c:3772 (sofia/internal/200 at 192.168.20.161) Callstate > Change EARLY -> ACTIVE > > cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:35:00.357041 [DEBUG] > sofia.c:7048 Channel sofia/internal/200 at 192.168.20.161 entering state > [completed][200] > > cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:35:00.377070 [DEBUG] > sofia.c:7048 Channel sofia/internal/200 at 192.168.20.161 entering state > [ready][200] > > cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:35:00.396999 [DEBUG] > switch_rtp.c:7229 Correct audio ip/port confirmed. > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:35:00.396999 [DEBUG] > switch_rtp.c:7229 Correct audio ip/port confirmed. > > 2017-05-03 11:35:00.396999 [INFO] sofia.c:1279 sofia/internal/15068559226 > <(506)%20855-9226> Update Callee ID to "Outbound Call" <+15068559226 > <(506)%20855-9226>> > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:36:02.717070 [DEBUG] > sofia.c:7048 Channel sofia/internal/15068559226 entering state [calling][0] > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:36:03.757073 [DEBUG] > sofia.c:7048 Channel sofia/internal/15068559226 entering state > [completing][200] > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:36:03.757073 [DEBUG] > sofia.c:7055 Duplicate SDP > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 v=0 > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 o=FreeSWITCH 1493799827 1493799828 > IN IP4 72.55.158.152 > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 s=FreeSWITCH > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 c=IN IP4 72.55.158.152 > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 t=0 0 > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 m=audio 22274 RTP/AVP 0 101 13 > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 a=rtpmap:0 PCMU/8000 > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 a=rtpmap:101 telephone-event/8000 > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 a=fmtp:101 0-16 > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 a=rtpmap:13 CN/8000 > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 a=ptime:20 > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:36:03.777033 [DEBUG] > sofia.c:7048 Channel sofia/internal/15068559226 entering state [ready][200] > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:03.737072 [DEBUG] > sofia.c:7048 Channel sofia/internal/15068559226 entering state [calling][0] > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.737038 [DEBUG] > sofia.c:7048 Channel sofia/internal/15068559226 entering state > [terminating][408] > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.737038 [NOTICE] > sofia.c:8237 Hangup sofia/internal/15068559226 [CS_EXCHANGE_MEDIA] > [RECOVERY_ON_TIMER_EXPIRE] > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.737038 [DEBUG] > switch_ivr_bridge.c:712 sofia/internal/15068559226 ending bridge by request > from read function > > *3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.737038 [DEBUG] > switch_ivr_bridge.c:787 BRIDGE THREAD DONE [sofia/internal/15068559226]* > > *3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.737038 [DEBUG] > switch_core_state_machine.c:653 (sofia/internal/15068559226) State > EXCHANGE_MEDIA going to sleep* > > *3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.737038 [DEBUG] > switch_core_state_machine.c:584 (sofia/internal/15068559226) Running State > Change CS_HANGUP (Cur 2 Tot 136)* > > *3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.737038 [DEBUG] > switch_core_state_machine.c:850 (sofia/internal/15068559226) Callstate > Change ACTIVE -> HANGUP* > > *3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.737038 [DEBUG] > switch_core_state_machine.c:852 (sofia/internal/15068559226) State HANGUP* > > *3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.737038 [DEBUG] > mod_sofia.c:438 Channel sofia/internal/15068559226 hanging up, cause: > RECOVERY_ON_TIMER_EXPIRE* > > *3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.737038 [DEBUG] > switch_core_state_machine.c:60 sofia/internal/15068559226 Standard HANGUP, > cause: RECOVERY_ON_TIMER_EXPIRE* > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.737038 [DEBUG] > switch_core_state_machine.c:852 (sofia/internal/15068559226) State HANGUP > going to sleep > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.737038 [DEBUG] > switch_core_state_machine.c:619 (sofia/internal/15068559226) State Change > CS_HANGUP -> CS_REPORTING > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.737038 [DEBUG] > switch_core_state_machine.c:584 (sofia/internal/15068559226) Running > State Change CS_REPORTING (Cur 2 Tot 136) > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.737038 [DEBUG] > switch_core_state_machine.c:938 (sofia/internal/15068559226) State > REPORTING > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.737038 [DEBUG] > switch_core_state_machine.c:174 sofia/internal/15068559226 Standard > REPORTING, cause: RECOVERY_ON_TIMER_EXPIRE > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.737038 [DEBUG] > switch_core_state_machine.c:938 (sofia/internal/15068559226) State > REPORTING going to sleep > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.737038 [DEBUG] > switch_core_state_machine.c:610 (sofia/internal/15068559226) State Change > CS_REPORTING -> CS_DESTROY > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.737038 [DEBUG] > switch_core_session.c:1664 Session 136 (sofia/internal/15068559226) Locked, > Waiting on external entities > > cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.757070 [DEBUG] > switch_ivr_bridge.c:706 sofia/internal/15068559226 ending bridge by request > from write function > > cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.757070 [DEBUG] > switch_ivr_bridge.c:787 BRIDGE THREAD DONE [sofia/internal/200 at 192.168. > 20.161] > > cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.757070 [NOTICE] > switch_ivr_bridge.c:1751 Hangup sofia/internal/200 at 192.168.20.161 > [CS_EXECUTE] [RECOVERY_ON_TIMER_EXPIRE] > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.757070 [NOTICE] > switch_core_session.c:1682 Session 136 (sofia/internal/15068559226) Ended > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.757070 [NOTICE] > switch_core_session.c:1686 Close Channel sofia/internal/15068559226 > [CS_DESTROY] > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.757070 [DEBUG] > switch_core_state_machine.c:741 (sofia/internal/15068559226) Running > State Change CS_DESTROY (Cur 1 Tot 136) > > cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.757070 [DEBUG] > switch_core_session.c:2814 sofia/internal/200 at 192.168.20.161 skip receive > message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) > > cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.757070 [DEBUG] > switch_core_state_machine.c:650 (sofia/internal/200 at 192.168.20.161) State > EXECUTE going to sleep > > cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.757070 [DEBUG] > switch_core_state_machine.c:584 (sofia/internal/200 at 192.168.20.161) > Running State Change CS_HANGUP (Cur 1 Tot 136) > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.757070 [DEBUG] > switch_core_state_machine.c:751 (sofia/internal/15068559226) State DESTROY > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.757070 [DEBUG] > mod_sofia.c:343 sofia/internal/15068559226 SOFIA DESTROY > > cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.757070 [DEBUG] > switch_core_state_machine.c:850 (sofia/internal/200 at 192.168.20.161) > Callstate Change ACTIVE -> HANGUP > > cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.757070 [DEBUG] > switch_core_state_machine.c:852 (sofia/internal/200 at 192.168.20.161) State > HANGUP > > cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.757070 [DEBUG] > mod_sofia.c:432 sofia/internal/200 at 192.168.20.161 Overriding SIP cause > 504 with 408 from the other leg > > cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.757070 [DEBUG] > mod_sofia.c:438 Channel sofia/internal/200 at 192.168.20.161 hanging up, > cause: RECOVERY_ON_TIMER_EXPIRE > > cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.757070 [DEBUG] > mod_sofia.c:491 Sending BYE to sofia/internal/200 at 192.168.20.161 > > cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.757070 [DEBUG] > switch_core_state_machine.c:60 sofia/internal/200 at 192.168.20.161 Standard > HANGUP, cause: RECOVERY_ON_TIMER_EXPIRE > > cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.757070 [DEBUG] > switch_core_state_machine.c:852 (sofia/internal/200 at 192.168.20.161) State > HANGUP going to sleep > > 2017-05-03 11:37:35.757070 [DEBUG] switch_nat.c:542 unmapped public port > 29764 protocol UDP to localport 29764 > > cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.757070 [DEBUG] > switch_core_state_machine.c:619 (sofia/internal/200 at 192.168.20.161) State > Change CS_HANGUP -> CS_REPORTING > > cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.757070 [DEBUG] > switch_core_state_machine.c:584 (sofia/internal/200 at 192.168.20.161) > Running State Change CS_REPORTING (Cur 1 Tot 136) > > cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.757070 [DEBUG] > switch_core_state_machine.c:938 (sofia/internal/200 at 192.168.20.161) State > REPORTING > > 2017-05-03 11:37:35.757070 [DEBUG] switch_nat.c:542 unmapped public port > 29765 protocol UDP to localport 29765 > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.757070 [DEBUG] > switch_core_state_machine.c:181 sofia/internal/15068559226 Standard > DESTROY > > 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.757070 [DEBUG] > switch_core_state_machine.c:751 (sofia/internal/15068559226) State > DESTROY going to sleep > > cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.776979 [DEBUG] > switch_core_state_machine.c:174 sofia/internal/200 at 192.168.20.161 > Standard REPORTING, cause: RECOVERY_ON_TIMER_EXPIRE > > cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.776979 [DEBUG] > switch_core_state_machine.c:938 (sofia/internal/200 at 192.168.20.161) State > REPORTING going to sleep > > cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.776979 [DEBUG] > switch_core_state_machine.c:610 (sofia/internal/200 at 192.168.20.161) State > Change CS_REPORTING -> CS_DESTROY > > cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.776979 [DEBUG] > switch_core_session.c:1664 Session 135 (sofia/internal/200 at 192.168.20.161) > Locked, Waiting on external entities > > cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.776979 [NOTICE] > switch_core_session.c:1682 Session 135 (sofia/internal/200 at 192.168.20.161) > Ended > > cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.776979 [NOTICE] > switch_core_session.c:1686 Close Channel sofia/internal/200 at 192.168.20.161 > [CS_DESTROY] > > cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.776979 [DEBUG] > switch_core_state_machine.c:741 (sofia/internal/200 at 192.168.20.161) > Running State Change CS_DESTROY (Cur 0 Tot 136) > > cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.796986 [DEBUG] > switch_core_state_machine.c:751 (sofia/internal/200 at 192.168.20.161) State > DESTROY > > cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.796986 [DEBUG] > mod_sofia.c:343 sofia/internal/200 at 192.168.20.161 SOFIA DESTROY > > cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.796986 [DEBUG] > switch_core_state_machine.c:181 sofia/internal/200 at 192.168.20.161 > Standard DESTROY > > cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.796986 [DEBUG] > switch_core_state_machine.c:751 (sofia/internal/200 at 192.168.20.161) State > DESTROY going to sleep > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Book a phone call (CST) Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170503/338a398b/attachment-0001.html From anthony.minessale at gmail.com Wed May 3 19:44:24 2017 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 3 May 2017 10:44:24 -0500 Subject: [Freeswitch-users] Outbound calls being dropped after update to Version 1.6.17 -34-0fc0946 64bit In-Reply-To: References: <006d01d2c41d$294be0e0$7be3a2a0$@accra.ca> Message-ID: Also, please do not paste logs into the mailing list or report bugs into the mailing list. BUGS belong on jira. On Wed, May 3, 2017 at 10:05 AM, Brian West wrote: > This is a NAT issue, 'sofia global siptrace on' and watch the signaling. > > /b > > > 2017-05-03 9:54 GMT-05:00 Charles Bujold : > >> Still no solutions, Here is a >> log of a call. Any suggestion would be appreciated >> >> >> >> >> >> >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:57.156987 [DEBUG] >> sofia.c:7048 Channel sofia/internal/15068559226 <(506)%20855-9226> >> entering state [calling][0] >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:57.156987 [DEBUG] >> switch_core_state_machine.c:643 (sofia/internal/15068559226 >> <(506)%20855-9226>) State ROUTING >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:57.156987 [DEBUG] >> mod_sofia.c:143 sofia/internal/15068559226 <(506)%20855-9226> SOFIA >> ROUTING >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:57.156987 [DEBUG] >> switch_ivr_originate.c:67 (sofia/internal/15068559226 <(506)%20855-9226>) >> State Change CS_ROUTING -> CS_CONSUME_MEDIA >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:57.156987 [DEBUG] >> switch_core_state_machine.c:643 (sofia/internal/15068559226 >> <(506)%20855-9226>) State ROUTING going to sleep >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:57.156987 [DEBUG] >> switch_core_state_machine.c:584 (sofia/internal/15068559226 >> <(506)%20855-9226>) Running State Change CS_CONSUME_MEDIA (Cur 2 Tot 136) >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:57.156987 [DEBUG] >> switch_core_state_machine.c:662 (sofia/internal/15068559226 >> <(506)%20855-9226>) State CONSUME_MEDIA >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:57.156987 [DEBUG] >> switch_core_state_machine.c:662 (sofia/internal/15068559226 >> <(506)%20855-9226>) State CONSUME_MEDIA going to sleep >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:57.236974 [DEBUG] >> sofia.c:7048 Channel sofia/internal/15068559226 <(506)%20855-9226> >> entering state [calling][0] >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.737079 [DEBUG] >> sofia.c:7048 Channel sofia/internal/15068559226 <(506)%20855-9226> >> entering state [proceeding][183] >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.737079 [DEBUG] >> sofia.c:7058 Remote SDP: >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 v=0 >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 o=FreeSWITCH 1493799827 1493799828 >> IN IP4 72.55.158.152 >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 s=FreeSWITCH >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 c=IN IP4 72.55.158.152 >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 t=0 0 >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 m=audio 22274 RTP/AVP 0 101 13 >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 a=rtpmap:0 PCMU/8000 >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 a=rtpmap:101 telephone-event/8000 >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 a=fmtp:101 0-16 >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 a=rtpmap:13 CN/8000 >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 a=ptime:20 >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.737079 [DEBUG] >> switch_core_media.c:4436 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU >> :0:8000:20:64000:1] >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.737079 [DEBUG] >> switch_core_media.c:4491 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ >> is saved as a match >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.737079 [DEBUG] >> switch_core_media.c:4436 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[G722 >> :9:8000:20:64000:1] >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.737079 [DEBUG] >> switch_core_media.c:4352 Set telephone-event payload to 101 at 8000 >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.737079 [DEBUG] >> switch_core_media.c:3056 Set Codec sofia/internal/15068559226 >> <(506)%20855-9226> PCMU/8000 20 ms 160 samples 64000 bits 1 channels >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.737079 [DEBUG] >> switch_core_codec.c:111 sofia/internal/15068559226 <(506)%20855-9226> >> Original read codec set to PCMU:0 >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.737079 [DEBUG] >> switch_core_media.c:4695 Set telephone-event payload to 101 at 8000 >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.737079 [DEBUG] >> switch_core_media.c:4754 sofia/internal/15068559226 <(506)%20855-9226> >> Set 2833 dtmf send payload to 101 recv payload to 101 >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.737079 [DEBUG] >> switch_core_media.c:6865 AUDIO RTP [sofia/internal/15068559226] >> 192.168.20.161 port 29764 -> 72.55.158.152 port 22274 codec: 0 ms: 20 >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.737079 [DEBUG] >> switch_rtp.c:4096 Starting timer [soft] 160 bytes per 20ms >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.737079 [DEBUG] >> switch_core_media.c:7166 sofia/internal/15068559226 <(506)%20855-9226> >> Set 2833 dtmf send payload to 101 >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.737079 [DEBUG] >> switch_core_media.c:7173 sofia/internal/15068559226 <(506)%20855-9226> >> Set 2833 dtmf receive payload to 101 >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.737079 [DEBUG] >> switch_core_media.c:7196 sofia/internal/15068559226 <(506)%20855-9226> >> Set rtp dtmf delay to 40 >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.737079 [DEBUG] >> switch_core_media.c:7202 Set comfort noise payload to 13 >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.737079 [NOTICE] >> sofia_media.c:92 Pre-Answer sofia/internal/15068559226 <(506)%20855-9226> >> ! >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.737079 [DEBUG] >> switch_channel.c:3473 (sofia/internal/15068559226 <(506)%20855-9226>) >> Callstate Change DOWN -> EARLY >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:34:58.757038 [DEBUG] >> switch_ivr_originate.c:410 Setting codec string on sofia/internal/ >> 200 at 192.168.20.161 to PCMU at 8000h@20i >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:34:58.757038 [INFO] >> switch_ivr_originate.c:3639 Sending early media >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:34:58.757038 [DEBUG] >> switch_core_media.c:6865 AUDIO RTP [sofia/internal/200 at 192.168.20.161] >> 192.168.20.161 port 16564 -> 192.168.20.150 port 19542 codec: 0 ms: 20 >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:34:58.757038 [DEBUG] >> switch_rtp.c:4096 Starting timer [soft] 160 bytes per 20ms >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:34:58.757038 [DEBUG] >> switch_core_media.c:7166 sofia/internal/200 at 192.168.20.161 Set 2833 dtmf >> send payload to 101 >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:34:58.757038 [DEBUG] >> switch_core_media.c:7173 sofia/internal/200 at 192.168.20.161 Set 2833 dtmf >> receive payload to 101 >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:34:58.757038 [DEBUG] >> switch_core_media.c:7196 sofia/internal/200 at 192.168.20.161 Set rtp dtmf >> delay to 40 >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:34:58.757038 [DEBUG] >> mod_sofia.c:2363 Ring SDP: >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c v=0 >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c o=FreeSWITCH 1493805534 1493805535 >> IN IP4 192.168.20.161 >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c s=FreeSWITCH >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c c=IN IP4 192.168.20.161 >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c t=0 0 >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c m=audio 16564 RTP/AVP 0 101 >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c a=rtpmap:0 PCMU/8000 >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c a=rtpmap:101 telephone-event/8000 >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c a=fmtp:101 0-16 >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c a=ptime:20 >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c a=sendrecv >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:34:58.757038 [NOTICE] >> mod_sofia.c:2366 Pre-Answer sofia/internal/200 at 192.168.20.161! >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:34:58.757038 [DEBUG] >> switch_channel.c:3473 (sofia/internal/200 at 192.168.20.161) Callstate >> Change RINGING -> EARLY >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:34:58.757038 [DEBUG] >> switch_ivr_originate.c:3690 Originate Resulted in Success: [sofia/internal/ >> 15068559226 <(506)%20855-9226>] >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:34:58.757038 [DEBUG] >> sofia.c:7048 Channel sofia/internal/200 at 192.168.20.161 entering state >> [early][183] >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.757038 [DEBUG] >> switch_ivr_bridge.c:1601 (sofia/internal/15068559226 <(506)%20855-9226>) >> State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.757038 [DEBUG] >> switch_core_state_machine.c:584 (sofia/internal/15068559226 >> <(506)%20855-9226>) Running State Change CS_EXCHANGE_MEDIA (Cur 2 Tot >> 136) >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.757038 [DEBUG] >> switch_core_state_machine.c:653 (sofia/internal/15068559226 >> <(506)%20855-9226>) State EXCHANGE_MEDIA >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.757038 [DEBUG] >> mod_sofia.c:631 SOFIA EXCHANGE_MEDIA >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:34:58.796985 [DEBUG] >> switch_rtp.c:7229 Correct audio ip/port confirmed. >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:34:58.877072 [DEBUG] >> switch_rtp.c:7229 Correct audio ip/port confirmed. >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:35:00.317073 [DEBUG] >> sofia.c:7048 Channel sofia/internal/15068559226 <(506)%20855-9226> >> entering state [completing][200] >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:35:00.317073 [DEBUG] >> sofia.c:7055 Duplicate SDP >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 v=0 >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 o=FreeSWITCH 1493799827 1493799828 >> IN IP4 72.55.158.152 >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 s=FreeSWITCH >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 c=IN IP4 72.55.158.152 >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 t=0 0 >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 m=audio 22274 RTP/AVP 0 101 13 >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 a=rtpmap:0 PCMU/8000 >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 a=rtpmap:101 telephone-event/8000 >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 a=fmtp:101 0-16 >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 a=rtpmap:13 CN/8000 >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 a=ptime:20 >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:35:00.337073 [DEBUG] >> sofia.c:7048 Channel sofia/internal/15068559226 <(506)%20855-9226> >> entering state [ready][200] >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:35:00.337073 [NOTICE] >> sofia.c:8123 Channel [sofia/internal/15068559226 <(506)%20855-9226>] has >> been answered >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:35:00.337073 [DEBUG] >> switch_channel.c:3772 (sofia/internal/15068559226 <(506)%20855-9226>) Callstate >> Change EARLY -> ACTIVE >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:35:00.357041 [DEBUG] >> switch_core_media.c:6848 Audio params are unchanged for sofia/internal/ >> 200 at 192.168.20.161. >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:35:00.357041 [DEBUG] >> mod_sofia.c:850 Local SDP sofia/internal/200 at 192.168.20.161: >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c v=0 >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c o=FreeSWITCH 1493805534 1493805536 >> IN IP4 192.168.20.161 >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c s=FreeSWITCH >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c c=IN IP4 192.168.20.161 >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c t=0 0 >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c m=audio 16564 RTP/AVP 0 101 >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c a=rtpmap:0 PCMU/8000 >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c a=rtpmap:101 telephone-event/8000 >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c a=fmtp:101 0-16 >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c a=ptime:20 >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c a=sendrecv >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:35:00.357041 [NOTICE] >> switch_ivr_bridge.c:623 Channel [sofia/internal/200 at 192.168.20.161] has >> been answered >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:35:00.357041 [DEBUG] >> switch_channel.c:3772 (sofia/internal/200 at 192.168.20.161) Callstate >> Change EARLY -> ACTIVE >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:35:00.357041 [DEBUG] >> sofia.c:7048 Channel sofia/internal/200 at 192.168.20.161 entering state >> [completed][200] >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:35:00.377070 [DEBUG] >> sofia.c:7048 Channel sofia/internal/200 at 192.168.20.161 entering state >> [ready][200] >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:35:00.396999 [DEBUG] >> switch_rtp.c:7229 Correct audio ip/port confirmed. >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:35:00.396999 [DEBUG] >> switch_rtp.c:7229 Correct audio ip/port confirmed. >> >> 2017-05-03 11:35:00.396999 [INFO] sofia.c:1279 sofia/internal/15068559226 >> <(506)%20855-9226> Update Callee ID to "Outbound Call" <+15068559226 >> <(506)%20855-9226>> >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:36:02.717070 [DEBUG] >> sofia.c:7048 Channel sofia/internal/15068559226 <(506)%20855-9226> >> entering state [calling][0] >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:36:03.757073 [DEBUG] >> sofia.c:7048 Channel sofia/internal/15068559226 <(506)%20855-9226> >> entering state [completing][200] >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:36:03.757073 [DEBUG] >> sofia.c:7055 Duplicate SDP >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 v=0 >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 o=FreeSWITCH 1493799827 1493799828 >> IN IP4 72.55.158.152 >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 s=FreeSWITCH >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 c=IN IP4 72.55.158.152 >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 t=0 0 >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 m=audio 22274 RTP/AVP 0 101 13 >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 a=rtpmap:0 PCMU/8000 >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 a=rtpmap:101 telephone-event/8000 >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 a=fmtp:101 0-16 >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 a=rtpmap:13 CN/8000 >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 a=ptime:20 >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:36:03.777033 [DEBUG] >> sofia.c:7048 Channel sofia/internal/15068559226 <(506)%20855-9226> >> entering state [ready][200] >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:03.737072 [DEBUG] >> sofia.c:7048 Channel sofia/internal/15068559226 <(506)%20855-9226> >> entering state [calling][0] >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.737038 [DEBUG] >> sofia.c:7048 Channel sofia/internal/15068559226 <(506)%20855-9226> >> entering state [terminating][408] >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.737038 [NOTICE] >> sofia.c:8237 Hangup sofia/internal/15068559226 <(506)%20855-9226> >> [CS_EXCHANGE_MEDIA] [RECOVERY_ON_TIMER_EXPIRE] >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.737038 [DEBUG] >> switch_ivr_bridge.c:712 sofia/internal/15068559226 <(506)%20855-9226> >> ending bridge by request from read function >> >> *3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.737038 [DEBUG] >> switch_ivr_bridge.c:787 BRIDGE THREAD DONE [sofia/internal/15068559226 >> <(506)%20855-9226>]* >> >> *3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.737038 [DEBUG] >> switch_core_state_machine.c:653 (sofia/internal/15068559226 >> <(506)%20855-9226>) State EXCHANGE_MEDIA going to sleep* >> >> *3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.737038 [DEBUG] >> switch_core_state_machine.c:584 (sofia/internal/15068559226 >> <(506)%20855-9226>) Running State Change CS_HANGUP (Cur 2 Tot 136)* >> >> *3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.737038 [DEBUG] >> switch_core_state_machine.c:850 (sofia/internal/15068559226 >> <(506)%20855-9226>) Callstate Change ACTIVE -> HANGUP* >> >> *3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.737038 [DEBUG] >> switch_core_state_machine.c:852 (sofia/internal/15068559226 >> <(506)%20855-9226>) State HANGUP* >> >> *3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.737038 [DEBUG] >> mod_sofia.c:438 Channel sofia/internal/15068559226 <(506)%20855-9226> >> hanging up, cause: RECOVERY_ON_TIMER_EXPIRE* >> >> *3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.737038 [DEBUG] >> switch_core_state_machine.c:60 sofia/internal/15068559226 >> <(506)%20855-9226> Standard HANGUP, cause: RECOVERY_ON_TIMER_EXPIRE* >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.737038 [DEBUG] >> switch_core_state_machine.c:852 (sofia/internal/15068559226 >> <(506)%20855-9226>) State HANGUP going to sleep >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.737038 [DEBUG] >> switch_core_state_machine.c:619 (sofia/internal/15068559226 >> <(506)%20855-9226>) State Change CS_HANGUP -> CS_REPORTING >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.737038 [DEBUG] >> switch_core_state_machine.c:584 (sofia/internal/15068559226 >> <(506)%20855-9226>) Running State Change CS_REPORTING (Cur 2 Tot 136) >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.737038 [DEBUG] >> switch_core_state_machine.c:938 (sofia/internal/15068559226 >> <(506)%20855-9226>) State REPORTING >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.737038 [DEBUG] >> switch_core_state_machine.c:174 sofia/internal/15068559226 >> <(506)%20855-9226> Standard REPORTING, cause: RECOVERY_ON_TIMER_EXPIRE >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.737038 [DEBUG] >> switch_core_state_machine.c:938 (sofia/internal/15068559226 >> <(506)%20855-9226>) State REPORTING going to sleep >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.737038 [DEBUG] >> switch_core_state_machine.c:610 (sofia/internal/15068559226 >> <(506)%20855-9226>) State Change CS_REPORTING -> CS_DESTROY >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.737038 [DEBUG] >> switch_core_session.c:1664 Session 136 (sofia/internal/15068559226) Locked, >> Waiting on external entities >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.757070 [DEBUG] >> switch_ivr_bridge.c:706 sofia/internal/15068559226 <(506)%20855-9226> >> ending bridge by request from write function >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.757070 [DEBUG] >> switch_ivr_bridge.c:787 BRIDGE THREAD DONE [sofia/internal/200 at 192.168.20 >> .161] >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.757070 [NOTICE] >> switch_ivr_bridge.c:1751 Hangup sofia/internal/200 at 192.168.20.161 >> [CS_EXECUTE] [RECOVERY_ON_TIMER_EXPIRE] >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.757070 [NOTICE] >> switch_core_session.c:1682 Session 136 (sofia/internal/15068559226) Ended >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.757070 [NOTICE] >> switch_core_session.c:1686 Close Channel sofia/internal/15068559226 >> <(506)%20855-9226> [CS_DESTROY] >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.757070 [DEBUG] >> switch_core_state_machine.c:741 (sofia/internal/15068559226 >> <(506)%20855-9226>) Running State Change CS_DESTROY (Cur 1 Tot 136) >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.757070 [DEBUG] >> switch_core_session.c:2814 sofia/internal/200 at 192.168.20.161 skip >> receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.757070 [DEBUG] >> switch_core_state_machine.c:650 (sofia/internal/200 at 192.168.20.161) >> State EXECUTE going to sleep >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.757070 [DEBUG] >> switch_core_state_machine.c:584 (sofia/internal/200 at 192.168.20.161) >> Running State Change CS_HANGUP (Cur 1 Tot 136) >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.757070 [DEBUG] >> switch_core_state_machine.c:751 (sofia/internal/15068559226 >> <(506)%20855-9226>) State DESTROY >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.757070 [DEBUG] >> mod_sofia.c:343 sofia/internal/15068559226 <(506)%20855-9226> SOFIA >> DESTROY >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.757070 [DEBUG] >> switch_core_state_machine.c:850 (sofia/internal/200 at 192.168.20.161) >> Callstate Change ACTIVE -> HANGUP >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.757070 [DEBUG] >> switch_core_state_machine.c:852 (sofia/internal/200 at 192.168.20.161) >> State HANGUP >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.757070 [DEBUG] >> mod_sofia.c:432 sofia/internal/200 at 192.168.20.161 Overriding SIP cause >> 504 with 408 from the other leg >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.757070 [DEBUG] >> mod_sofia.c:438 Channel sofia/internal/200 at 192.168.20.161 hanging up, >> cause: RECOVERY_ON_TIMER_EXPIRE >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.757070 [DEBUG] >> mod_sofia.c:491 Sending BYE to sofia/internal/200 at 192.168.20.161 >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.757070 [DEBUG] >> switch_core_state_machine.c:60 sofia/internal/200 at 192.168.20.161 >> Standard HANGUP, cause: RECOVERY_ON_TIMER_EXPIRE >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.757070 [DEBUG] >> switch_core_state_machine.c:852 (sofia/internal/200 at 192.168.20.161) >> State HANGUP going to sleep >> >> 2017-05-03 11:37:35.757070 [DEBUG] switch_nat.c:542 unmapped public port >> 29764 protocol UDP to localport 29764 >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.757070 [DEBUG] >> switch_core_state_machine.c:619 (sofia/internal/200 at 192.168.20.161) >> State Change CS_HANGUP -> CS_REPORTING >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.757070 [DEBUG] >> switch_core_state_machine.c:584 (sofia/internal/200 at 192.168.20.161) >> Running State Change CS_REPORTING (Cur 1 Tot 136) >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.757070 [DEBUG] >> switch_core_state_machine.c:938 (sofia/internal/200 at 192.168.20.161) >> State REPORTING >> >> 2017-05-03 11:37:35.757070 [DEBUG] switch_nat.c:542 unmapped public port >> 29765 protocol UDP to localport 29765 >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.757070 [DEBUG] >> switch_core_state_machine.c:181 sofia/internal/15068559226 >> <(506)%20855-9226> Standard DESTROY >> >> 3960b42a-6c36-49c9-96a0-a844b3f0f259 2017-05-03 11:37:35.757070 [DEBUG] >> switch_core_state_machine.c:751 (sofia/internal/15068559226 >> <(506)%20855-9226>) State DESTROY going to sleep >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.776979 [DEBUG] >> switch_core_state_machine.c:174 sofia/internal/200 at 192.168.20.161 >> Standard REPORTING, cause: RECOVERY_ON_TIMER_EXPIRE >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.776979 [DEBUG] >> switch_core_state_machine.c:938 (sofia/internal/200 at 192.168.20.161) >> State REPORTING going to sleep >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.776979 [DEBUG] >> switch_core_state_machine.c:610 (sofia/internal/200 at 192.168.20.161) >> State Change CS_REPORTING -> CS_DESTROY >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.776979 [DEBUG] >> switch_core_session.c:1664 Session 135 (sofia/internal/200 at 192.168.20.161) >> Locked, Waiting on external entities >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.776979 [NOTICE] >> switch_core_session.c:1682 Session 135 (sofia/internal/200 at 192.168.20.161) >> Ended >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.776979 [NOTICE] >> switch_core_session.c:1686 Close Channel sofia/internal/200 at 192.168.20. >> 161 [CS_DESTROY] >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.776979 [DEBUG] >> switch_core_state_machine.c:741 (sofia/internal/200 at 192.168.20.161) >> Running State Change CS_DESTROY (Cur 0 Tot 136) >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.796986 [DEBUG] >> switch_core_state_machine.c:751 (sofia/internal/200 at 192.168.20.161) >> State DESTROY >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.796986 [DEBUG] >> mod_sofia.c:343 sofia/internal/200 at 192.168.20.161 SOFIA DESTROY >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.796986 [DEBUG] >> switch_core_state_machine.c:181 sofia/internal/200 at 192.168.20.161 >> Standard DESTROY >> >> cc1d0c18-e576-47a6-bf4f-35ab835f920c 2017-05-03 11:37:35.796986 [DEBUG] >> switch_core_state_machine.c:751 (sofia/internal/200 at 192.168.20.161) >> State DESTROY going to sleep >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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.org > > *Twitter: @FreeSWITCH , @briankwest* > > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > Book a phone call (CST) > > Allison prompts for FreeSWITCH: > > *https://www.gofundme.com/allison-prompts-for-freeswitch* > > > Got Bugs? Report them here ! | Reddit: > /r/freeswitch > > *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002> > | *M:*+1918424WEST (9378) > *Skype:*briankwest > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170503/851c6600/attachment-0001.html From brian at freeswitch.org Wed May 3 19:59:54 2017 From: brian at freeswitch.org (Brian West) Date: Wed, 3 May 2017 10:59:54 -0500 Subject: [Freeswitch-users] Fax receive with INCOMPATIBLE_DESTINATION In-Reply-To: References: Message-ID: Not helpful, Get the full debug logs with sip trace on and you'll need to report bugs to freeswitch.org/jira /b On Wed, May 3, 2017 at 5:17 AM, Hardik Patel wrote: > Hello, > > I am using opensips as entry point using dispatcher to opensips( > 127.0.0.1), i am routing call to freeswitch server (127.0.0.2). > > Now I am trying to send fax using span_dsp modules, my issue is when i > receive a fax perfectly but in fs_cli i receive 'INCOMPATIBLE_DESTINATION' > ,i need to 'NORMAL_CLEARING'. > > 127.0.0.3 => carrier/provider IP > 98989898 => Fax number > > and here i am attaching sip trace. > > -- > Hardik Patel > iNextrix Technologies Pvt Ltd > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Book a phone call (CST) Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170503/690b18f4/attachment.html From roamer2998 at gmail.com Wed May 3 23:35:14 2017 From: roamer2998 at gmail.com (Roamer2998) Date: Thu, 4 May 2017 03:35:14 +0800 Subject: [Freeswitch-users] IP PBX selection In-Reply-To: References: Message-ID: Hi Colton, We have tested PortSIP PBX free version(https://www.portsip.com/portsip-pbx), it's very easy to config and manage, the usage almost likes the 3CX( https://www.3cx.com)! Since we are new to VoIP, that's why we chose them, and they also provide their VoIP SDK to us, so far the PBX is works fine. As PortSIP described, their PBX isn't base on freeswitch or other open source PBX, they said expect the reSIProcate stack and WebRTC media engine, other codes all are developed by their team. BR On Fri, Apr 28, 2017 at 7:45 AM, Colton Conor wrote: > I at first through by PortSIP you meant PortaSwitch by PortaOne, but now > looking at it PortSIP is something entirely different. I have never heard > about this company, so I am a little leary on your selection. What doe > PortSIP use? Do they user Freeswitch? > > On Fri, Apr 21, 2017 at 3:18 AM, Roamer2998 wrote: > >> Hi all, >> >> Finally we make decision that go with PortSIP, the reasons are below: >> >> 1. Support the easy cluster deployment for handle large concurrent calls >> and provide >> 2. All REST API(this is very important to us for integrate the PBX with >> our current system), and also offer the rebrand app for free. >> 3. The multi-tenant arch. >> >> Thanks all for your suggestions, we have learned a lot of ! >> >> BR >> >> On Mon, Apr 17, 2017 at 2:33 PM, Roamer2998 wrote: >> >>> Hi all, I'm new to VoIP, now we have a project that needs a >>> PBX with client APPs. >>> In our team we have argument for choosing PBX. By so far, we >>> have following candidates: >>> >>> A: Open source >>> >>> 1) Asterisk PBX (http://www.asterisk.org) (with longest >>> history that almost every one knows it) >>> 2) FreeSwitch (http://www.freeswitch.org) (A lot people >>> recommended it to us) >>> >>> >>> B: Commercial >>> >>> 1) Vodia PBX (http://www.vodia.com). It comes from SNOM, >>> but acquired by a HongKong company now >>> 2) PortSIP PBX (http://www.portsip.com/portsip-pbx). It >>> also includes VoIP SDK, WebRTC and offer rebranding app for free. >>> >>> My boss prefers the Open Source PBX since they are free, but >>> our CTO prefers the commercial editions, according to whom >>> the business PBX has better support, and the performance >>> is good, and easy to use - considering our team all are new to VoIP/PBX. >>> >>> We have did some searching of FreeSwitch, it seems the sofia >>> stack comes from Nodia. >>> Here's my question: >>> >>> 1. Does the Nokia still maintain this stack or not? >>> 2. Any commercial support by FreeSwitch developers? Can I go >>> t the quotation? >>> 3. Is it easy to compile and setup FreeSwitch? >>> 4. Which FreeSwitch version is recommended? ( 1.6?) >>> >>> Thanks in advance . >>> >>> 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170504/4e2490ef/attachment.html From koki.roul at gmail.com Wed May 3 21:42:07 2017 From: koki.roul at gmail.com (Lyubo Popov) Date: Wed, 3 May 2017 14:42:07 -0300 Subject: [Freeswitch-users] 300 Multiple Choices - how to force FS to replace From field Message-ID: Hello everyone, I would like to express my thanks in advance to anyone who may be able to help me with some insides. I am using a routing software with SIP Redirect to send routes to FS with 300 Multiple Choices and mod_xml_radius to authenticate the SIP users. In the Sip redirect server I am manipulating as well the FROM number and sending back to FS, but FS will not respect this and continue using the SIP account that sent the call in the first place in the FROM field. Here are some SIP packets from both sides to clarify the whole process. 1. Sending call to FS with Zoiper, destination 556230951662 INVITE sip:556230951662 at 216.x.x.x:5080;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 177.x.x.x:1048;branch=z9hG4bK-d8754z-038f1c7d251308c2-1---d8754z-;rport Max-Forwards: 70 Contact: To: From: "551000";tag=25599d20 Call-ID: NmUzYTAwNmQ1NTZjMDM2ZjVhYTgzMDdiY2RiMmI0ZTc. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Zoiper for Windows 2.43 r24984 Allow-Events: presence, kpml Content-Length: 232 v=0 o=Zoiper_user 0 0 IN IP4 177.x.x.x s=Zoiper_session c=IN IP4 177.x.x.x t=0 0 m=audio 8000 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv 2. FS sending INVITE to SIP Redirect server INVITE sip:556230951662 at 69.x.x.x:5060 SIP/2.0 Via: SIP/2.0/UDP 216.245.218.230;rport;branch=z9hG4bKateZg87rDBpZa Max-Forwards: 69 From: "551000" ;tag=FeNXS71300N0c To: Call-ID: e70ee145-aac6-1235-79ba-002590a0ec9b CSeq: 106579790 INVITE Contact: User-Agent: FreeSWITCH 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: 397 X-FS-Support: update_display,send_info Remote-Party-ID: "551000" ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1493809233 1493809234 IN IP4 216.x.x.x 2017-05-03 12:15:09.816119 [ERR] mod_xml_radius.c:911 Didn't match: 69.x.x.x:5060 == ^69\.x\.x\.x s=FreeSWITCH c=IN IP4 216.x.x.x t=0 0 m=audio 22476 RTP/AVP 8 0 18 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 m=audio 22476 RTP/AVP 4 101 13 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:30 3. SIP Redirect returns 300 Multiple choices with the termination IP in Contact and with FROM field as instructed ( update 551000 with 1140031556) SIP/2.0 300 Multiple Choices Via: SIP/2.0/UDP 216.x.x.x;rport;branch=z9hG4bKateZg87rDBpZa From: "1140031556" ;tag=FeNXS71300N0c To: Contact: ;q=1.00 Call-ID: e70ee145-aac6-1235-79ba-002590a0ec9b CSeq: 106579790 INVITE Max-Forwards: 69 Content-Length: 0 Server: SIP Redirect Server 4. FS will send the call to the Termination IP WITHOUT changing the FROM field INVITE sip:556230951662 at 162.x.x.x:5060 SIP/2.0 Via: SIP/2.0/UDP 216.x.x.x;rport;branch=z9hG4bKB37Qj3rvamcjp Max-Forwards: 68 From: "551000" ;tag=gQepU2j7X9BKr To: Call-ID: e7159715-aac6-1235-79ba-002590a0ec9b CSeq: 106579790 INVITE Contact: 2017-05-03 12:15:09.856127 [ERR] mod_xml_radius.c:914 Result of true match: 162.x.x.x:5060 == ^69\.x\.x\.x User-Agent: FreeSWITCH 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: 397 X-FS-Support: update_display,send_info Remote-Party-ID: "551000" ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1493811013 1493811014 IN IP4 216.x.x.x s=FreeSWITCH c=IN IP4 216.x.x.x t=0 0 m=audio 20696 RTP/AVP 8 0 18 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 m=audio 20696 RTP/AVP 4 101 13 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:30 This is the dialplan I use to send calls to SIP redirect server References: Message-ID: Why are you using the from field for authentication? On Wed, May 3, 2017 at 12:42 PM, Lyubo Popov wrote: > Hello everyone, > > I would like to express my thanks in advance to anyone who may be able to > help me with some insides. > > I am using a routing software with SIP Redirect to send routes to FS with > 300 Multiple Choices and mod_xml_radius to authenticate the SIP users. In > the Sip redirect server I am manipulating as well the FROM number and > sending back to FS, but FS will not respect this and continue using the SIP > account that sent the call in the first place in the FROM field. Here are > some SIP packets from both sides to clarify the whole process. > > 1. Sending call to FS with Zoiper, destination 556230951662 > > INVITE sip:556230951662 at 216.x.x.x:5080;transport=UDP SIP/2.0 > Via: SIP/2.0/UDP 177.x.x.x:1048;branch=z9hG4bK- > d8754z-038f1c7d251308c2-1---d8754z-;rport > Max-Forwards: 70 > Contact: > To: > From: "551000";tag=25599d20 > Call-ID: NmUzYTAwNmQ1NTZjMDM2ZjVhYTgzMDdiY2RiMmI0ZTc. > CSeq: 1 INVITE > Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, > SUBSCRIBE > Content-Type: application/sdp > Supported: replaces, norefersub, extended-refer, timer, > X-cisco-serviceuri > User-Agent: Zoiper for Windows 2.43 r24984 > Allow-Events: presence, kpml > Content-Length: 232 > > v=0 > o=Zoiper_user 0 0 IN IP4 177.x.x.x > s=Zoiper_session > c=IN IP4 177.x.x.x > t=0 0 > m=audio 8000 RTP/AVP 8 0 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=sendrecv > > > 2. FS sending INVITE to SIP Redirect server > > INVITE sip:556230951662 at 69.x.x.x:5060 SIP/2.0 > Via: SIP/2.0/UDP 216.245.218.230;rport;branch=z9hG4bKateZg87rDBpZa > Max-Forwards: 69 > From: "551000" ;tag=FeNXS71300N0c > To: > Call-ID: e70ee145-aac6-1235-79ba-002590a0ec9b > CSeq: 106579790 INVITE > Contact: > User-Agent: FreeSWITCH > 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: 397 > X-FS-Support: update_display,send_info > Remote-Party-ID: "551000" ;party= > calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 1493809233 1493809234 IN IP4 216.x.x.x > 2017-05-03 12:15:09.816119 [ERR] mod_xml_radius.c:911 Didn't match: > 69.x.x.x:5060 == ^69\.x\.x\.x > s=FreeSWITCH > c=IN IP4 216.x.x.x > t=0 0 > m=audio 22476 RTP/AVP 8 0 18 101 13 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > m=audio 22476 RTP/AVP 4 101 13 > a=rtpmap:4 G723/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:30 > > > 3. SIP Redirect returns 300 Multiple choices with the termination IP in > Contact and with FROM field as instructed ( update 551000 with 1140031556) > > SIP/2.0 300 Multiple Choices > Via: SIP/2.0/UDP 216.x.x.x;rport;branch=z9hG4bKateZg87rDBpZa > From: "1140031556" ;tag=FeNXS71300N0c > To: > Contact: ;q=1.00 > Call-ID: e70ee145-aac6-1235-79ba-002590a0ec9b > CSeq: 106579790 INVITE > Max-Forwards: 69 > Content-Length: 0 > Server: SIP Redirect Server > > > 4. FS will send the call to the Termination IP WITHOUT changing the FROM > field > > INVITE sip:556230951662 at 162.x.x.x:5060 SIP/2.0 > Via: SIP/2.0/UDP 216.x.x.x;rport;branch=z9hG4bKB37Qj3rvamcjp > Max-Forwards: 68 > From: "551000" ;tag=gQepU2j7X9BKr > To: > Call-ID: e7159715-aac6-1235-79ba-002590a0ec9b > CSeq: 106579790 INVITE > Contact: > 2017-05-03 12:15:09.856127 [ERR] mod_xml_radius.c:914 Result of true > match: 162.x.x.x:5060 == ^69\.x\.x\.x > User-Agent: FreeSWITCH > 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: 397 > X-FS-Support: update_display,send_info > Remote-Party-ID: "551000" ;party= > calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 1493811013 1493811014 IN IP4 216.x.x.x > s=FreeSWITCH > c=IN IP4 216.x.x.x > t=0 0 > m=audio 20696 RTP/AVP 8 0 18 101 13 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > m=audio 20696 RTP/AVP 4 101 13 > a=rtpmap:4 G723/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:30 > > This is the dialplan I use to send calls to SIP redirect server > > > > > > > > > > > > > > > > > > > > > > > > /> > > > > > > /> > > > > > > > > > Is there any variable that will force FS to change the FROM field as > returned by the SIP Redirect server and send it to the termination > provider? Any help on this is really greatly appreciated! > > > Best regards, > > L. Popov > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Book a phone call (CST) Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170503/95f99678/attachment.html From gb at cm.nl Thu May 4 13:08:41 2017 From: gb at cm.nl (Grant Bagdasarian) Date: Thu, 4 May 2017 09:08:41 +0000 Subject: [Freeswitch-users] mod_distributor from db In-Reply-To: <4fa3d198c0034e9cbf17f88d14cf3cc6@cm.nl> References: <4fa3d198c0034e9cbf17f88d14cf3cc6@cm.nl> Message-ID: Does mod_xml_curl not request sip profiles? I see a request going out to the configured URL but it?s only for sofia.conf and some other modules like distributor, translate, lua, etc. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Grant Bagdasarian Sent: dinsdag 2 mei 2017 09:53 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] mod_distributor from db Nice! Thanks! From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: maandag 1 mei 2017 18:18 To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] mod_distributor from db mod_xml_curl could. On Mon, May 1, 2017 at 9:38 AM, Grant Bagdasarian > wrote: Hello, Is it possible to load the gateways of a sip profile from a database and the distributor config also from a database, using built in modules/functions? Regards, Grant _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170504/0dd77778/attachment-0001.html From deepikay at iiitd.ac.in Thu May 4 13:12:04 2017 From: deepikay at iiitd.ac.in (Deepika Yadav) Date: Thu, 4 May 2017 14:42:04 +0530 Subject: [Freeswitch-users] Freeswitch PRI support Message-ID: Hi, I am using Freeswitch in an application that initiates conference calls amongst 20 users in cellular network (mobile phones in GSM network). Currently, for the VOIP-GSM gateway, we are using the service from a company called Doorvaani. But, since, the cost of call estimates to be high and we cannot debug the call drops; we are thinking to buy our own PRI card. I am seeking recommendation on following points: 1. Which card should I buy that is most easily configurable with the Freeswitch i.e. company and type. 2. Reference on how should I start to make Freeswitch configure with the PRI card and start sending and receiving calls. For the current gateway service in use, I simply put the authentication credentials for the corresponding VOIP line offered by the company Doorvaani in the external SIP Profile. In case of PRI, what all changes are needed? Regards, Deepika -- Regards Deepika https://www.iiitd.edu.in/~deepikay/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170504/42ea9bc7/attachment.html From gmaruzz at gmail.com Thu May 4 13:24:13 2017 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Thu, 4 May 2017 11:24:13 +0200 Subject: [Freeswitch-users] Freeswitch PRI support In-Reply-To: References: Message-ID: For PRI you use Sangoma or Patton. But, why don't you use an hardware gateway sip<->gsm? It would save you very big money. Check on ebay and google, there are many of them, you put SIMs inside, and you are good to go. -giovanni sent from mobile cell: +39 347 266 56 18 Giovanni Maruzzelli OpenTelecom.IT Il 04 mag 2017 11:13, "Deepika Yadav" ha scritto: > Hi, > > I am using Freeswitch in an application that initiates conference calls > amongst 20 users in cellular network (mobile phones in GSM network). > Currently, for the VOIP-GSM gateway, we are using the service from a > company called Doorvaani. > > But, since, the cost of call estimates to be high and we cannot debug the > call drops; we are thinking to buy our own PRI card. > > I am seeking recommendation on following points: > > 1. Which card should I buy that is most easily configurable with the > Freeswitch i.e. company and type. > 2. Reference on how should I start to make Freeswitch configure with > the PRI card and start sending and receiving calls. For the current gateway > service in use, I simply put the authentication credentials for the > corresponding VOIP line offered by the company Doorvaani in the external > SIP Profile. In case of PRI, what all changes are needed? > > Regards, > Deepika > > -- > Regards > Deepika > https://www.iiitd.edu.in/~deepikay/ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170504/03f04797/attachment.html From deepikay at iiitd.ac.in Thu May 4 13:54:46 2017 From: deepikay at iiitd.ac.in (Deepika Yadav) Date: Thu, 4 May 2017 15:24:46 +0530 Subject: [Freeswitch-users] Freeswitch PRI support In-Reply-To: References: Message-ID: I am preferring PRI card because firstly, its cost is less than a VOIP-GSM gateway ( A VOIP-GSM gateway supporting 40 channels costs around $900 while a single PRI card that supports 30 channels cost around $500 ) and secondly, the voice quality is high. PRI line also does the automatic channel allocation when a particular channel is busy. Regards, Deepika On Thu, May 4, 2017 at 2:54 PM, Giovanni Maruzzelli wrote: > For PRI you use Sangoma or Patton. > > But, why don't you use an hardware gateway sip<->gsm? > > It would save you very big money. > > Check on ebay and google, there are many of them, you put SIMs inside, and > you are good to go. > > -giovanni > > > sent from mobile > cell: +39 347 266 56 18 > Giovanni Maruzzelli > OpenTelecom.IT > > Il 04 mag 2017 11:13, "Deepika Yadav" ha scritto: > >> Hi, >> >> I am using Freeswitch in an application that initiates conference calls >> amongst 20 users in cellular network (mobile phones in GSM network). >> Currently, for the VOIP-GSM gateway, we are using the service from a >> company called Doorvaani. >> >> But, since, the cost of call estimates to be high and we cannot debug the >> call drops; we are thinking to buy our own PRI card. >> >> I am seeking recommendation on following points: >> >> 1. Which card should I buy that is most easily configurable with the >> Freeswitch i.e. company and type. >> 2. Reference on how should I start to make Freeswitch configure with >> the PRI card and start sending and receiving calls. For the current gateway >> service in use, I simply put the authentication credentials for the >> corresponding VOIP line offered by the company Doorvaani in the external >> SIP Profile. In case of PRI, what all changes are needed? >> >> Regards, >> Deepika >> >> -- >> Regards >> Deepika >> https://www.iiitd.edu.in/~deepikay/ >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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://www.iiitd.edu.in/~deepikay/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170504/a68cc117/attachment.html From rbetancor at gmail.com Thu May 4 14:24:11 2017 From: rbetancor at gmail.com (=?UTF-8?Q?Ra=C3=BAl_Alexis_Betancor_Santana?=) Date: Thu, 4 May 2017 11:24:11 +0100 Subject: [Freeswitch-users] Freeswitch PRI support In-Reply-To: References: Message-ID: The best option would be to locate another VoIP provider with better service. For the PRI card, you will either need a PRI provider But eitherway ... if you want to use PSTN cards with FS, you should check the manuals as depending on the branch you buy, you should use a module or another. 2017-05-04 10:12 GMT+01:00 Deepika Yadav : > Hi, > > I am using Freeswitch in an application that initiates conference calls > amongst 20 users in cellular network (mobile phones in GSM network). > Currently, for the VOIP-GSM gateway, we are using the service from a > company called Doorvaani. > > But, since, the cost of call estimates to be high and we cannot debug the > call drops; we are thinking to buy our own PRI card. > > I am seeking recommendation on following points: > > 1. Which card should I buy that is most easily configurable with the > Freeswitch i.e. company and type. > 2. Reference on how should I start to make Freeswitch configure with > the PRI card and start sending and receiving calls. For the current gateway > service in use, I simply put the authentication credentials for the > corresponding VOIP line offered by the company Doorvaani in the external > SIP Profile. In case of PRI, what all changes are needed? > > Regards, > Deepika > > -- > Regards > Deepika > https://www.iiitd.edu.in/~deepikay/ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170504/4d9cb66f/attachment-0001.html From bipin at xbipin.com Thu May 4 15:33:51 2017 From: bipin at xbipin.com (Bipin Patel) Date: Thu, 4 May 2017 15:33:51 +0400 Subject: [Freeswitch-users] Freeswitch PRI support In-Reply-To: References: Message-ID: An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170504/6000a2bb/attachment.html From deepikay at iiitd.ac.in Thu May 4 15:51:49 2017 From: deepikay at iiitd.ac.in (Deepika Yadav) Date: Thu, 4 May 2017 17:21:49 +0530 Subject: [Freeswitch-users] Freeswitch PRI support In-Reply-To: References: Message-ID: Hi, The average outgoing call charges per minute in GSM network is 0.01 USD but if we use service from a VOIP-GSM service provider its thrice of it. Hence, considering to buy our own VOIP-GSM harware. I believe VOIP-GSM gateway hardware would be much convenient compared to PRI line. Regards, Deepika On Thu, May 4, 2017 at 5:03 PM, Bipin Patel wrote: > hi, > > as i know call termination rates for india are dead cheap if thats where > you are calling the gsm parties in > > > Regards, > Bipin > > > ------------------------------ > -------- Original Message -------- > Subject: Re: [Freeswitch-users] Freeswitch PRI support > From: Ra?l Alexis Betancor Santana > > To: FreeSWITCH Users Help > > Date: 5/4/2017, 2:24:11 PM > > The best option would be to locate another VoIP provider with better > service. > For the PRI card, you will either need a PRI provider > > But eitherway ... if you want to use PSTN cards with FS, you should check > the manuals as depending on the branch you buy, you should use a module or > another. > > 2017-05-04 10:12 GMT+01:00 Deepika Yadav : > >> Hi, >> >> I am using Freeswitch in an application that initiates conference calls >> amongst 20 users in cellular network (mobile phones in GSM network). >> Currently, for the VOIP-GSM gateway, we are using the service from a >> company called Doorvaani. >> >> But, since, the cost of call estimates to be high and we cannot debug the >> call drops; we are thinking to buy our own PRI card. >> >> I am seeking recommendation on following points: >> >> 1. Which card should I buy that is most easily configurable with the >> Freeswitch i.e. company and type. >> 2. Reference on how should I start to make Freeswitch configure with >> the PRI card and start sending and receiving calls. For the current gateway >> service in use, I simply put the authentication credentials for the >> corresponding VOIP line offered by the company Doorvaani in the external >> SIP Profile. In case of PRI, what all changes are needed? >> >> Regards, >> Deepika >> >> -- >> Regards >> Deepika >> https://www.iiitd.edu.in/~deepikay/ >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 > -- Regards Deepika https://www.iiitd.edu.in/~deepikay/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170504/b8f2f620/attachment.html From bipin at xbipin.com Thu May 4 15:57:21 2017 From: bipin at xbipin.com (Bipin Patel) Date: Thu, 4 May 2017 15:57:21 +0400 Subject: [Freeswitch-users] Freeswitch PRI support In-Reply-To: References: Message-ID: <75103a00-4790-ff27-9000-4e5787239781@xbipin.com> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170504/80d01b5d/attachment-0001.html From david.ponzone at gmail.com Thu May 4 10:37:27 2017 From: david.ponzone at gmail.com (David Ponzone) Date: Thu, 4 May 2017 08:37:27 +0200 Subject: [Freeswitch-users] Changing mode for CDRs In-Reply-To: <3bd101d2afc2$1bb15b50$531411f0$@freeswitch.org> References: <4CBD5239-0CA6-44F9-9200-DC982B289D43@magicmail.mooo.com> <57085CE8-0DC4-4D4B-8504-BE9F488A6B07@magicmail.mooo.com> <3bd101d2afc2$1bb15b50$531411f0$@freeswitch.org> Message-ID: <9EE63687-090F-486F-9896-EE5101CA3DA3@gmail.com> As i was facing the same issue, I just took 2 minutes to check that, and in mod_cdr_csv.c: if ((fd->fd = open(fd->path, O_WRONLY | O_CREAT | O_APPEND, S_IRUSR | S_IWUSR)) > -1) { > Le 7 avr. 2017 ? 19:12, Ken Rice a ?crit : > > You might want to specify which CDR module you are using? there could be different behavior between them? the umask used could be hard coded or could follow umask that FS was executed with > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org ] On Behalf Of Rick Jarvis > Sent: Friday, April 7, 2017 12:09 PM > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] Changing mode for CDRs > > Still can?t work out where this umask / permissions setting is coming from. Considering writing a hack that will monitor for files and chmod then move them, but I?d rather not add to the complexity. Someone out there must have a clue?! :) > >> On 7 Apr 2017, at 10:49, Rick Jarvis > wrote: >> >> I?ve tried setting the umask in .profile, .bashrc and even in freeswitch.service but nothing seems to affect it, even after a restart... >> >>> On 6 Apr 2017, at 23:02, Joel Serrano > wrote: >>> >>> Just a guess here... Have you tried setting the umask for the user launching the freeswitch process before starting it? >>> >>> On Thu, Apr 6, 2017 at 11:10 AM, Rick Jarvis > wrote: >>>> How do I set the permissions for the CDR files that FreeSWITCH writes? There don?t appear to be any settings that I can find apart from in config.FS0, although changing in this file doesn?t make any difference (and it doesn?t appear to be in typical FS format, I?m not actually sure what it does?!)? >>>> >>>> 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170504/6331e0c2/attachment-0001.html From er.kapildadheech at gmail.com Thu May 4 13:27:44 2017 From: er.kapildadheech at gmail.com (kapil dadheech) Date: Thu, 4 May 2017 14:57:44 +0530 Subject: [Freeswitch-users] Freeswitch PRI support In-Reply-To: References: Message-ID: Hello Deeplika, Please find my response below On Thu, May 4, 2017 at 2:42 PM, Deepika Yadav wrote: > Hi, > > I am using Freeswitch in an application that initiates conference calls > amongst 20 users in cellular network (mobile phones in GSM network). > Currently, for the VOIP-GSM gateway, we are using the service from a > company called Doorvaani. > > But, since, the cost of call estimates to be high and we cannot debug the > call drops; we are thinking to buy our own PRI card. > > I am seeking recommendation on following points: > > 1. Which card should I buy that is most easily configurable with the > Freeswitch i.e. company and type. > > - You can use sangoma pri card. We are using this from last 7years. > > 1. Reference on how should I start to make Freeswitch configure with > the PRI card and start sending and receiving calls. For the current gateway > service in use, I simply put the authentication credentials for the > corresponding VOIP line offered by the company Doorvaani in the external > SIP Profile. In case of PRI, what all changes are needed > > - If you are using sangoma card. Then you need to install wanpipe driver and configure wanpipe driver with freeswitch. Make sure to compile freeswitch with libpri. You can follow this https://wiki.freeswitch.org/wiki/FreeTDM#Installation > Regards, > Deepika > > -- > Regards > Deepika > https://www.iiitd.edu.in/~deepikay/ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Thanks & Regards Kapil Dadheech Phone - +918800098471 Skype - kapildadheech86 Contact on Facebook Follow on Twitter -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170504/f092099c/attachment-0001.html From snehsach at rediffmail.com Thu May 4 17:16:28 2017 From: snehsach at rediffmail.com (sachin ) Date: 4 May 2017 13:16:28 -0000 Subject: [Freeswitch-users] =?utf-8?q?Changing_audio_DTLS_state_from_HANDS?= =?utf-8?q?HAKE_to_FAIL_issue?= Message-ID: <1493903221.S.81594.autosave.drafts.1493903788.8725@webmail.rediffmail.com> Hello All,?I am trying to setup the Webrtc call between 2 SIPML5 client. I am using FreeSWITCH Version 1.6.14 on Windows. I am using internal LAN.The clients are getting registered over wss. I have created self signed certificates.  In var.xml I have set the codecs setting as follows  <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMU,PCMA">  <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA">?I am able to establish the call and there is 2 way voice when I call from Sipmpl5 to Linphone and it works. But when I call from Linphone to Firefox (Simpl5) then the call is not getting established and I am getting following error2017-05-04 11:29:04.543434 [ERR] switch_rtp.c:2907 audio Handshake failure 12017-05-04 11:29:04.543434 [INFO] switch_rtp.c:2908 Changing audio DTLS state from HANDSHAKE to FAIL2017-05-04 11:29:04.543434 [NOTICE] switch_rtp.c:2889 Hangup sofia/internal/sip:1001 at df7jal23ls0d.invalid [CS_EXCHANGE_MEDIA] [DESTINATION_OUT_OF_ORDER]My setup is as followsSIPML5 (FireFox browser 53.0)<-----> FS (v1.5)(Windows) <----> LinphoneI am attaching the logs for the reference.fs-logleve9.txt : Debug trace with loglevel =9?fs-sip-trace.txt : Sip tracePlease let me know what could the issue and pointers to resolve the same.?Thanks and RegardsSD -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170504/191d56a7/attachment-0001.html -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: fs-loglevl9.txt Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170504/191d56a7/attachment-0002.txt -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: fs-sip-trace.txt Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170504/191d56a7/attachment-0003.txt From shaun.stokes at itec-support.co.uk Thu May 4 18:31:36 2017 From: shaun.stokes at itec-support.co.uk (Shaun Stokes) Date: Thu, 4 May 2017 14:31:36 +0000 Subject: [Freeswitch-users] Freeswitch PRI support In-Reply-To: References: Message-ID: <6FD2F8B5BB72834E9939AEDF9FB802A901E865EAEF@mbx-01.sysconfig.co.uk> Highly recommend Sangoma for PRI or BRI, used both in the past, they?re straight forward to install and Sangoma officially support FreeSWITCH. We used the Sangoma A101E for PRI. FreeSWITCH installation guide and sample configuration here: https://wiki.freepbx.org/display/PC/Telephony+Cards+for+FreeSWITCH There are various models available depending on the number of ports and channels but you should always get the E version which includes the echo cancellation module (it?s tricky to buy the echo cancellation module separately): http://www.sangoma.com/products/digital-telephony-cards/ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Deepika Yadav Sent: 04 May 2017 10:12 To: FreeSWITCH Users Help Subject: [Freeswitch-users] Freeswitch PRI support Hi, I am using Freeswitch in an application that initiates conference calls amongst 20 users in cellular network (mobile phones in GSM network). Currently, for the VOIP-GSM gateway, we are using the service from a company called Doorvaani. But, since, the cost of call estimates to be high and we cannot debug the call drops; we are thinking to buy our own PRI card. I am seeking recommendation on following points: 1. Which card should I buy that is most easily configurable with the Freeswitch i.e. company and type. 2. Reference on how should I start to make Freeswitch configure with the PRI card and start sending and receiving calls. For the current gateway service in use, I simply put the authentication credentials for the corresponding VOIP line offered by the company Doorvaani in the external SIP Profile. In case of PRI, what all changes are needed? Regards, Deepika -- Regards Deepika https://www.iiitd.edu.in/~deepikay/ ______________________________________________________________________ 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170504/7f663a8c/attachment.html From deepikay at iiitd.ac.in Thu May 4 18:37:58 2017 From: deepikay at iiitd.ac.in (Deepika Yadav) Date: Thu, 4 May 2017 20:07:58 +0530 Subject: [Freeswitch-users] Freeswitch PRI support In-Reply-To: <6FD2F8B5BB72834E9939AEDF9FB802A901E865EAEF@mbx-01.sysconfig.co.uk> References: <6FD2F8B5BB72834E9939AEDF9FB802A901E865EAEF@mbx-01.sysconfig.co.uk> Message-ID: Thanks for the replies and would be happy to also know about the VOIP-GSM gateway model having at least 20 ports. Regards, Deepika On May 4, 2017 8:03 PM, "Shaun Stokes" wrote: > Highly recommend Sangoma for PRI or BRI, used both in the past, they?re > straight forward to install and Sangoma officially support FreeSWITCH. > > > > We used the Sangoma A101E for PRI. > > FreeSWITCH installation guide and sample configuration here: > https://wiki.freepbx.org/display/PC/Telephony+Cards+for+FreeSWITCH > > > > There are various models available depending on the number of ports and > channels but you should always get the E version which includes the echo > cancellation module (it?s tricky to buy the echo cancellation module > separately): > > http://www.sangoma.com/products/digital-telephony-cards/ > > > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Deepika > Yadav > *Sent:* 04 May 2017 10:12 > *To:* FreeSWITCH Users Help > *Subject:* [Freeswitch-users] Freeswitch PRI support > > > > Hi, > > > > I am using Freeswitch in an application that initiates conference calls > amongst 20 users in cellular network (mobile phones in GSM network). > Currently, for the VOIP-GSM gateway, we are using the service from a > company called Doorvaani. > > > > But, since, the cost of call estimates to be high and we cannot debug the > call drops; we are thinking to buy our own PRI card. > > > > I am seeking recommendation on following points: > > 1. Which card should I buy that is most easily configurable with the > Freeswitch i.e. company and type. > 2. Reference on how should I start to make Freeswitch configure with > the PRI card and start sending and receiving calls. For the current gateway > service in use, I simply put the authentication credentials for the > corresponding VOIP line offered by the company Doorvaani in the external > SIP Profile. In case of PRI, what all changes are needed? > > Regards, > > Deepika > > > > -- > > Regards > Deepika > https://www.iiitd.edu.in/~deepikay/ > > > ______________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170504/72193005/attachment-0001.html From joel at gogii.net Thu May 4 19:10:00 2017 From: joel at gogii.net (Joel Serrano) Date: Thu, 04 May 2017 15:10:00 +0000 Subject: [Freeswitch-users] Freeswitch PRI support In-Reply-To: References: <6FD2F8B5BB72834E9939AEDF9FB802A901E865EAEF@mbx-01.sysconfig.co.uk> Message-ID: We run into similar problem. Bought a GSM-VoIP gateway and worked perfectly but it was pricy. We found a Chinese brand that was very affordable. I'll send the name and model later today. Joel. On Thu, May 4, 2017 at 07:39 Deepika Yadav wrote: > Thanks for the replies and would be happy to also know about the VOIP-GSM > gateway model having at least 20 ports. > > Regards, > Deepika > > On May 4, 2017 8:03 PM, "Shaun Stokes" > wrote: > >> Highly recommend Sangoma for PRI or BRI, used both in the past, they?re >> straight forward to install and Sangoma officially support FreeSWITCH. >> >> >> >> We used the Sangoma A101E for PRI. >> >> FreeSWITCH installation guide and sample configuration here: >> https://wiki.freepbx.org/display/PC/Telephony+Cards+for+FreeSWITCH >> >> >> >> There are various models available depending on the number of ports and >> channels but you should always get the E version which includes the echo >> cancellation module (it?s tricky to buy the echo cancellation module >> separately): >> >> http://www.sangoma.com/products/digital-telephony-cards/ >> >> >> >> >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Deepika >> Yadav >> *Sent:* 04 May 2017 10:12 >> *To:* FreeSWITCH Users Help >> *Subject:* [Freeswitch-users] Freeswitch PRI support >> >> >> >> Hi, >> >> >> >> I am using Freeswitch in an application that initiates conference calls >> amongst 20 users in cellular network (mobile phones in GSM network). >> Currently, for the VOIP-GSM gateway, we are using the service from a >> company called Doorvaani. >> >> >> >> But, since, the cost of call estimates to be high and we cannot debug the >> call drops; we are thinking to buy our own PRI card. >> >> >> >> I am seeking recommendation on following points: >> >> 1. Which card should I buy that is most easily configurable with the >> Freeswitch i.e. company and type. >> 2. Reference on how should I start to make Freeswitch configure with >> the PRI card and start sending and receiving calls. For the current gateway >> service in use, I simply put the authentication credentials for the >> corresponding VOIP line offered by the company Doorvaani in the external >> SIP Profile. In case of PRI, what all changes are needed? >> >> Regards, >> >> Deepika >> >> >> >> -- >> >> Regards >> Deepika >> https://www.iiitd.edu.in/~deepikay/ >> >> >> ______________________________________________________________________ >> 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 >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170504/acd13382/attachment.html From italo at freeswitch.org Thu May 4 21:04:25 2017 From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=) Date: Thu, 04 May 2017 17:04:25 +0000 Subject: [Freeswitch-users] Changing audio DTLS state from HANDSHAKE to FAIL issue In-Reply-To: <1493903221.S.81594.autosave.drafts.1493903788.8725@webmail.rediffmail.com> References: <1493903221.S.81594.autosave.drafts.1493903788.8725@webmail.rediffmail.com> Message-ID: Old openssl version maybe? Em qui, 4 de mai de 2017 ?s 11:19, sachin escreveu: > Hello All, > > ?I am trying to setup the Webrtc call between 2 SIPML5 client. I am using > FreeSWITCH Version 1.6.14 on Windows. I am using internal LAN. > > The clients are getting registered over wss. I have created self signed > certificates. In var.xml I have set the codecs setting as follows > > > > > ?I am able to establish the call and there is 2 way voice when I call from > Sipmpl5 to Linphone and it works. But when I call from Linphone to Firefox > (Simpl5) then the call is not getting established and I am getting > following error > > 2017-05-04 11:29:04.543434 [ERR] switch_rtp.c:2907 audio Handshake failure > 1 > 2017-05-04 11:29:04.543434 [INFO] switch_rtp.c:2908 Changing audio DTLS > state from HANDSHAKE to FAIL > 2017-05-04 11:29:04.543434 [NOTICE] switch_rtp.c:2889 Hangup > sofia/internal/sip:1001 at df7jal23ls0d.invalid [CS_EXCHANGE_MEDIA] > [DESTINATION_OUT_OF_ORDER] > > My setup is as follows > > SIPML5 (FireFox browser 53.0)<-----> FS (v1.5)(Windows) <----> Linphone > > I am attaching the logs for the reference. > fs-logleve9.txt : Debug trace with loglevel =9 > ?fs-sip-trace.txt : Sip trace > Please let me know what could the issue and pointers to resolve the same. > > ?Thanks and Regards > SD > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170504/96d0d326/attachment.html From anthony.minessale at gmail.com Thu May 4 21:15:21 2017 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 4 May 2017 12:15:21 -0500 Subject: [Freeswitch-users] Changing audio DTLS state from HANDSHAKE to FAIL issue In-Reply-To: References: <1493903221.S.81594.autosave.drafts.1493903788.8725@webmail.rediffmail.com> Message-ID: FS 1.5 sounds like a bad plan. Try latest FS 1.6 or master. On Thu, May 4, 2017 at 12:04 PM, ?talo Rossi wrote: > Old openssl version maybe? > Em qui, 4 de mai de 2017 ?s 11:19, sachin > escreveu: > >> Hello All, >> >> ?I am trying to setup the Webrtc call between 2 SIPML5 client. I am using >> FreeSWITCH Version 1.6.14 on Windows. I am using internal LAN. >> >> The clients are getting registered over wss. I have created self signed >> certificates. In var.xml I have set the codecs setting as follows >> >> >> >> >> ?I am able to establish the call and there is 2 way voice when I call >> from Sipmpl5 to Linphone and it works. But when I call from Linphone to >> Firefox (Simpl5) then the call is not getting established and I am getting >> following error >> >> 2017-05-04 11:29:04.543434 [ERR] switch_rtp.c:2907 audio Handshake >> failure 1 >> 2017-05-04 11:29:04.543434 [INFO] switch_rtp.c:2908 Changing audio DTLS >> state from HANDSHAKE to FAIL >> 2017-05-04 11:29:04.543434 [NOTICE] switch_rtp.c:2889 Hangup >> sofia/internal/sip:1001 at df7jal23ls0d.invalid [CS_EXCHANGE_MEDIA] >> [DESTINATION_OUT_OF_ORDER] >> >> My setup is as follows >> >> SIPML5 (FireFox browser 53.0)<-----> FS (v1.5)(Windows) <----> Linphone >> >> I am attaching the logs for the reference. >> fs-logleve9.txt : Debug trace with loglevel =9 >> ?fs-sip-trace.txt : Sip trace >> Please let me know what could the issue and pointers to resolve the same. >> >> ?Thanks and Regards >> SD >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170504/084cab13/attachment-0001.html From koki.roul at gmail.com Thu May 4 19:20:27 2017 From: koki.roul at gmail.com (Lyubo Popov) Date: Thu, 4 May 2017 12:20:27 -0300 Subject: [Freeswitch-users] 300 Multiple Choices - how to force FS to replace From field In-Reply-To: References: Message-ID: Hello Brian, The authentication is DIGEST done via RADIUS. The Username is the same as the Caller ID...or maybe I understood your question wrong..? The number that shows in the from field is actually the SIP username created in the system ( for routing, billing, radius AAA, etc. ) and it is as well the callers number ( Caller ID). I use Raduis AAA to authenticate and account the calls and the user you see 551000 is actually a username of voip account created in the billing. That is why you see the incoming call from that username (551000). This is what FS is using in the FROM field. Since many of the accounts are created in format different from E164, often it is necessary to rewrite the account number ( the caller number ) to E164 or the termination will not accept the call. This is what I am trying to do now, set a rewrite rule in the billing system to convert 551000 to 1140031556 and this is what it is returned to FS as you can see in the packets the billing sends back.. Cheers, L.Popov On Wed, May 3, 2017 at 7:20 PM, Brian West wrote: > Why are you using the from field for authentication? > > On Wed, May 3, 2017 at 12:42 PM, Lyubo Popov wrote: > >> Hello everyone, >> >> I would like to express my thanks in advance to anyone who may be able to >> help me with some insides. >> >> I am using a routing software with SIP Redirect to send routes to FS with >> 300 Multiple Choices and mod_xml_radius to authenticate the SIP users. In >> the Sip redirect server I am manipulating as well the FROM number and >> sending back to FS, but FS will not respect this and continue using the SIP >> account that sent the call in the first place in the FROM field. Here are >> some SIP packets from both sides to clarify the whole process. >> >> 1. Sending call to FS with Zoiper, destination 556230951662 >> >> INVITE sip:556230951662 at 216.x.x.x:5080;transport=UDP SIP/2.0 >> Via: SIP/2.0/UDP 177.x.x.x:1048;branch=z9hG4bK- >> d8754z-038f1c7d251308c2-1---d8754z-;rport >> Max-Forwards: 70 >> Contact: >> To: >> From: "551000";tag=25599d20 >> Call-ID: NmUzYTAwNmQ1NTZjMDM2ZjVhYTgzMDdiY2RiMmI0ZTc. >> CSeq: 1 INVITE >> Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, >> INFO, SUBSCRIBE >> Content-Type: application/sdp >> Supported: replaces, norefersub, extended-refer, timer, >> X-cisco-serviceuri >> User-Agent: Zoiper for Windows 2.43 r24984 >> Allow-Events: presence, kpml >> Content-Length: 232 >> >> v=0 >> o=Zoiper_user 0 0 IN IP4 177.x.x.x >> s=Zoiper_session >> c=IN IP4 177.x.x.x >> t=0 0 >> m=audio 8000 RTP/AVP 8 0 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> a=sendrecv >> >> >> 2. FS sending INVITE to SIP Redirect server >> >> INVITE sip:556230951662 at 69.x.x.x:5060 SIP/2.0 >> Via: SIP/2.0/UDP 216.245.218.230;rport;branch=z9hG4bKateZg87rDBpZa >> Max-Forwards: 69 >> From: "551000" ;tag=FeNXS71300N0c >> To: >> Call-ID: e70ee145-aac6-1235-79ba-002590a0ec9b >> CSeq: 106579790 INVITE >> Contact: >> User-Agent: FreeSWITCH >> 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: 397 >> X-FS-Support: update_display,send_info >> Remote-Party-ID: "551000" ;party=c >> alling;screen=yes;privacy=off >> >> v=0 >> o=FreeSWITCH 1493809233 1493809234 IN IP4 216.x.x.x >> 2017-05-03 12:15:09.816119 [ERR] mod_xml_radius.c:911 Didn't match: >> 69.x.x.x:5060 == ^69\.x\.x\.x >> s=FreeSWITCH >> c=IN IP4 216.x.x.x >> t=0 0 >> m=audio 22476 RTP/AVP 8 0 18 101 13 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:18 G729/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=ptime:20 >> m=audio 22476 RTP/AVP 4 101 13 >> a=rtpmap:4 G723/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=ptime:30 >> >> >> 3. SIP Redirect returns 300 Multiple choices with the termination IP in >> Contact and with FROM field as instructed ( update 551000 with 1140031556) >> >> SIP/2.0 300 Multiple Choices >> Via: SIP/2.0/UDP 216.x.x.x;rport;branch=z9hG4bKateZg87rDBpZa >> From: "1140031556" ;tag=FeNXS71300N0c >> To: >> Contact: ;q=1.00 >> Call-ID: e70ee145-aac6-1235-79ba-002590a0ec9b >> CSeq: 106579790 INVITE >> Max-Forwards: 69 >> Content-Length: 0 >> Server: SIP Redirect Server >> >> >> 4. FS will send the call to the Termination IP WITHOUT changing the FROM >> field >> >> INVITE sip:556230951662 at 162.x.x.x:5060 SIP/2.0 >> Via: SIP/2.0/UDP 216.x.x.x;rport;branch=z9hG4bKB37Qj3rvamcjp >> Max-Forwards: 68 >> From: "551000" ;tag=gQepU2j7X9BKr >> To: >> Call-ID: e7159715-aac6-1235-79ba-002590a0ec9b >> CSeq: 106579790 INVITE >> Contact: >> 2017-05-03 12:15:09.856127 [ERR] mod_xml_radius.c:914 Result of true >> match: 162.x.x.x:5060 == ^69\.x\.x\.x >> User-Agent: FreeSWITCH >> 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: 397 >> X-FS-Support: update_display,send_info >> Remote-Party-ID: "551000" ;party=c >> alling;screen=yes;privacy=off >> >> v=0 >> o=FreeSWITCH 1493811013 1493811014 IN IP4 216.x.x.x >> s=FreeSWITCH >> c=IN IP4 216.x.x.x >> t=0 0 >> m=audio 20696 RTP/AVP 8 0 18 101 13 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:18 G729/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=ptime:20 >> m=audio 20696 RTP/AVP 4 101 13 >> a=rtpmap:4 G723/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=ptime:30 >> >> This is the dialplan I use to send calls to SIP redirect server >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> > /> >> >> >> >> >> >> > /> >> >> >> >> >> >> >> >> > >> Is there any variable that will force FS to change the FROM field as >> returned by the SIP Redirect server and send it to the termination >> provider? Any help on this is really greatly appreciated! >> >> >> Best regards, >> >> L. Popov >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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.org > > *Twitter: @FreeSWITCH , @briankwest* > > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > Book a phone call (CST) > > Allison prompts for FreeSWITCH: > > *https://www.gofundme.com/allison-prompts-for-freeswitch* > > > Got Bugs? Report them here ! | Reddit: > /r/freeswitch > > *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002> > | *M:*+1918424WEST (9378) > *Skype:*briankwest > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170504/0cc4fb54/attachment-0001.html From mike at jerris.com Thu May 4 21:41:55 2017 From: mike at jerris.com (Michael Jerris) Date: Thu, 4 May 2017 13:41:55 -0400 Subject: [Freeswitch-users] mod_distributor from db In-Reply-To: References: <4fa3d198c0034e9cbf17f88d14cf3cc6@cm.nl> Message-ID: <9C5AD1DC-A7F6-4E29-BE69-C0860A70E6F5@jerris.com> it requests the sip profiles. that is the sofia.conf request you see. > On May 4, 2017, at 5:08 AM, Grant Bagdasarian wrote: > > Does mod_xml_curl not request sip profiles? > I see a request going out to the configured URL but it?s only for sofia.conf and some other modules like distributor, translate, lua, etc. > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Grant Bagdasarian > Sent: dinsdag 2 mei 2017 09:53 > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] mod_distributor from db > > Nice! Thanks! > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org ] On Behalf Of Anthony Minessale > Sent: maandag 1 mei 2017 18:18 > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] mod_distributor from db > > mod_xml_curl could. > > > On Mon, May 1, 2017 at 9:38 AM, Grant Bagdasarian > wrote: > Hello, > > Is it possible to load the gateways of a sip profile from a database and the distributor config also from a database, using built in modules/functions? > > Regards, > > Grant > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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=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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170504/f8d47aad/attachment-0001.html From gregor at infomedia.si Thu May 4 23:31:19 2017 From: gregor at infomedia.si (Gregor Nanger) Date: Thu, 4 May 2017 21:31:19 +0200 Subject: [Freeswitch-users] Changing audio DTLS state from HANDSHAKE to FAIL issue In-Reply-To: References: <1493903221.S.81594.autosave.drafts.1493903788.8725@webmail.rediffmail.com> Message-ID: This error is familiar to me, I think so, if I remembered correctly. I am using windows. I had this error when client was Chrome. I just updated openssl dlls. I think that openssl library is updated on windows build now with version 1.6.17. And this version is also compiled on FS FTP to just download and install it. 2017-05-04 19:15 GMT+02:00 Anthony Minessale : > FS 1.5 sounds like a bad plan. > Try latest FS 1.6 or master. > > > On Thu, May 4, 2017 at 12:04 PM, ?talo Rossi wrote: > >> Old openssl version maybe? >> Em qui, 4 de mai de 2017 ?s 11:19, sachin >> escreveu: >> >>> Hello All, >>> >>> ?I am trying to setup the Webrtc call between 2 SIPML5 client. I am >>> using FreeSWITCH Version 1.6.14 on Windows. I am using internal LAN. >>> >>> The clients are getting registered over wss. I have created self signed >>> certificates. In var.xml I have set the codecs setting as follows >>> >>> >>> >>> >>> ?I am able to establish the call and there is 2 way voice when I call >>> from Sipmpl5 to Linphone and it works. But when I call from Linphone to >>> Firefox (Simpl5) then the call is not getting established and I am getting >>> following error >>> >>> 2017-05-04 11:29:04.543434 [ERR] switch_rtp.c:2907 audio Handshake >>> failure 1 >>> 2017-05-04 11:29:04.543434 [INFO] switch_rtp.c:2908 Changing audio DTLS >>> state from HANDSHAKE to FAIL >>> 2017-05-04 11:29:04.543434 [NOTICE] switch_rtp.c:2889 Hangup >>> sofia/internal/sip:1001 at df7jal23ls0d.invalid [CS_EXCHANGE_MEDIA] >>> [DESTINATION_OUT_OF_ORDER] >>> >>> My setup is as follows >>> >>> SIPML5 (FireFox browser 53.0)<-----> FS (v1.5)(Windows) <----> Linphone >>> >>> I am attaching the logs for the reference. >>> fs-logleve9.txt : Debug trace with loglevel =9 >>> ?fs-sip-trace.txt : Sip trace >>> Please let me know what could the issue and pointers to resolve the same. >>> >>> ?Thanks and Regards >>> SD >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 <+1%20919-386-9900> > > 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 > -- 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170504/f7c5a654/attachment.html From gb at cm.nl Fri May 5 12:07:34 2017 From: gb at cm.nl (Grant Bagdasarian) Date: Fri, 5 May 2017 08:07:34 +0000 Subject: [Freeswitch-users] mod_distributor from db In-Reply-To: <9C5AD1DC-A7F6-4E29-BE69-C0860A70E6F5@jerris.com> References: <4fa3d198c0034e9cbf17f88d14cf3cc6@cm.nl> <9C5AD1DC-A7F6-4E29-BE69-C0860A70E6F5@jerris.com> Message-ID: <5b8ea216bcf44a35ab36d7bb316d9e5a@cm.nl> Right! Thanks. What if I want to return multiple sip profiles? I see that it does 2 sofia.conf requests, does each request expect exactly one xml element? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris Sent: donderdag 4 mei 2017 19:42 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] mod_distributor from db it requests the sip profiles. that is the sofia.conf request you see. On May 4, 2017, at 5:08 AM, Grant Bagdasarian > wrote: Does mod_xml_curl not request sip profiles? I see a request going out to the configured URL but it?s only for sofia.conf and some other modules like distributor, translate, lua, etc. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Grant Bagdasarian Sent: dinsdag 2 mei 2017 09:53 To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] mod_distributor from db Nice! Thanks! From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: maandag 1 mei 2017 18:18 To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] mod_distributor from db mod_xml_curl could. On Mon, May 1, 2017 at 9:38 AM, Grant Bagdasarian > wrote: Hello, Is it possible to load the gateways of a sip profile from a database and the distributor config also from a database, using built in modules/functions? Regards, Grant _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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=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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170505/82cca6fa/attachment-0001.html From mylists at polite.se Fri May 5 13:38:46 2017 From: mylists at polite.se (Oivvio Polite) Date: Fri, 5 May 2017 11:38:46 +0200 Subject: [Freeswitch-users] Script language choice Message-ID: <20170505093846.GA2462@blomma.liberationtech.net> There's a page about script language choice on the wiki. There's a caveat at to top warning that the information is stale. https://freeswitch.org/confluence/display/FREESWITCH/Script+Language+Choice Anyway, it states that Lua is the preferred scripting language due efficacy. In a scenario where I'm not expecting very high loads are there other reasons for going with Lua anyway? I'm thinking that it's probably more battle tested if it's the most common choice among FS-developers. regards Oivvio From gmaruzz at gmail.com Fri May 5 14:31:36 2017 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Fri, 5 May 2017 12:31:36 +0200 Subject: [Freeswitch-users] Script language choice In-Reply-To: <20170505093846.GA2462@blomma.liberationtech.net> References: <20170505093846.GA2462@blomma.liberationtech.net> Message-ID: Ciao Oivvio, exactly as you said, Lua is the most popular choice, and so has the most distributed knowledge, discussions in mailing list, in IRC, in documentation, etc sent from mobile cell: +39 347 266 56 18 Giovanni Maruzzelli OpenTelecom.IT Il 05 mag 2017 11:40 AM, "Oivvio Polite" ha scritto: There's a page about script language choice on the wiki. There's a caveat at to top warning that the information is stale. https://freeswitch.org/confluence/display/FREESWITCH/Script+Language+Choice Anyway, it states that Lua is the preferred scripting language due efficacy. In a scenario where I'm not expecting very high loads are there other reasons for going with Lua anyway? I'm thinking that it's probably more battle tested if it's the most common choice among FS-developers. regards Oivvio _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170505/729cbadd/attachment.html From david.villasmil.work at gmail.com Fri May 5 15:08:01 2017 From: david.villasmil.work at gmail.com (David Villasmil) Date: Fri, 05 May 2017 11:08:01 +0000 Subject: [Freeswitch-users] Script language choice In-Reply-To: References: <20170505093846.GA2462@blomma.liberationtech.net> Message-ID: And it's very, very simple, efficient and fast! On Fri, May 5, 2017 at 12:32 PM Giovanni Maruzzelli wrote: > Ciao Oivvio, > > exactly as you said, Lua is the most popular choice, and so has the most > distributed knowledge, discussions in mailing list, in IRC, in > documentation, etc > > > > sent from mobile > cell: +39 347 266 56 18 > > Giovanni Maruzzelli > OpenTelecom.IT > > Il 05 mag 2017 11:40 AM, "Oivvio Polite" ha scritto: > > > There's a page about script language choice on the wiki. There's a > caveat at to top warning that the information is stale. > > https://freeswitch.org/confluence/display/FREESWITCH/Script+Language+Choice > > Anyway, it states that Lua is the preferred scripting language due > efficacy. > > In a scenario where I'm not expecting very high loads are there other > reasons for going with Lua anyway? I'm thinking that it's probably more > battle tested if it's the most common choice among FS-developers. > > regards Oivvio > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170505/67b13fd7/attachment.html From xxxman2008 at 126.com Fri May 5 15:15:16 2017 From: xxxman2008 at 126.com (Raymond) Date: Fri, 5 May 2017 19:15:16 +0800 (CST) Subject: [Freeswitch-users] Script language choice In-Reply-To: <20170505093846.GA2462@blomma.liberationtech.net> References: <20170505093846.GA2462@blomma.liberationtech.net> Message-ID: <79618397.ac53.15bd8523039.Coremail.xxxman2008@126.com> HI?Oivvio It's really a choice "up to u". For me, i'm using javascript . because i'm too lazy to study a "new" script . And many my script for FS ,taken about 20-40ms to run. That's enough for my business. So, my advise, just choose a familiar script. don't care if it's lua. But if you want to study a "new" one. Lua is the first choice. Raymond At 2017-05-05 17:38:46, "Oivvio Polite" wrote: > >There's a page about script language choice on the wiki. There's a >caveat at to top warning that the information is stale. > >https://freeswitch.org/confluence/display/FREESWITCH/Script+Language+Choice > >Anyway, it states that Lua is the preferred scripting language due efficacy. > >In a scenario where I'm not expecting very high loads are there other >reasons for going with Lua anyway? I'm thinking that it's probably more >battle tested if it's the most common choice among FS-developers. > >regards Oivvio > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://confluence.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170505/c2f3ed9f/attachment.html From mylists at polite.se Fri May 5 15:24:21 2017 From: mylists at polite.se (Oivvio Polite) Date: Fri, 5 May 2017 13:24:21 +0200 Subject: [Freeswitch-users] Script language choice In-Reply-To: References: <20170505093846.GA2462@blomma.liberationtech.net> Message-ID: <20170505112421.GB2462@blomma.liberationtech.net> On fre, maj 05, 2017 at 12:31:36 +0200, Giovanni Maruzzelli wrote: > Ciao Oivvio, > > exactly as you said, Lua is the most popular choice, and so has the most > distributed knowledge, discussions in mailing list, in IRC, in > documentation, etc > I kind of expected that. When jumping in to new technologies I always try to figure out what usage patterns are most well used/tested/documented rather than just picking whatever fits best with my old stack/experience. Tell me, does this preference for Lua hold when talking to FS over ESL as well? Or are FS-oldtimers more language agnostic in that scenario? Oivvio From xxxman2008 at 126.com Fri May 5 15:28:49 2017 From: xxxman2008 at 126.com (Raymond) Date: Fri, 5 May 2017 19:28:49 +0800 (CST) Subject: [Freeswitch-users] Freeswitch PRI support In-Reply-To: References: Message-ID: <27a9d063.ad3b.15bd85e9762.Coremail.xxxman2008@126.com> HI, Regarding this. i have asked some question of module "GSMOpen". In fact ,i want to make Freeswitch box as "GSM VOIP Gateway" by using "gsmopen". But...seems no one help me . :-(( . If it worked , can save more money than hardware gsm voip gateway ( GOIP ). Raymond At 2017-05-04 17:24:13, "Giovanni Maruzzelli" wrote: For PRI you use Sangoma or Patton. But, why don't you use an hardware gateway sip<->gsm? It would save you very big money. Check on ebay and google, there are many of them, you put SIMs inside, and you are good to go. -giovanni sent from mobile cell: +39 347 266 56 18 Giovanni Maruzzelli OpenTelecom.IT Il 04 mag 2017 11:13, "Deepika Yadav" ha scritto: Hi, I am using Freeswitch in an application that initiates conference calls amongst 20 users in cellular network (mobile phones in GSM network). Currently, for the VOIP-GSM gateway, we are using the service from a company called Doorvaani. But, since, the cost of call estimates to be high and we cannot debug the call drops; we are thinking to buy our own PRI card. I am seeking recommendation on following points: Which card should I buy that is most easily configurable with the Freeswitch i.e. company and type. Reference on how should I start to make Freeswitch configure with the PRI card and start sending and receiving calls. For the current gateway service in use, I simply put the authentication credentials for the corresponding VOIP line offered by the company Doorvaani in the external SIP Profile. In case of PRI, what all changes are needed? Regards, Deepika -- Regards Deepika https://www.iiitd.edu.in/~deepikay/ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170505/f3df0f49/attachment-0001.html From david.villasmil.work at gmail.com Fri May 5 15:28:34 2017 From: david.villasmil.work at gmail.com (David Villasmil) Date: Fri, 05 May 2017 11:28:34 +0000 Subject: [Freeswitch-users] Script language choice In-Reply-To: <20170505112421.GB2462@blomma.liberationtech.net> References: <20170505093846.GA2462@blomma.liberationtech.net> <20170505112421.GB2462@blomma.liberationtech.net> Message-ID: For ESL i use perl, can't beat it, there's a module for fs-esl you can compile... But with it really doesn't matter the choice of language you use, as long as you're comfortable with and it's fast. On Fri, May 5, 2017 at 1:25 PM Oivvio Polite wrote: > On fre, maj 05, 2017 at 12:31:36 +0200, Giovanni Maruzzelli wrote: > > Ciao Oivvio, > > > > exactly as you said, Lua is the most popular choice, and so has the most > > distributed knowledge, discussions in mailing list, in IRC, in > > documentation, etc > > > > I kind of expected that. When jumping in to new technologies I always > try to figure out what usage patterns are most well > used/tested/documented rather than just picking whatever fits best with > my old stack/experience. > > Tell me, does this preference for Lua hold when talking to > FS over ESL as well? Or are FS-oldtimers more language agnostic in that > scenario? > > Oivvio > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170505/ad761d90/attachment.html From mylists at polite.se Fri May 5 15:38:24 2017 From: mylists at polite.se (Oivvio Polite) Date: Fri, 5 May 2017 13:38:24 +0200 Subject: [Freeswitch-users] Looking to hire a consultant In-Reply-To: References: Message-ID: <20170505113824.GC2462@blomma.liberationtech.net> On tis, maj 02, 2017 at 04:23:58 +0000, Sean Devoy wrote: > Can?t afford them. I emailed them at least once before. Perhaps I misunderstood, but I came away thinking anything less than $1000 wasn?t worth their time or at least worth the overhead they charge for ?planning?. This is more like a $400 to $500 job max, I can?t spend $300 ?planning.? > > They wanted hours (2 min I think) of conf calls and planning at a pretty high rate just to talk about what I wanted to do. There would be more money spent ?planning the project? than just doing the work. I am all for specifications and project planning for something that will take more than 4 or 5 hours or is in some way vague, but that is not required on simple jobs. They are just priced out of my reach, I wish they were not but I don?t have hundreds of users to pass this on to. I would rather send the FS team the money than anyone else, but not at that premium. > > The project is pretty simple: > > * Install a new FS Stable build on a new Debian 8.7 server > * Move/Convert my current XML config over (about 30 extensions) > * Help config Cisco SPA phones for SCA. > > That should not cost $250+ to plan. Maybe you should try your luck with one of the hire a freelancer sites? https://www.upwork.com/o/profiles/browse/?q=freeswitch&rate=0-10 Oivvio From brian at freeswitch.org Fri May 5 17:23:28 2017 From: brian at freeswitch.org (Brian West) Date: Fri, 5 May 2017 08:23:28 -0500 Subject: [Freeswitch-users] Looking to hire a consultant In-Reply-To: References: Message-ID: Sean, You emailed back in 2015, I have no recent emails from you on this subject on consulting at freeswitch.org. I've actually been working on a more affordable support package for smaller businesses. I would like your feedback. /b On Mon, May 1, 2017 at 11:23 PM, Sean Devoy wrote: > Can?t afford them. I emailed them at least once before. Perhaps I > misunderstood, but I came away thinking anything less than $1000 wasn?t > worth their time or at least worth the overhead they charge for > ?planning?. This is more like a $400 to $500 job max, I can?t spend $300 > ?planning.? > > > > They wanted hours (2 min I think) of conf calls and planning at a pretty > high rate just to talk about what I wanted to do. There would be more money > spent ?planning the project? than just doing the work. I am all for > specifications and project planning for something that will take more than > 4 or 5 hours or is in some way vague, but that is not required on simple > jobs. They are just priced out of my reach, I wish they were not but I > don?t have hundreds of users to pass this on to. I would rather send the > FS team the money than anyone else, but not at that premium. > > > > The project is pretty simple: > > - Install a new FS Stable build on a new Debian 8.7 server > - Move/Convert my current XML config over (about 30 extensions) > - Help config Cisco SPA phones for SCA. > > > > That should not cost $250+ to plan. > > > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Giovanni > Maruzzelli > *Sent:* Monday, May 1, 2017 11:15 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Looking to hire a consultant > > > > Please email consulting at freeswitch.org > > -giovanni > > > > On 2 May 2017 at 04:39, Sean Devoy wrote: > > Hi, > > > > We want to hire someone for a few hours to get a configuration working > with FS and Cisco SPA504G phones. > > > > We have all the basics working great for FS on SPA phones. We need a > single DID number to ring on 3 line buttons on 2 different phones. If a > call is placed on hold on phone 1, it must be picked up on phone 2. The > customer claims the staff is too stupid to use call parking. > > > > I think this really needs to be someone who knows both FS and Cisco SPA > phones. There are just too many possible parameter combos for us to work it > out. > > > > Contact me directly if you can handle the task and have availability soon. > > > > Thanks, > > Sean > > SDevoy(-at-)bizfocused.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 > > > > > -- > > > Sincerely, > > Giovanni Maruzzelli > OpenTelecom.IT > cell: +39 347 266 56 18 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Book a phone call (CST) Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170505/09234f9d/attachment-0001.html From brian at freeswitch.org Fri May 5 17:26:46 2017 From: brian at freeswitch.org (Brian West) Date: Fri, 5 May 2017 08:26:46 -0500 Subject: [Freeswitch-users] Changing audio DTLS state from HANDSHAKE to FAIL issue In-Reply-To: References: <1493903221.S.81594.autosave.drafts.1493903788.8725@webmail.rediffmail.com> Message-ID: OpenSSL on the windows build needs to be updated. https://freeswitch.org/jira/browse/FS-9510 On Thu, May 4, 2017 at 2:31 PM, Gregor Nanger wrote: > This error is familiar to me, I think so, if I remembered correctly. > > I am using windows. I had this error when client was Chrome. I just > updated openssl dlls. I think that openssl library is updated on windows > build now with version 1.6.17. And this version is also compiled on FS FTP > to just download and install it. > > 2017-05-04 19:15 GMT+02:00 Anthony Minessale > : > >> FS 1.5 sounds like a bad plan. >> Try latest FS 1.6 or master. >> >> >> On Thu, May 4, 2017 at 12:04 PM, ?talo Rossi >> wrote: >> >>> Old openssl version maybe? >>> Em qui, 4 de mai de 2017 ?s 11:19, sachin >>> escreveu: >>> >>>> Hello All, >>>> >>>> ?I am trying to setup the Webrtc call between 2 SIPML5 client. I am >>>> using FreeSWITCH Version 1.6.14 on Windows. I am using internal LAN. >>>> >>>> The clients are getting registered over wss. I have created self signed >>>> certificates. In var.xml I have set the codecs setting as follows >>>> >>>> >>>> >>>> >>>> ?I am able to establish the call and there is 2 way voice when I call >>>> from Sipmpl5 to Linphone and it works. But when I call from Linphone to >>>> Firefox (Simpl5) then the call is not getting established and I am getting >>>> following error >>>> >>>> 2017-05-04 11:29:04.543434 [ERR] switch_rtp.c:2907 audio Handshake >>>> failure 1 >>>> 2017-05-04 11:29:04.543434 [INFO] switch_rtp.c:2908 Changing audio DTLS >>>> state from HANDSHAKE to FAIL >>>> 2017-05-04 11:29:04.543434 [NOTICE] switch_rtp.c:2889 Hangup >>>> sofia/internal/sip:1001 at df7jal23ls0d.invalid [CS_EXCHANGE_MEDIA] >>>> [DESTINATION_OUT_OF_ORDER] >>>> >>>> My setup is as follows >>>> >>>> SIPML5 (FireFox browser 53.0)<-----> FS (v1.5)(Windows) <----> Linphone >>>> >>>> I am attaching the logs for the reference. >>>> fs-logleve9.txt : Debug trace with loglevel =9 >>>> ?fs-sip-trace.txt : Sip trace >>>> Please let me know what could the issue and pointers to resolve the >>>> same. >>>> >>>> ?Thanks and Regards >>>> SD >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-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 >>> >> >> >> >> -- >> 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 <+1%20919-386-9900> >> >> 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 >> > > > > -- > 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 > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Book a phone call (CST) Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170505/b7225a5e/attachment.html From brian at freeswitch.org Fri May 5 17:31:20 2017 From: brian at freeswitch.org (Brian West) Date: Fri, 5 May 2017 08:31:20 -0500 Subject: [Freeswitch-users] 300 Multiple Choices - how to force FS to replace From field In-Reply-To: References: Message-ID: are you setting manual-redirect in your Sofia profile? https://freeswitch.org/confluence/display/FREESWITCH/Handling+SIP+Redirect On Thu, May 4, 2017 at 10:20 AM, Lyubo Popov wrote: > Hello Brian, > > The authentication is DIGEST done via RADIUS. The Username is the same as > the Caller ID...or maybe I understood your question wrong..? The number > that shows in the from field is actually the SIP username created in the > system ( for routing, billing, radius AAA, etc. ) and it is as well the > callers number ( Caller ID). I use Raduis AAA to authenticate and account > the calls and the user you see 551000 is actually a username of voip > account created in the billing. That is why you see the incoming call from > that username (551000). This is what FS is using in the FROM field. Since > many of the accounts are created in format different from E164, often it is > necessary to rewrite the account number ( the caller number ) to E164 or > the termination will not accept the call. This is what I am trying to do > now, set a rewrite rule in the billing system to convert 551000 to > 1140031556 and this is what it is returned to FS as you can see in the > packets the billing sends back.. > > Cheers, > L.Popov > > > On Wed, May 3, 2017 at 7:20 PM, Brian West wrote: > >> Why are you using the from field for authentication? >> >> On Wed, May 3, 2017 at 12:42 PM, Lyubo Popov wrote: >> >>> Hello everyone, >>> >>> I would like to express my thanks in advance to anyone who may be able >>> to help me with some insides. >>> >>> I am using a routing software with SIP Redirect to send routes to FS >>> with 300 Multiple Choices and mod_xml_radius to authenticate the SIP users. >>> In the Sip redirect server I am manipulating as well the FROM number and >>> sending back to FS, but FS will not respect this and continue using the SIP >>> account that sent the call in the first place in the FROM field. Here are >>> some SIP packets from both sides to clarify the whole process. >>> >>> 1. Sending call to FS with Zoiper, destination 556230951662 >>> >>> INVITE sip:556230951662 at 216.x.x.x:5080;transport=UDP SIP/2.0 >>> Via: SIP/2.0/UDP 177.x.x.x:1048;branch=z9hG4bK- >>> d8754z-038f1c7d251308c2-1---d8754z-;rport >>> Max-Forwards: 70 >>> Contact: >>> To: >>> From: "551000";tag=25599d20 >>> Call-ID: NmUzYTAwNmQ1NTZjMDM2ZjVhYTgzMDdiY2RiMmI0ZTc. >>> CSeq: 1 INVITE >>> Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, >>> INFO, SUBSCRIBE >>> Content-Type: application/sdp >>> Supported: replaces, norefersub, extended-refer, timer, >>> X-cisco-serviceuri >>> User-Agent: Zoiper for Windows 2.43 r24984 >>> Allow-Events: presence, kpml >>> Content-Length: 232 >>> >>> v=0 >>> o=Zoiper_user 0 0 IN IP4 177.x.x.x >>> s=Zoiper_session >>> c=IN IP4 177.x.x.x >>> t=0 0 >>> m=audio 8000 RTP/AVP 8 0 101 >>> a=rtpmap:8 PCMA/8000 >>> a=rtpmap:0 PCMU/8000 >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-15 >>> a=sendrecv >>> >>> >>> 2. FS sending INVITE to SIP Redirect server >>> >>> INVITE sip:556230951662 at 69.x.x.x:5060 SIP/2.0 >>> Via: SIP/2.0/UDP 216.245.218.230;rport;branch=z9hG4bKateZg87rDBpZa >>> Max-Forwards: 69 >>> From: "551000" ;tag=FeNXS71300N0c >>> To: >>> Call-ID: e70ee145-aac6-1235-79ba-002590a0ec9b >>> CSeq: 106579790 INVITE >>> Contact: >>> User-Agent: FreeSWITCH >>> 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: 397 >>> X-FS-Support: update_display,send_info >>> Remote-Party-ID: "551000" ;party=c >>> alling;screen=yes;privacy=off >>> >>> v=0 >>> o=FreeSWITCH 1493809233 1493809234 IN IP4 216.x.x.x >>> 2017-05-03 12:15:09.816119 [ERR] mod_xml_radius.c:911 Didn't match: >>> 69.x.x.x:5060 == ^69\.x\.x\.x >>> s=FreeSWITCH >>> c=IN IP4 216.x.x.x >>> t=0 0 >>> m=audio 22476 RTP/AVP 8 0 18 101 13 >>> a=rtpmap:8 PCMA/8000 >>> a=rtpmap:0 PCMU/8000 >>> a=rtpmap:18 G729/8000 >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-16 >>> a=ptime:20 >>> m=audio 22476 RTP/AVP 4 101 13 >>> a=rtpmap:4 G723/8000 >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-16 >>> a=ptime:30 >>> >>> >>> 3. SIP Redirect returns 300 Multiple choices with the termination IP in >>> Contact and with FROM field as instructed ( update 551000 with 1140031556) >>> >>> SIP/2.0 300 Multiple Choices >>> Via: SIP/2.0/UDP 216.x.x.x;rport;branch=z9hG4bKateZg87rDBpZa >>> From: "1140031556" ;tag=FeNXS71300N0c >>> To: >>> Contact: ;q=1.00 >>> Call-ID: e70ee145-aac6-1235-79ba-002590a0ec9b >>> CSeq: 106579790 INVITE >>> Max-Forwards: 69 >>> Content-Length: 0 >>> Server: SIP Redirect Server >>> >>> >>> 4. FS will send the call to the Termination IP WITHOUT changing the FROM >>> field >>> >>> INVITE sip:556230951662 at 162.x.x.x:5060 SIP/2.0 >>> Via: SIP/2.0/UDP 216.x.x.x;rport;branch=z9hG4bKB37Qj3rvamcjp >>> Max-Forwards: 68 >>> From: "551000" ;tag=gQepU2j7X9BKr >>> To: >>> Call-ID: e7159715-aac6-1235-79ba-002590a0ec9b >>> CSeq: 106579790 INVITE >>> Contact: >>> 2017-05-03 12:15:09.856127 [ERR] mod_xml_radius.c:914 Result of true >>> match: 162.x.x.x:5060 == ^69\.x\.x\.x >>> User-Agent: FreeSWITCH >>> 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: 397 >>> X-FS-Support: update_display,send_info >>> Remote-Party-ID: "551000" ;party=c >>> alling;screen=yes;privacy=off >>> >>> v=0 >>> o=FreeSWITCH 1493811013 1493811014 IN IP4 216.x.x.x >>> s=FreeSWITCH >>> c=IN IP4 216.x.x.x >>> t=0 0 >>> m=audio 20696 RTP/AVP 8 0 18 101 13 >>> a=rtpmap:8 PCMA/8000 >>> a=rtpmap:0 PCMU/8000 >>> a=rtpmap:18 G729/8000 >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-16 >>> a=ptime:20 >>> m=audio 20696 RTP/AVP 4 101 13 >>> a=rtpmap:4 G723/8000 >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-16 >>> a=ptime:30 >>> >>> This is the dialplan I use to send calls to SIP redirect server >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >> /> >>> >>> >>> >>> >>> >>> >> /> >>> >>> >>> >>> >>> >>> >>> >>> >> >>> Is there any variable that will force FS to change the FROM field as >>> returned by the SIP Redirect server and send it to the termination >>> provider? Any help on this is really greatly appreciated! >>> >>> >>> Best regards, >>> >>> L. Popov >>> >>> >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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.org >> >> *Twitter: @FreeSWITCH , @briankwest* >> >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> Book a phone call (CST) >> >> Allison prompts for FreeSWITCH: >> >> *https://www.gofundme.com/allison-prompts-for-freeswitch* >> >> >> Got Bugs? Report them here ! | Reddit: >> /r/freeswitch >> >> *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002> >> | *M:*+1918424WEST (9378) >> *Skype:*briankwest >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Book a phone call (CST) Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170505/9f7c1b1b/attachment-0001.html From brian at freeswitch.org Fri May 5 17:33:39 2017 From: brian at freeswitch.org (Brian West) Date: Fri, 5 May 2017 08:33:39 -0500 Subject: [Freeswitch-users] OPUS Codec In-Reply-To: <20170503071638.GA14349@mail.marktcontact.com> References: <20170503071638.GA14349@mail.marktcontact.com> Message-ID: None of the hardware phones do FEC/PLC last I tested, Sonus has OPUS support, but again no FEC/PLC support. /b On Wed, May 3, 2017 at 2:16 AM, EL wrote: > > Yealink is supporting OPUS on several other models since firmware > V81: > > Quote: > "We will support opus on the standard V81 of > SIP-T40P/T23P/T23G/T2 1(P) E2/T19(P) E2." > > Source: http://forum.yealink.com/forum/showthread.php?tid= > 29650&pid=51262&mode=threaded > > I can confirm OPUS implementation on the 'T21P E2' model. > > -- > EL > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Book a phone call (CST) Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170505/5051b2fa/attachment.html From gregor at infomedia.si Fri May 5 17:47:12 2017 From: gregor at infomedia.si (Gregor Nanger) Date: Fri, 5 May 2017 15:47:12 +0200 Subject: [Freeswitch-users] Changing audio DTLS state from HANDSHAKE to FAIL issue In-Reply-To: References: <1493903221.S.81594.autosave.drafts.1493903788.8725@webmail.rediffmail.com> Message-ID: Brian, isn't this solved in 1.6.16? - FS-10037 [core] Update OpenSSL to version 1.0.2k for Windows 2017-05-05 15:26 GMT+02:00 Brian West : > OpenSSL on the windows build needs to be updated. > > https://freeswitch.org/jira/browse/FS-9510 > > On Thu, May 4, 2017 at 2:31 PM, Gregor Nanger wrote: > >> This error is familiar to me, I think so, if I remembered correctly. >> >> I am using windows. I had this error when client was Chrome. I just >> updated openssl dlls. I think that openssl library is updated on windows >> build now with version 1.6.17. And this version is also compiled on FS FTP >> to just download and install it. >> >> 2017-05-04 19:15 GMT+02:00 Anthony Minessale > >: >> >>> FS 1.5 sounds like a bad plan. >>> Try latest FS 1.6 or master. >>> >>> >>> On Thu, May 4, 2017 at 12:04 PM, ?talo Rossi >>> wrote: >>> >>>> Old openssl version maybe? >>>> Em qui, 4 de mai de 2017 ?s 11:19, sachin >>>> escreveu: >>>> >>>>> Hello All, >>>>> >>>>> ?I am trying to setup the Webrtc call between 2 SIPML5 client. I am >>>>> using FreeSWITCH Version 1.6.14 on Windows. I am using internal LAN. >>>>> >>>>> The clients are getting registered over wss. I have created self >>>>> signed certificates. In var.xml I have set the codecs setting as follows >>>>> >>>>> >>>>> >>>>> >>>>> ?I am able to establish the call and there is 2 way voice when I call >>>>> from Sipmpl5 to Linphone and it works. But when I call from Linphone to >>>>> Firefox (Simpl5) then the call is not getting established and I am getting >>>>> following error >>>>> >>>>> 2017-05-04 11:29:04.543434 [ERR] switch_rtp.c:2907 audio Handshake >>>>> failure 1 >>>>> 2017-05-04 11:29:04.543434 [INFO] switch_rtp.c:2908 Changing audio >>>>> DTLS state from HANDSHAKE to FAIL >>>>> 2017-05-04 11:29:04.543434 [NOTICE] switch_rtp.c:2889 Hangup >>>>> sofia/internal/sip:1001 at df7jal23ls0d.invalid [CS_EXCHANGE_MEDIA] >>>>> [DESTINATION_OUT_OF_ORDER] >>>>> >>>>> My setup is as follows >>>>> >>>>> SIPML5 (FireFox browser 53.0)<-----> FS (v1.5)(Windows) <----> Linphone >>>>> >>>>> I am attaching the logs for the reference. >>>>> fs-logleve9.txt : Debug trace with loglevel =9 >>>>> ?fs-sip-trace.txt : Sip trace >>>>> Please let me know what could the issue and pointers to resolve the >>>>> same. >>>>> >>>>> ?Thanks and Regards >>>>> SD >>>>> ____________________________________________________________ >>>>> _____________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-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 >>>> >>> >>> >>> >>> -- >>> 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 <+1%20919-386-9900> >>> >>> 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 >>> >> >> >> >> -- >> 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 >> > > > > -- > > *Brian West* > brian at freeswitch.org > > *Twitter: @FreeSWITCH , @briankwest* > > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > Book a phone call (CST) > > > Allison prompts for FreeSWITCH: > > *https://www.gofundme.com/allison-prompts-for-freeswitch* > > > Got Bugs? Report them here > ! > | Reddit: /r/freeswitch > > > *T:*+19184209001 <+1%20918-420-9001> | *F:*+19184209002 > <+1%20918-420-9002> | *M:*+1918424WEST (9378) > *Skype:*briankwest > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170505/fbd78121/attachment-0001.html From brian at freeswitch.org Fri May 5 17:53:55 2017 From: brian at freeswitch.org (Brian West) Date: Fri, 5 May 2017 08:53:55 -0500 Subject: [Freeswitch-users] Changing audio DTLS state from HANDSHAKE to FAIL issue In-Reply-To: References: <1493903221.S.81594.autosave.drafts.1493903788.8725@webmail.rediffmail.com> Message-ID: Clearly that isn't the case, I suspect a define is missing to actually enable ECDSA at this point. On Fri, May 5, 2017 at 8:47 AM, Gregor Nanger wrote: > Brian, isn't this solved in 1.6.16? > > > - FS-10037 > [core] > Update OpenSSL to version 1.0.2k for Windows > > > 2017-05-05 15:26 GMT+02:00 Brian West : > >> OpenSSL on the windows build needs to be updated. >> >> https://freeswitch.org/jira/browse/FS-9510 >> >> On Thu, May 4, 2017 at 2:31 PM, Gregor Nanger >> wrote: >> >>> This error is familiar to me, I think so, if I remembered correctly. >>> >>> I am using windows. I had this error when client was Chrome. I just >>> updated openssl dlls. I think that openssl library is updated on windows >>> build now with version 1.6.17. And this version is also compiled on FS FTP >>> to just download and install it. >>> >>> 2017-05-04 19:15 GMT+02:00 Anthony Minessale < >>> anthony.minessale at gmail.com>: >>> >>>> FS 1.5 sounds like a bad plan. >>>> Try latest FS 1.6 or master. >>>> >>>> >>>> On Thu, May 4, 2017 at 12:04 PM, ?talo Rossi >>>> wrote: >>>> >>>>> Old openssl version maybe? >>>>> Em qui, 4 de mai de 2017 ?s 11:19, sachin >>>>> escreveu: >>>>> >>>>>> Hello All, >>>>>> >>>>>> ?I am trying to setup the Webrtc call between 2 SIPML5 client. I am >>>>>> using FreeSWITCH Version 1.6.14 on Windows. I am using internal LAN. >>>>>> >>>>>> The clients are getting registered over wss. I have created self >>>>>> signed certificates. In var.xml I have set the codecs setting as follows >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> ?I am able to establish the call and there is 2 way voice when I call >>>>>> from Sipmpl5 to Linphone and it works. But when I call from Linphone to >>>>>> Firefox (Simpl5) then the call is not getting established and I am getting >>>>>> following error >>>>>> >>>>>> 2017-05-04 11:29:04.543434 [ERR] switch_rtp.c:2907 audio Handshake >>>>>> failure 1 >>>>>> 2017-05-04 11:29:04.543434 [INFO] switch_rtp.c:2908 Changing audio >>>>>> DTLS state from HANDSHAKE to FAIL >>>>>> 2017-05-04 11:29:04.543434 [NOTICE] switch_rtp.c:2889 Hangup >>>>>> sofia/internal/sip:1001 at df7jal23ls0d.invalid [CS_EXCHANGE_MEDIA] >>>>>> [DESTINATION_OUT_OF_ORDER] >>>>>> >>>>>> My setup is as follows >>>>>> >>>>>> SIPML5 (FireFox browser 53.0)<-----> FS (v1.5)(Windows) <----> >>>>>> Linphone >>>>>> >>>>>> I am attaching the logs for the reference. >>>>>> fs-logleve9.txt : Debug trace with loglevel =9 >>>>>> ?fs-sip-trace.txt : Sip trace >>>>>> Please let me know what could the issue and pointers to resolve the >>>>>> same. >>>>>> >>>>>> ?Thanks and Regards >>>>>> SD >>>>>> ____________________________________________________________ >>>>>> _____________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-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 >>>>> >>>> >>>> >>>> >>>> -- >>>> 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 <+1%20919-386-9900> >>>> >>>> >>>> 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/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 >>> >> >> >> >> -- >> >> *Brian West* >> brian at freeswitch.org >> >> *Twitter: @FreeSWITCH , @briankwest* >> >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> Book a phone call (CST) >> >> >> Allison prompts for FreeSWITCH: >> >> *https://www.gofundme.com/allison-prompts-for-freeswitch* >> >> >> Got Bugs? Report them here >> ! >> | Reddit: /r/freeswitch >> >> >> *T:*+19184209001 <+1%20918-420-9001> | *F:*+19184209002 >> <+1%20918-420-9002> | *M:*+1918424WEST (9378) >> *Skype:*briankwest >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Book a phone call (CST) Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170505/4b33ef56/attachment-0001.html From caioebassis at hotmail.com Fri May 5 19:27:21 2017 From: caioebassis at hotmail.com (Caio Assis) Date: Fri, 5 May 2017 15:27:21 +0000 Subject: [Freeswitch-users] MOD_SOFIA Message-ID: Good Afternoon. Is there a command I type that loads a new SIP account? When I type reload mod_sofia and it's already in use, it gives me an error message "+OK Reloading XML -ERR unloading module [Module in use.] -ERR loading module [Module already loaded]" I'm looking for an asterisk 'sip reload' freeswitch equivalent. If I have calls in progress and create a SIP account to use, I either have to wait for all calls to finish or restart Freeswitch, which makes it almost inviable. Can anyone help me with this? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170505/53c5e0d1/attachment.html From eschmidbauer at gmail.com Fri May 5 19:34:33 2017 From: eschmidbauer at gmail.com (E. Schmidbauer) Date: Fri, 5 May 2017 11:34:33 -0400 Subject: [Freeswitch-users] MOD_SOFIA In-Reply-To: References: Message-ID: try this `sofia profile [profile name] rescan` On Fri, May 5, 2017 at 11:27 AM, Caio Assis wrote: > Good Afternoon. > > > Is there a command I type that loads a new SIP account? When I type reload > mod_sofia and it's already in use, it gives me an error message "+OK > Reloading XML > -ERR unloading module [Module in use.] > -ERR loading module [Module already loaded]" > > > I'm looking for an asterisk 'sip reload' freeswitch equivalent. If I have > calls in progress and create a SIP account to use, I either have to wait > for all calls to finish or restart Freeswitch, which makes it almost > inviable. > > > Can anyone help me with this? > > > 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170505/db08f6ff/attachment.html From vladislaus at gmail.com Fri May 5 19:43:45 2017 From: vladislaus at gmail.com (Andres Gomez) Date: Fri, 5 May 2017 10:43:45 -0500 Subject: [Freeswitch-users] More pixelated video canvas on larger number of users In-Reply-To: <22925FE0-EA4C-40E4-81FE-9A5828790791@jerris.com> References: <22925FE0-EA4C-40E4-81FE-9A5828790791@jerris.com> Message-ID: Hello I have the same problem and I applied the changes that are made in the mails, however I did not work, at the moment of connecting 3 or more to videoconference the image is pixelated. My network schema is as follows FS <-------- MPLS 20M -------> Video Hardphones . My configuration is as follows: varls.xml conference.xml.conf Regards On Thu, Oct 13, 2016 at 12:08 PM, Michael Jerris wrote: > the verto js settings actually just do a set var in mod_verto of the rtp_video_max_bandwidth_ > vars > > On Oct 13, 2016, at 12:47 PM, Chad Phillips > wrote: > > Very clear, thanks. > > One more question: how does the Verto.newCall() method?s > ?incomingBandwidth? setting play with rtp_video_max_bandwidth_out and video-codec-bandwidth? > If incomingBandwidth is set lower than the other two, will it be used as > the max value for that call? > > On Wed, Oct 12, 2016 at 9:10 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> The value in the vars.xml is the absolute max. 4m is an acceptable value. >> >> We calculate the quality we send based on the canvas resolution using the >> kush guage >> >> http://vzaar.com/blog/video-encoding-guide/ >> >> >> The motion factor or quality mentioned in the link above matches the >> quality field in the config. >> >> If the calculated val exceeds the defined max, it will be limited to that >> max value. >> >> The max inbound is transmitted in the sdp limiting the max the browser >> will send. >> >> >> 1080p at 30fps quality 1 is in the vicinity of 4mb >> >> 1920 x 1080 x 30 x 1 x 0.07 / 1000 = 4354.56 >> >> >> >> >> >> On Wednesday, October 12, 2016, Chad Phillips >> wrote: >> >>> Tony, >>> >>> rtp_video_max_bandwidth_out was the one I was missing, thank you so >>> much! I had set video-codec-bandwidth in the conference config, but >>> totally forgot about that global setting. Once I upped it to match the >>> conference setting, quality issues disappeared :) >>> >>> I ended up using 4mb for both settings, would love a double check on my >>> reasoning to see if that?s the optimal value: >>> >>> I figure a 640x480 video at 30FPS uses about 1.5mbps >>> >>> My canvas is 1080x720, so: >>> >>> 640x480 = 307200 pixels >>> 1080x720 = 777600 pixels >>> >>> 777600 pixels / 307200 pixels = 2.53 times as many pixels >>> 1.5mbps x 2.53 = 3.8mbps, 4mbps fer good measure :) >>> >>> Sound right, or did I miss something? >>> >>> >>> On Wed, Oct 12, 2016 at 11:12 AM, Anthony Minessale < >>> anthony.minessale at gmail.com> wrote: >>> >>>> open another window/tab to chrome://webrtc-internals and look at all >>>> of the stats. >>>> >>>> Have you made sure you are not just maxing our your local bandwidth in >>>> either direction? >>>> Did you modify the rtp_video_max_bandwidth_in >>>> and rtp_video_max_bandwidth_out vars in vars.xml it defaults to 1m >>>> >>>> >>>> On Wed, Oct 12, 2016 at 12:29 PM, Chad Phillips < >>>> chad at apartmentlines.com> wrote: >>>> >>>>> Here?s another example, perhaps this illustrates it better: >>>>> https://youtu.be/PqIjubx4-wI >>>>> >>>>> And it doesn?t seem to be the entire canvas at once, it?s more of a >>>>> ?washing over?. I can definitely see it also affecting the banners in this >>>>> second example. >>>>> >>>>> I?m also happy to drop you into a live example, it?s pretty easy to >>>>> see what I?m talking about when you?re in the conference. >>>>> >>>>> On Wed, Oct 12, 2016 at 9:44 AM, Michael Jerris >>>>> wrote: >>>>> >>>>>> It does not seem to be the entire canvas to me. Look at the text >>>>>> labels on the layers? they don?t seem bad at all. >>>>>> >>>>>> On Oct 12, 2016, at 12:37 PM, Chad Phillips >>>>>> wrote: >>>>>> >>>>>> Mike, it?s the same if I remove video-codec-bandwidth and >>>>>> video-quality. This short video illustrates the issue: >>>>>> https://youtu.be/l8gpHhgmWRI >>>>>> >>>>>> Notice how with just my feed the quality is much better than with the >>>>>> multiple feeds. The pixelation effect seems to periodically ?wash over? the >>>>>> entire canvas. I?ve had many users report this same issue, even if they >>>>>> have excellent internet bandwidth. >>>>>> >>>>>> Gonzalo, I?ve got a quite beefy physical server, Xeon 32 core, 32GB >>>>>> RAM, and a nice fat network pipe. I haven?t ever pulled stats on packets >>>>>> in/out. >>>>>> >>>>>> On Wed, Oct 12, 2016 at 8:48 AM, Michael Jerris >>>>>> wrote: >>>>>> >>>>>>> is it the same if you remove the following: >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> ? >>>>>>> >>>>>>> On Oct 12, 2016, at 11:37 AM, Chad Phillips >>>>>>> wrote: >>>>>>> >>>>>>> Ok, I tested in Firefox, same pixelation issue. One or two video >>>>>>> feeds looks good, then it degrades as more feeds are added. >>>>>>> >>>>>>> Here are the relevant params from my conference config: >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> On Mon, Oct 10, 2016 at 1:29 PM, ?talo Rossi >>>>>>> wrote: >>>>>>> >>>>>>>> In which browser? >>>>>>>> >>>>>>>> On Mon, Oct 10, 2016 at 1:51 PM, Chad Phillips < >>>>>>>> chad at apartmentlines.com> wrote: >>>>>>>> >>>>>>>>> Running Verto/mod_conference videoconference on 1.6.11, I?ve >>>>>>>>> noticed that the entire canvas resolution seems more pixelated when the >>>>>>>>> number of users connected to the videoconference goes up. >>>>>>>>> >>>>>>>>> If just one person is connected, the image is very consistent and >>>>>>>>> clear, but getting into the 7-10 person range, the quality drops >>>>>>>>> noticeably. And I?m not talking about the quality of one particular video >>>>>>>>> on the canvas, but the entire canvas quality. >>>>>>>>> >>>>>>>>> I?ve done a recording of a videoconference on the server with a >>>>>>>>> larger number of users, and the video quality there is clear and >>>>>>>>> consistent, so it doesn?t seem to be an issue with either receiving or >>>>>>>>> muxing the feeds, but in how the end user is receiving the muxed video. >>>>>>>>> >>>>>>>>> I tried playing with the the ?video-quality? and >>>>>>>>> ?video-codec-bandwidth? conference params ? increasing the video-quality >>>>>>>>> from 1 to 3 didn?t seem to have much of an impact, increasing the >>>>>>>>> video-codec-bandwidth from 1mb to 2mb used quite a bit more CPU, but didn?t >>>>>>>>> seem to positively impact the video quality, either. >>>>>>>>> >>>>>>>>> Curious if I?m missing something in the config, or if there?s >>>>>>>>> something else I can do to improve the video quality with a larger number >>>>>>>>> of users. >>>>>>>>> >>>>>>>>> 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170505/96d8ed69/attachment-0001.html From mike at jerris.com Fri May 5 19:44:22 2017 From: mike at jerris.com (Michael Jerris) Date: Fri, 5 May 2017 11:44:22 -0400 Subject: [Freeswitch-users] mod_distributor from db In-Reply-To: <5b8ea216bcf44a35ab36d7bb316d9e5a@cm.nl> References: <4fa3d198c0034e9cbf17f88d14cf3cc6@cm.nl> <9C5AD1DC-A7F6-4E29-BE69-C0860A70E6F5@jerris.com> <5b8ea216bcf44a35ab36d7bb316d9e5a@cm.nl> Message-ID: <6C6E1A74-3263-487A-92AB-759CC9E6F038@jerris.com> no, its not per profile, the second one is something different, whats the details of the two requests? > On May 5, 2017, at 4:07 AM, Grant Bagdasarian wrote: > > Right! Thanks. > > What if I want to return multiple sip profiles? I see that it does 2 sofia.conf requests, does each request expect exactly one xml element? > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris > Sent: donderdag 4 mei 2017 19:42 > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] mod_distributor from db > > it requests the sip profiles. that is the sofia.conf request you see. > > On May 4, 2017, at 5:08 AM, Grant Bagdasarian > wrote: > > Does mod_xml_curl not request sip profiles? > I see a request going out to the configured URL but it?s only for sofia.conf and some other modules like distributor, translate, lua, etc. > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org ] On Behalf Of Grant Bagdasarian > Sent: dinsdag 2 mei 2017 09:53 > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] mod_distributor from db > > Nice! Thanks! > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org ] On Behalf Of Anthony Minessale > Sent: maandag 1 mei 2017 18:18 > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] mod_distributor from db > > mod_xml_curl could. > > > On Mon, May 1, 2017 at 9:38 AM, Grant Bagdasarian > wrote: > Hello, > > Is it possible to load the gateways of a sip profile from a database and the distributor config also from a database, using built in modules/functions? > > Regards, > > Grant -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170505/946c0c9c/attachment.html From mike at jerris.com Fri May 5 19:49:24 2017 From: mike at jerris.com (Michael Jerris) Date: Fri, 5 May 2017 11:49:24 -0400 Subject: [Freeswitch-users] Changing audio DTLS state from HANDSHAKE to FAIL issue In-Reply-To: References: <1493903221.S.81594.autosave.drafts.1493903788.8725@webmail.rediffmail.com> Message-ID: <239F18B3-90D8-41B7-AA63-92C33502E0EC@jerris.com> can you try this using sip.js or something thats confirmed to work when using linux as a server please. This may just be an issue with sipml5. > On May 5, 2017, at 9:53 AM, Brian West wrote: > > Clearly that isn't the case, I suspect a define is missing to actually enable ECDSA at this point. > > On Fri, May 5, 2017 at 8:47 AM, Gregor Nanger > wrote: > Brian, isn't this solved in 1.6.16? > > FS-10037 [core] Update OpenSSL to version 1.0.2k for Windows > > 2017-05-05 15:26 GMT+02:00 Brian West >: > OpenSSL on the windows build needs to be updated. > > https://freeswitch.org/jira/browse/FS-9510 > > On Thu, May 4, 2017 at 2:31 PM, Gregor Nanger > wrote: > This error is familiar to me, I think so, if I remembered correctly. > > I am using windows. I had this error when client was Chrome. I just updated openssl dlls. I think that openssl library is updated on windows build now with version 1.6.17. And this version is also compiled on FS FTP to just download and install it. > > 2017-05-04 19:15 GMT+02:00 Anthony Minessale >: > FS 1.5 sounds like a bad plan. > Try latest FS 1.6 or master. > > > On Thu, May 4, 2017 at 12:04 PM, ?talo Rossi > wrote: > Old openssl version maybe? > Em qui, 4 de mai de 2017 ?s 11:19, sachin > escreveu: > Hello All, > > ?I am trying to setup the Webrtc call between 2 SIPML5 client. I am using FreeSWITCH Version 1.6.14 on Windows. I am using internal LAN. > > The clients are getting registered over wss. I have created self signed certificates. In var.xml I have set the codecs setting as follows > > > > > ?I am able to establish the call and there is 2 way voice when I call from Sipmpl5 to Linphone and it works. But when I call from Linphone to Firefox (Simpl5) then the call is not getting established and I am getting following error > > 2017-05-04 11:29:04.543434 [ERR] switch_rtp.c:2907 audio Handshake failure 1 > 2017-05-04 11:29:04.543434 [INFO] switch_rtp.c:2908 Changing audio DTLS state from HANDSHAKE to FAIL > 2017-05-04 11:29:04.543434 [NOTICE] switch_rtp.c:2889 Hangup sofia/internal/sip:1001 at df7jal23ls0d.invalid [CS_EXCHANGE_MEDIA] [DESTINATION_OUT_OF_ORDER] > > My setup is as follows > > SIPML5 (FireFox browser 53.0)<-----> FS (v1.5)(Windows) <----> Linphone > > I am attaching the logs for the reference. > fs-logleve9.txt : Debug trace with loglevel =9 > ?fs-sip-trace.txt : Sip trace > Please let me know what could the issue and pointers to resolve the same. > > ?Thanks and Regards > SD -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170505/a0c79503/attachment-0001.html From mbodbg at gmx.net Fri May 5 19:58:12 2017 From: mbodbg at gmx.net (=?utf-8?Q?Markus_B=C3=B6nke?=) Date: Fri, 5 May 2017 17:58:12 +0200 Subject: [Freeswitch-users] Freeswitch stops handling register requests Message-ID: <6A32EAD3-FC96-46CB-BE70-0921F26CF70F@gmx.net> Hello, We faced today the following situation: Freeswitch suddenly stopped to respond to answer to REGISTER requests. At the same time it created more and more channels. A lot of them ?hung" in status ?RINGING?, after some time all calls were rejected with 503 Maximum Calls In Progress. After a restart, everything went back to normal I cannot find any WARNING or ERRROR/CRIT messages in the log. Also there is nothing special in the syslog at the time the issue appeared. We are using freeswitch 1.6.17 on debian jessie installed from debian packages. Calls are handled via esl / mod_event socket. Is there any known issue with this version? Is there anything else I can check? Thanks and Regards Markus From david.villasmil.work at gmail.com Fri May 5 20:13:16 2017 From: david.villasmil.work at gmail.com (David Villasmil) Date: Fri, 05 May 2017 16:13:16 +0000 Subject: [Freeswitch-users] MOD_SOFIA In-Reply-To: References: Message-ID: If you're talking about a new use account, relodxml On Fri, May 5, 2017 at 5:35 PM E. Schmidbauer wrote: > try this > `sofia profile [profile name] rescan` > > On Fri, May 5, 2017 at 11:27 AM, Caio Assis > wrote: > >> Good Afternoon. >> >> >> Is there a command I type that loads a new SIP account? When I type >> reload mod_sofia and it's already in use, it gives me an error message "+OK >> Reloading XML >> -ERR unloading module [Module in use.] >> -ERR loading module [Module already loaded]" >> >> >> I'm looking for an asterisk 'sip reload' freeswitch equivalent. If I have >> calls in progress and create a SIP account to use, I either have to wait >> for all calls to finish or restart Freeswitch, which makes it almost >> inviable. >> >> >> Can anyone help me with this? >> >> >> 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170505/b3bbe752/attachment.html From mike at jerris.com Fri May 5 21:03:10 2017 From: mike at jerris.com (Michael Jerris) Date: Fri, 5 May 2017 13:03:10 -0400 Subject: [Freeswitch-users] Freeswitch stops handling register requests In-Reply-To: <6A32EAD3-FC96-46CB-BE70-0921F26CF70F@gmx.net> References: <6A32EAD3-FC96-46CB-BE70-0921F26CF70F@gmx.net> Message-ID: What you describe could happen if your esl script suddenly sent many thousands of originate commands. > On May 5, 2017, at 11:58 AM, Markus B?nke wrote: > > Hello, > > We faced today the following situation: > > Freeswitch suddenly stopped to respond to answer to REGISTER requests. > At the same time it created more and more channels. A lot of them ?hung" in status ?RINGING?, after some time all calls were rejected with 503 Maximum Calls In Progress. > > After a restart, everything went back to normal > > I cannot find any WARNING or ERRROR/CRIT messages in the log. > Also there is nothing special in the syslog at the time the issue appeared. > > We are using freeswitch 1.6.17 on debian jessie installed from debian packages. Calls are handled via esl / mod_event socket. > > Is there any known issue with this version? Is there anything else I can check? > > Thanks and Regards > > Markus > From vladislaus at gmail.com Sat May 6 00:57:02 2017 From: vladislaus at gmail.com (Andres Gomez) Date: Fri, 5 May 2017 15:57:02 -0500 Subject: [Freeswitch-users] Video pixelated problem inconferences calls. Message-ID: Hello I have a problem with videoconference and I applied the changes that are made in the mails, however I did not work, at the moment of connecting 3 or more to videoconference the image is pixelated. My network schema is as follows softphones <---- lan ----> FS <-------- MPLS 20M -------> Video Hardphones . My configuration is as follows: varls.xml conference.xml.conf Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170505/e9e422ad/attachment.html From mbodbg at gmx.net Sat May 6 11:31:23 2017 From: mbodbg at gmx.net (=?utf-8?Q?Markus_B=C3=B6nke?=) Date: Sat, 6 May 2017 09:31:23 +0200 Subject: [Freeswitch-users] Freeswitch stops handling register requests In-Reply-To: References: <6A32EAD3-FC96-46CB-BE70-0921F26CF70F@gmx.net> Message-ID: <0261E3CA-BD15-4F58-A137-0B5E0E4D2FD6@gmx.net> I just checked the logs of our esl client and this was not the case. There is only one parrallel bridge to 8 destinations near the time where the problem occurred. Also the esl socket continued to receive and send events after freeswitch stopped to answer on register events. But maybe it has to do with the esl library like too many open threads or something like that. We recently updated the system from 1.4.26 to 1.6.17, everything was running for 2 days with around 150-200 calls parallel before the problem occurred. I?ve increased debug and monitoring, let?s see if it appears again in the next days. Thanks and regards Markus > Am 05.05.2017 um 19:03 schrieb Michael Jerris : > > What you describe could happen if your esl script suddenly sent many thousands of originate commands. > >> On May 5, 2017, at 11:58 AM, Markus B?nke wrote: >> >> Hello, >> >> We faced today the following situation: >> >> Freeswitch suddenly stopped to respond to answer to REGISTER requests. >> At the same time it created more and more channels. A lot of them ?hung" in status ?RINGING?, after some time all calls were rejected with 503 Maximum Calls In Progress. >> >> After a restart, everything went back to normal >> >> I cannot find any WARNING or ERRROR/CRIT messages in the log. >> Also there is nothing special in the syslog at the time the issue appeared. >> >> We are using freeswitch 1.6.17 on debian jessie installed from debian packages. Calls are handled via esl / mod_event socket. >> >> Is there any known issue with this version? Is there anything else I can check? >> >> Thanks and Regards >> >> Markus >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 colton.conor at gmail.com Sat May 6 18:41:39 2017 From: colton.conor at gmail.com (Colton Conor) Date: Sat, 6 May 2017 09:41:39 -0500 Subject: [Freeswitch-users] OPUS Codec In-Reply-To: References: <20170503071638.GA14349@mail.marktcontact.com> Message-ID: So what does today support FEC/PLC today? We tested two Polycom VVX phones with the opus codec, and overall thought that G.722 sounded much better than OPUS for some reason. So using Opus at 8 KHz, does that mean that there will be less or no transcoding to G.711 since its also in 8 KHz? What are most of the web only WebRTC companies using as far as OPUS goes? What bitrate and KHz? On Fri, May 5, 2017 at 8:33 AM, Brian West wrote: > None of the hardware phones do FEC/PLC last I tested, Sonus has OPUS > support, but again no FEC/PLC support. > > /b > > > On Wed, May 3, 2017 at 2:16 AM, EL wrote: > >> >> Yealink is supporting OPUS on several other models since firmware >> V81: >> >> Quote: >> "We will support opus on the standard V81 of >> SIP-T40P/T23P/T23G/T2 1(P) E2/T19(P) E2." >> >> Source: http://forum.yealink.com/forum/showthread.php?tid=29650&pid= >> 51262&mode=threaded >> >> I can confirm OPUS implementation on the 'T21P E2' model. >> >> -- >> EL >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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.org > > *Twitter: @FreeSWITCH , @briankwest* > > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > Book a phone call (CST) > > Allison prompts for FreeSWITCH: > > *https://www.gofundme.com/allison-prompts-for-freeswitch* > > > Got Bugs? Report them here ! | Reddit: > /r/freeswitch > > *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002> > | *M:*+1918424WEST (9378) > *Skype:*briankwest > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170506/7940cc2b/attachment.html From colin.morelli at gmail.com Sat May 6 19:21:24 2017 From: colin.morelli at gmail.com (Colin Morelli) Date: Sat, 6 May 2017 11:21:24 -0400 Subject: [Freeswitch-users] OPUS Codec In-Reply-To: References: <20170503071638.GA14349@mail.marktcontact.com> Message-ID: Transcoding != resampling. Often times you will have to do both to convert audio streams, where transcoding is the act of converting between the two audio codecs' data format, and resampling is changing the sampling rate of the audio (8khz in your case). Opus is a completely different audio codec than G.711. Even if they're both at 8 khz, you'll have to transcode between the two, though you may be able to avoid the cost of resampling the audio to a different rate. Opus can handle sampling rates from 8khz-48khz. Sampling rates make a *huge* difference to audio quality (they'll effectively define the range of audio frequencies you can capture). If you're running opus at a constant 8khz, the reason G.722 sounds better is almost certainly because it's using a 16khz sampling rate. Most likely if you run opus at 16khz, it'll sound similar to G.722. If you have the capacity (and the devices support it), pushing opus to super-wideband or full band (24khz/48khz sampling rates, respectively) will yield much better audio quality. My guess (admittedly it's just an educated assumption), is that most WebRTC companies are using fullband opus in VBR mode simply because this is the default in WebRTC. Without mangling the SDP, this is what you're going to get. Best, Colin On Sat, May 6, 2017 at 10:41 AM, Colton Conor wrote: > So what does today support FEC/PLC today? > > We tested two Polycom VVX phones with the opus codec, and overall thought > that G.722 sounded much better than OPUS for some reason. > > So using Opus at 8 KHz, does that mean that there will be less or no > transcoding to G.711 since its also in 8 KHz? > > What are most of the web only WebRTC companies using as far as OPUS goes? > What bitrate and KHz? > > On Fri, May 5, 2017 at 8:33 AM, Brian West wrote: > >> None of the hardware phones do FEC/PLC last I tested, Sonus has OPUS >> support, but again no FEC/PLC support. >> >> /b >> >> >> On Wed, May 3, 2017 at 2:16 AM, EL wrote: >> >>> >>> Yealink is supporting OPUS on several other models since firmware >>> V81: >>> >>> Quote: >>> "We will support opus on the standard V81 of >>> SIP-T40P/T23P/T23G/T2 1(P) E2/T19(P) E2." >>> >>> Source: http://forum.yealink.com/forum/showthread.php?tid=29650&pid= >>> 51262&mode=threaded >>> >>> I can confirm OPUS implementation on the 'T21P E2' model. >>> >>> -- >>> EL >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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.org >> >> *Twitter: @FreeSWITCH , @briankwest* >> >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> Book a phone call (CST) >> >> Allison prompts for FreeSWITCH: >> >> *https://www.gofundme.com/allison-prompts-for-freeswitch* >> >> >> Got Bugs? Report them here ! | Reddit: >> /r/freeswitch >> >> *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002> >> | *M:*+1918424WEST (9378) >> *Skype:*briankwest >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170506/79ac3980/attachment-0001.html From kris at kriskinc.com Sat May 6 22:26:12 2017 From: kris at kriskinc.com (Kristian Kielhofner) Date: Sat, 6 May 2017 13:26:12 -0500 Subject: [Freeswitch-users] OPUS Codec In-Reply-To: References: <20170503071638.GA14349@mail.marktcontact.com> Message-ID: FreeSWITCH can limit Opus to a configured sample rate but the OPUS specification says that the SDP must always specify 48 kHz and 2 channels. OPUS is a combination of SILK and CELT, with the original PLC+FEC coming from SILK. There are effectively three modes in OPUS - SILK, CELT, and hybrid. It's not clear to me how well PLC+FEC works in the hybrid and CELT modes used at higher sample rates. I haven't looked at it specifically (or tested it) but my suspicion is that PLC+FEC works much better the closer you get to "pure" SILK mode, which is essentially what limiting the sample rate to 8 kHz does. On Sat, May 6, 2017 at 10:21 AM, Colin Morelli wrote: > Transcoding != resampling. Often times you will have to do both to convert > audio streams, where transcoding is the act of converting between the two > audio codecs' data format, and resampling is changing the sampling rate of > the audio (8khz in your case). Opus is a completely different audio codec > than G.711. Even if they're both at 8 khz, you'll have to transcode between > the two, though you may be able to avoid the cost of resampling the audio to > a different rate. > > Opus can handle sampling rates from 8khz-48khz. Sampling rates make a huge > difference to audio quality (they'll effectively define the range of audio > frequencies you can capture). If you're running opus at a constant 8khz, the > reason G.722 sounds better is almost certainly because it's using a 16khz > sampling rate. Most likely if you run opus at 16khz, it'll sound similar to > G.722. If you have the capacity (and the devices support it), pushing opus > to super-wideband or full band (24khz/48khz sampling rates, respectively) > will yield much better audio quality. > > My guess (admittedly it's just an educated assumption), is that most WebRTC > companies are using fullband opus in VBR mode simply because this is the > default in WebRTC. Without mangling the SDP, this is what you're going to > get. > > Best, > Colin > > On Sat, May 6, 2017 at 10:41 AM, Colton Conor > wrote: >> >> So what does today support FEC/PLC today? >> >> We tested two Polycom VVX phones with the opus codec, and overall thought >> that G.722 sounded much better than OPUS for some reason. >> >> So using Opus at 8 KHz, does that mean that there will be less or no >> transcoding to G.711 since its also in 8 KHz? >> >> What are most of the web only WebRTC companies using as far as OPUS goes? >> What bitrate and KHz? >> >> On Fri, May 5, 2017 at 8:33 AM, Brian West wrote: >>> >>> None of the hardware phones do FEC/PLC last I tested, Sonus has OPUS >>> support, but again no FEC/PLC support. >>> >>> /b >>> >>> >>> On Wed, May 3, 2017 at 2:16 AM, EL wrote: >>>> >>>> >>>> Yealink is supporting OPUS on several other models since firmware >>>> V81: >>>> >>>> Quote: >>>> "We will support opus on the standard V81 of >>>> SIP-T40P/T23P/T23G/T2 1(P) E2/T19(P) E2." >>>> >>>> Source: >>>> http://forum.yealink.com/forum/showthread.php?tid=29650&pid=51262&mode=threaded >>>> >>>> I can confirm OPUS implementation on the 'T21P E2' model. >>>> >>>> -- >>>> EL >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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.org >>> >>> Twitter: @FreeSWITCH , @briankwest >>> >>> http://www.freeswitchbook.com >>> http://www.freeswitchcookbook.com >>> >>> Book a phone call (CST) >>> >>> Allison prompts for FreeSWITCH: >>> >>> https://www.gofundme.com/allison-prompts-for-freeswitch >>> >>> Got Bugs? Report them here! | Reddit: /r/freeswitch >>> >>> T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) >>> Skype:briankwest >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 -- Kristian Kielhofner From ssinyagin at gmail.com Sat May 6 22:49:40 2017 From: ssinyagin at gmail.com (Stanislav Sinyagin) Date: Sat, 6 May 2017 20:49:40 +0200 Subject: [Freeswitch-users] Script language choice In-Reply-To: <20170505093846.GA2462@blomma.liberationtech.net> References: <20170505093846.GA2462@blomma.liberationtech.net> Message-ID: Choose the language where you are the most efficient as a developer. I use Perl exactly for that reason. On 5 May 2017 11:40, "Oivvio Polite" wrote: > > There's a page about script language choice on the wiki. There's a > caveat at to top warning that the information is stale. > > https://freeswitch.org/confluence/display/FREESWITCH/ > Script+Language+Choice > > Anyway, it states that Lua is the preferred scripting language due > efficacy. > > In a scenario where I'm not expecting very high loads are there other > reasons for going with Lua anyway? I'm thinking that it's probably more > battle tested if it's the most common choice among FS-developers. > > regards Oivvio > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170506/f5b1721b/attachment.html From ssinyagin at gmail.com Sat May 6 23:04:20 2017 From: ssinyagin at gmail.com (Stanislav Sinyagin) Date: Sat, 6 May 2017 21:04:20 +0200 Subject: [Freeswitch-users] Freeswitch and IoT In-Reply-To: References: Message-ID: You can call originate with park() from an ESL client, and set your own SIP address as destination. Then the b-leg will go through the dialplan. On 1 May 2017 16:10, "Marcel Haldemann" wrote: > Hi Giovanni, > > > > Thanks for ur reply. Here is what im currently thinking of to use > FreeSwitch for (currently it?s just an idea): > > > > I would like to use FreeSwitch for IoT (for example to run or check > something against a dialplan if sensor or ?alarm event? values arrive via > http/mqtt/amqp or similar protocols). For this purpose I would like to > create a channel without a sip call (I would write wrappers for all the > necessary protocols, get the info on all protocols and set the variables in > the same way no matter what protocol they come from). > > > > The variables I would set would be things such as location and name of the > sensor and the value (Values coming from mqtt, amqp or http). > > > > Then I would like to run it against a dialplan with all the possibilities > as it would be a call (for sure I would have to check some variables). (I > would use mod_xml_curl to get the dialplan in redis) > > > > If necessary i would like to add an a-leg with bridge and maybe play some > files an maybe get some dtmfs with lua (ivr). > > > > If necessary I would like to add a b-leg with bridge. > > I now asked me whether it?s possible to create a channel out of nothing, > set some variables and let it run against the dialplan context. > > > > Is this possible in some way ? > > It must not be with event_socket I also could use some other mod. > > But maybe it?s necessary to write a mod_endpoint_mqtt or similar for this ? > > > > Regards, > > Marcel > > > > > > *Von:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *Giovanni > Maruzzelli > *Gesendet:* Samstag, 29. April 2017 19:46 > *An:* FreeSWITCH Users Help > *Betreff:* Re: [Freeswitch-users] Freeswitch and IoT > > > > Hello Marcel, > > can you elaborate? Don't be so terse, there can be a lot of meanings in > your question, and lot of different answers. > > Please specify a clear use case, etc etc > > -giovanni > > > > On 29 April 2017 at 19:31, Marcel Haldemann > wrote: > > Hi Guys, > > > > is it possible to create a channel using event socket (without calling a > number) ? > > > > LG, > > Marcel > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170506/610a21d1/attachment-0001.html From lists at telefaks.de Sun May 7 18:39:54 2017 From: lists at telefaks.de (Peter Steinbach) Date: Sun, 07 May 2017 16:39:54 +0200 Subject: [Freeswitch-users] Freeswitch stops handling register requests In-Reply-To: <0261E3CA-BD15-4F58-A137-0B5E0E4D2FD6@gmx.net> References: <6A32EAD3-FC96-46CB-BE70-0921F26CF70F@gmx.net> <0261E3CA-BD15-4F58-A137-0B5E0E4D2FD6@gmx.net> Message-ID: <590F31BA.5070304@telefaks.de> >But maybe it has to do with the esl library like too many open threads or something like that. Hello Markus, I remember, we had some issues a while ago, when Freeswitch fired a huge amount of esl events under heavy load, and we could not handle them in time. Therefore we built an ESL process which simply and quickly pushed the events to some Queues on a Redis Server. This process worked much quicker, than Freeswitch could fire events, so events may not be stuck inside Freeswitch waiting to be processed. On the other side we then had workers for each Redis Queue, who pulled the events and processed them. That way we can process all events from Freeswitch in time without losing any events. Maybe you experience something similar? Best regards Peter On 05/06/17 09:31, Markus B?nke wrote: > I just checked the logs of our esl client and this was not the case. There is only one parrallel bridge to 8 destinations near the time where the problem occurred. Also the esl socket continued to receive and send events after freeswitch stopped to answer on register events. But maybe it has to do with the esl library like too many open threads or something like that. We recently updated the system from 1.4.26 to 1.6.17, everything was running for 2 days with around 150-200 calls parallel before the problem occurred. I?ve increased debug and monitoring, let?s see if it appears again in the next days. > > Thanks and regards > > Markus > > >> Am 05.05.2017 um 19:03 schrieb Michael Jerris : >> >> What you describe could happen if your esl script suddenly sent many thousands of originate commands. >> >>> On May 5, 2017, at 11:58 AM, Markus B?nke wrote: >>> >>> Hello, >>> >>> We faced today the following situation: >>> >>> Freeswitch suddenly stopped to respond to answer to REGISTER requests. >>> At the same time it created more and more channels. A lot of them ?hung" in status ?RINGING?, after some time all calls were rejected with 503 Maximum Calls In Progress. >>> >>> After a restart, everything went back to normal >>> >>> I cannot find any WARNING or ERRROR/CRIT messages in the log. >>> Also there is nothing special in the syslog at the time the issue appeared. >>> >>> We are using freeswitch 1.6.17 on debian jessie installed from debian packages. Calls are handled via esl / mod_event socket. >>> >>> Is there any known issue with this version? Is there anything else I can check? >>> >>> Thanks and Regards >>> >>> Markus >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 gregor at infomedia.si Sun May 7 21:15:34 2017 From: gregor at infomedia.si (Gregor Nanger) Date: Sun, 7 May 2017 19:15:34 +0200 Subject: [Freeswitch-users] Freeswitch stops handling register requests In-Reply-To: <590F31BA.5070304@telefaks.de> References: <6A32EAD3-FC96-46CB-BE70-0921F26CF70F@gmx.net> <0261E3CA-BD15-4F58-A137-0B5E0E4D2FD6@gmx.net> <590F31BA.5070304@telefaks.de> Message-ID: Is there a way to see waiting ESL events to be processed? 2017-05-07 16:39 GMT+02:00 Peter Steinbach : > >But maybe it has to do with the esl library like too many open threads > or something like that. > > Hello Markus, > > I remember, we had some issues a while ago, when Freeswitch fired a huge > amount of esl events under heavy load, and we could not handle them in > time. Therefore we built an ESL process which simply and quickly pushed > the events to some Queues on a Redis Server. This process worked much > quicker, than Freeswitch could fire events, so events may not be stuck > inside Freeswitch waiting to be processed. On the other side we then had > workers for each Redis Queue, who pulled the events and processed them. > That way we can process all events from Freeswitch in time without > losing any events. > > Maybe you experience something similar? > > Best regards > Peter > > On 05/06/17 09:31, Markus B?nke wrote: > > I just checked the logs of our esl client and this was not the case. > There is only one parrallel bridge to 8 destinations near the time where > the problem occurred. Also the esl socket continued to receive and send > events after freeswitch stopped to answer on register events. But maybe it > has to do with the esl library like too many open threads or something like > that. We recently updated the system from 1.4.26 to 1.6.17, everything was > running for 2 days with around 150-200 calls parallel before the problem > occurred. I?ve increased debug and monitoring, let?s see if it appears > again in the next days. > > > > Thanks and regards > > > > Markus > > > > > >> Am 05.05.2017 um 19:03 schrieb Michael Jerris : > >> > >> What you describe could happen if your esl script suddenly sent many > thousands of originate commands. > >> > >>> On May 5, 2017, at 11:58 AM, Markus B?nke wrote: > >>> > >>> Hello, > >>> > >>> We faced today the following situation: > >>> > >>> Freeswitch suddenly stopped to respond to answer to REGISTER requests. > >>> At the same time it created more and more channels. A lot of them > ?hung" in status ?RINGING?, after some time all calls were rejected with > 503 Maximum Calls In Progress. > >>> > >>> After a restart, everything went back to normal > >>> > >>> I cannot find any WARNING or ERRROR/CRIT messages in the log. > >>> Also there is nothing special in the syslog at the time the issue > appeared. > >>> > >>> We are using freeswitch 1.6.17 on debian jessie installed from debian > packages. Calls are handled via esl / mod_event socket. > >>> > >>> Is there any known issue with this version? Is there anything else I > can check? > >>> > >>> Thanks and Regards > >>> > >>> Markus > >>> > >> > >> ____________________________________________________________ > _____________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://confluence.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at 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 -- 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170507/28597cff/attachment.html From karl at xtronics.com Mon May 8 04:47:09 2017 From: karl at xtronics.com (Karl Schmidt) Date: Sun, 7 May 2017 19:47:09 -0500 Subject: [Freeswitch-users] Realtime kernels - CONFIG_PREEMPT_RT In-Reply-To: References: <6A32EAD3-FC96-46CB-BE70-0921F26CF70F@gmx.net> <0261E3CA-BD15-4F58-A137-0B5E0E4D2FD6@gmx.net> <590F31BA.5070304@telefaks.de> Message-ID: I just searched the subject line of some 28000 messages here - and did not find anyone looking into Real Time(RT)kernels. I remember many years ago using an RT kernel for some audio programs - didn't appear to make any difference - likely everything has just gotten that much faster. If anyone has worked with this - please let me know if you found any differences. -- -------------------------------------------------------------------------------- Karl Schmidt EMail Karl at xtronics.com Transtronics, Inc. WEB http://secure.transtronics.com 3209 West 9th Street Ph (785) 841-3089 Lawrence, KS 66049 FAX (785) 841-0434 Coordination does not run in my family, it stumbles. -------------------------------------------------------------------------------- From gb at cm.nl Mon May 8 10:46:23 2017 From: gb at cm.nl (Grant Bagdasarian) Date: Mon, 8 May 2017 06:46:23 +0000 Subject: [Freeswitch-users] mod_distributor from db In-Reply-To: <6C6E1A74-3263-487A-92AB-759CC9E6F038@jerris.com> References: <4fa3d198c0034e9cbf17f88d14cf3cc6@cm.nl> <9C5AD1DC-A7F6-4E29-BE69-C0860A70E6F5@jerris.com> <5b8ea216bcf44a35ab36d7bb316d9e5a@cm.nl> <6C6E1A74-3263-487A-92AB-759CC9E6F038@jerris.com> Message-ID: <55d2040e11234f8385d5fa283897ddd3@cm.nl> First Request: GET /?hostname=freeswitch-app-ivr-03.§ion=configuration&tag_name=configuration&key_name=name&key_value=sofia.conf&Event-Name=REQUEST_PARAMS&Core-UUID=909a6aa8-b885-48cc-b8e8-7a09fe93c4ea&FreeSWITCH-Hostname=freeswitch-app-ivr-03.&FreeSWITCH-Switchname=freeswitch-app-ivr-03.&FreeSWITCH-IPv4=10.14.26.25&FreeSWITCH-IPv6=%3A%3A1&Event-Date-Local=2017-05-08%2008%3A42%3A57&Event-Date-GMT=Mon,%2008%20May%202017%2006%3A42%3A57%20GMT&Event-Date-Timestamp=1494225777727856&Event-Calling-File=sofia.c&Event-Calling-Function=config_sofia&Event-Calling-Line-Number=4246&Event-Sequence=21 HTTP/1.1. Second Request: GET /?hostname=freeswitch-app-ivr-03.§ion=configuration&tag_name=configuration&key_name=name&key_value=sofia.conf&Event-Name=REQUEST_PARAMS&Core-UUID=909a6aa8-b885-48cc-b8e8-7a09fe93c4ea&FreeSWITCH-Hostname=freeswitch-app-ivr-03.&FreeSWITCH-Switchname=freeswitch-app-ivr-03.&FreeSWITCH-IPv4=10.14.26.25&FreeSWITCH-IPv6=%3A%3A1&Event-Date-Local=2017-05-08%2008%3A42%3A57&Event-Date-GMT=Mon,%2008%20May%202017%2006%3A42%3A57%20GMT&Event-Date-Timestamp=1494225777736626&Event-Calling-File=sofia.c&Event-Calling-Function=launch_sofia_worker_thread&Event-Calling-Line-Number=2949&Event-Sequence=24&profile=local_eth0 HTTP/1.1. First request has the config_sofia in the Event-Calliing-Function, while the second one has launch_sofia_worker_thread for what it seems like an existing sip_profile ?local_eth0? which actually does exist. But I?m trying to get all possible sip profiles from the web api, since the hostname will determine the number of sip profiles required for the fs server. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris Sent: vrijdag 5 mei 2017 17:44 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] mod_distributor from db no, its not per profile, the second one is something different, whats the details of the two requests? On May 5, 2017, at 4:07 AM, Grant Bagdasarian > wrote: Right! Thanks. What if I want to return multiple sip profiles? I see that it does 2 sofia.conf requests, does each request expect exactly one xml element? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris Sent: donderdag 4 mei 2017 19:42 To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] mod_distributor from db it requests the sip profiles. that is the sofia.conf request you see. On May 4, 2017, at 5:08 AM, Grant Bagdasarian > wrote: Does mod_xml_curl not request sip profiles? I see a request going out to the configured URL but it?s only for sofia.conf and some other modules like distributor, translate, lua, etc. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Grant Bagdasarian Sent: dinsdag 2 mei 2017 09:53 To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] mod_distributor from db Nice! Thanks! From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: maandag 1 mei 2017 18:18 To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] mod_distributor from db mod_xml_curl could. On Mon, May 1, 2017 at 9:38 AM, Grant Bagdasarian > wrote: Hello, Is it possible to load the gateways of a sip profile from a database and the distributor config also from a database, using built in modules/functions? Regards, Grant -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170508/f7e0e1f4/attachment-0001.html From ak at hejdu.dk Mon May 8 16:03:40 2017 From: ak at hejdu.dk (Allan Kristensen) Date: Mon, 08 May 2017 12:03:40 +0000 Subject: [Freeswitch-users] Noisy logs (switch_rtp.c: 1086 missed 1) Message-ID: Hello, I'm getting thousand of these every hour on my Freeswitch log: May 8 08:09:32 sip1b media-switch-container[18466]: 2017-05-08 08:09:24.807956 [WARNING] switch_rtp.c:1086 missed 1 May 8 08:09:32 sip1b media-switch-container[18466]: 2017-05-08 08:09:25.787957 [WARNING] switch_rtp.c:1086 missed 1 : I can see it's related to calls coming from an Asterisk server but the INVITE's look fine to me and calls are active. The customer says there are no problems, but it's making my log monitor unhappy (they are warnings after all). I've looked into the code and I can see it's something to do with handling of ICE, but why it appears is a little mysterious to me. (I using "apply-candidate-acl = wan.auto", so nothing fancy there). Any clues? /Allan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170508/078e96f0/attachment.html From italo at freeswitch.org Mon May 8 16:21:06 2017 From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=) Date: Mon, 08 May 2017 12:21:06 +0000 Subject: [Freeswitch-users] Noisy logs (switch_rtp.c: 1086 missed 1) In-Reply-To: References: Message-ID: It's probably rtp lost packets, enable jitterbuffer and see if it helps Em seg, 8 de mai de 2017 ?s 09:05, Allan Kristensen escreveu: > Hello, > > I'm getting thousand of these every hour on my Freeswitch log: > > May 8 08:09:32 sip1b media-switch-container[18466]: 2017-05-08 > 08:09:24.807956 [WARNING] switch_rtp.c:1086 missed 1 > May 8 08:09:32 sip1b media-switch-container[18466]: 2017-05-08 > 08:09:25.787957 [WARNING] switch_rtp.c:1086 missed 1 > : > > I can see it's related to calls coming from an Asterisk server but the > INVITE's look fine to me and calls are active. > The customer says there are no problems, but it's making my log monitor > unhappy (they are warnings after all). > > I've looked into the code and I can see it's something to do with handling > of ICE, but why it appears is a little mysterious to me. > (I using "apply-candidate-acl = wan.auto", so nothing fancy there). > > Any clues? > > /Allan > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170508/fa581eb3/attachment.html From mbodbg at gmx.net Mon May 8 16:59:20 2017 From: mbodbg at gmx.net (=?utf-8?Q?Markus_B=C3=B6nke?=) Date: Mon, 8 May 2017 14:59:20 +0200 Subject: [Freeswitch-users] Freeswitch stops handling register requests In-Reply-To: <590F31BA.5070304@telefaks.de> References: <6A32EAD3-FC96-46CB-BE70-0921F26CF70F@gmx.net> <0261E3CA-BD15-4F58-A137-0B5E0E4D2FD6@gmx.net> <590F31BA.5070304@telefaks.de> Message-ID: <35CA8D2C-D8CE-40CE-82C7-9E8A106EEFF8@gmx.net> Hello Peter, thanks for the hint, but can it really happen that events stuck inside freeswitch and freeswitch partially crashed because if this? Or wasn't it more a problem of the Client that event?s were discarded if they haven?t been processed fast enough. How many calls did you handle on your server when the problem occurred, are you using inbound or outbound mode of mod_event_socket? Best regards Markus > Am 07.05.2017 um 16:39 schrieb Peter Steinbach : > >> But maybe it has to do with the esl library like too many open threads > or something like that. > > Hello Markus, > > I remember, we had some issues a while ago, when Freeswitch fired a huge > amount of esl events under heavy load, and we could not handle them in > time. Therefore we built an ESL process which simply and quickly pushed > the events to some Queues on a Redis Server. This process worked much > quicker, than Freeswitch could fire events, so events may not be stuck > inside Freeswitch waiting to be processed. On the other side we then had > workers for each Redis Queue, who pulled the events and processed them. > That way we can process all events from Freeswitch in time without > losing any events. > > Maybe you experience something similar? > > Best regards > Peter > > On 05/06/17 09:31, Markus B?nke wrote: >> I just checked the logs of our esl client and this was not the case. There is only one parrallel bridge to 8 destinations near the time where the problem occurred. Also the esl socket continued to receive and send events after freeswitch stopped to answer on register events. But maybe it has to do with the esl library like too many open threads or something like that. We recently updated the system from 1.4.26 to 1.6.17, everything was running for 2 days with around 150-200 calls parallel before the problem occurred. I?ve increased debug and monitoring, let?s see if it appears again in the next days. >> >> Thanks and regards >> >> Markus >> >> >>> Am 05.05.2017 um 19:03 schrieb Michael Jerris : >>> >>> What you describe could happen if your esl script suddenly sent many thousands of originate commands. >>> >>>> On May 5, 2017, at 11:58 AM, Markus B?nke wrote: >>>> >>>> Hello, >>>> >>>> We faced today the following situation: >>>> >>>> Freeswitch suddenly stopped to respond to answer to REGISTER requests. >>>> At the same time it created more and more channels. A lot of them ?hung" in status ?RINGING?, after some time all calls were rejected with 503 Maximum Calls In Progress. >>>> >>>> After a restart, everything went back to normal >>>> >>>> I cannot find any WARNING or ERRROR/CRIT messages in the log. >>>> Also there is nothing special in the syslog at the time the issue appeared. >>>> >>>> We are using freeswitch 1.6.17 on debian jessie installed from debian packages. Calls are handled via esl / mod_event socket. >>>> >>>> Is there any known issue with this version? Is there anything else I can check? >>>> >>>> Thanks and Regards >>>> >>>> Markus >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 snehsach at rediffmail.com Mon May 8 08:45:54 2017 From: snehsach at rediffmail.com (sachin ) Date: 8 May 2017 04:45:54 -0000 Subject: [Freeswitch-users] =?utf-8?q?Changing_audio_DTLS_state_from_HANDS?= =?utf-8?q?HAKE_to_FAIL_issue?= In-Reply-To: Message-ID: <1493992618.S.41103.25639.f4-234-233.1494218754.24193@webmail.rediffmail.com> Hello Gregor,I have upgraded the openssl version to 1.1.0e from the following link. But the issue remains the same.http://slproweb.com/download/Win64OpenSSL-1_1_0e.exe?Is the above way correct to upgrade the open ssl.?Thanks and Regards,?SachinFrom: Brian West <brian at freeswitch.org>Sent: Fri, 05 May 2017 19:26:58To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>Subject: Re: [Freeswitch-users] Changing audio DTLS state from HANDSHAKE to FAIL issue Clearly that isn't the case, I suspect a define is missing to actually enable ECDSA at this point.  On Fri, May 5, 2017 at 8:47 AM, Gregor Nanger <gregor at infomedia.si> wrote: Brian, isn't this solved in 1.6.16?   FS-10037 [core] Update OpenSSL to version 1.0.2k for Windows  2017-05-05 15:26 GMT+02:00 Brian West <brian at freeswitch.org>: OpenSSL on the windows build needs to be updated.  https://freeswitch.org/jira/browse/FS-9510  On Thu, May 4, 2017 at 2:31 PM, Gregor Nanger <gregor at infomedia.si> wrote: This error is familiar to me, I think so, if I remembered correctly.    I am using windows. I had this error when client was Chrome. I just updated openssl dlls. I think that openssl library is updated on windows build now with version 1.6.17. And this version is also compiled on FS FTP to just download and install it.  2017-05-04 19:15 GMT+02:00 Anthony Minessale <anthony.minessale at gmail.com>: FS 1.5 sounds like a bad plan.Try latest FS 1.6 or master.    On Thu, May 4, 2017 at 12:04 PM, Ítalo Rossi <italo at freeswitch.org> wrote: Old openssl version maybe?Em qui, 4 de mai de 2017 às 11:19, sachin <snehsach at rediffmail.com> escreveu: Hello All,?I am trying to setup the Webrtc call between 2 SIPML5 client. I am using FreeSWITCH Version 1.6.14 on Windows. I am using internal LAN.The clients are getting registered over wss. I have created self signed certificates.  In var.xml I have set the codecs setting as follows  <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMU,PCMA">  <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA">?I am able to establish the call and there is 2 way voice when I call from Sipmpl5 to Linphone and it works. But when I call from Linphone to Firefox (Simpl5) then the call is not getting established and I am getting following error2017-05-04 11:29:04.543434 [ERR] switch_rtp.c:2907 audio Handshake failure 12017-05-04 11:29:04.543434 [INFO] switch_rtp.c:2908 Changing audio DTLS state from HANDSHAKE to FAIL2017-05-04 11:29:04.543434 [NOTICE] switch_rtp.c:2889 Hangup sofia/internal/sip:1001 at df7jal23ls0d.invalid [CS_EXCHANGE_MEDIA] [DESTINATION_OUT_OF_ORDER]My setup is as followsSIPML5 (FireFox browser 53.0)<-----> FS (v1.5)(Windows) <----> LinphoneI am attaching the logs for the reference.fs-logleve9.txt : Debug trace with loglevel =9?fs-sip-trace.txt : Sip tracePlease let me know what could the issue and pointers to resolve the same.?Thanks and RegardsSD_________________________________________________________________________Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.comOfficial FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.comFreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org_________________________________________________________________________Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.comOfficial FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.comFreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://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=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA _________________________________________________________________________Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.comOfficial FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.comFreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://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.orghttp://www.freeswitchsolutions.comOfficial FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.comFreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org   -- Brian Westbrian at freeswitch.org Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com Book a phone call (CST)Allison prompts for FreeSWITCH: https://www.gofundme.com/allison-prompts-for-freeswitch Got Bugs? Report them here! | Reddit: /r/freeswitch T: 19184209001 | F: 19184209002 | M:+1918424WEST (9378)Skype:briankwest_________________________________________________________________________Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.comOfficial FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.comFreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://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.orghttp://www.freeswitchsolutions.comOfficial FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.comFreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org   -- Brian Westbrian at freeswitch.org Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com Book a phone call (CST)Allison prompts for FreeSWITCH: https://www.gofundme.com/allison-prompts-for-freeswitch Got Bugs? Report them here! | Reddit: /r/freeswitch T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378)Skype:briankwest_________________________________________________________________________Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.comOfficial FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.comFreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170508/5be9eac7/attachment-0001.html From g.volpe at studiofox.it Mon May 8 10:19:35 2017 From: g.volpe at studiofox.it (Gaetano Volpe) Date: Mon, 8 May 2017 08:19:35 +0200 Subject: [Freeswitch-users] One way audio after sometime Message-ID: Hi guys, I'm running the last stable version of Freeswitch in my environment. FreeSWITCH Version 1.6.14~64bit ( 64bit) I've a very strange problem with a group of extensions of sales department An isdn pri line is provided by a cisco router (2921) through a sip trunk. When the calls volume increases, suddenly some phones experience one way audio (the customer hears, sellers don't). When it happesn the only way to get the phone back to work is to reboot it. We're running all polycom vvx 300 phones. What I've tried so far: 1. Change the codec to g.711 2. Upgrade freeswitch 3. Upgrade Polycom firmware 4. Downgrade Polycom firmware I've captured packets with wireshark packets and I can see both audio channels in the call trace. So I guessed it could be the phones but it's very strange. What do you think about it? Thanks *Gaetano VolpeCEO, CTO & Systems Analyst*g.volpe at studiofox.it *- Mob.+39.3485260691* -- *STUDIO FOX SERVICE di G.Volpe* Via Spineto, 24 - 70038 - Terlizzi (Bari) - Italy Ph. +39.0803540043 - Fax +39.0803515004 C.F. VLPGTN83E10L109S - P.Iva IT05803550721 www.studiofox.it La presente comunicazione ed i suoi eventuali allegati potrebbero contenere informazioni legate al segreto professionale o dati soggetti alla vigente Legge sulla Privacy e sono comunque strettamente riservati e confidenziali per il destinatario. In mancanza di autorizzazione ? vietata la lettura, la riproduzione, la diffusione e l'estrazione di copie. Pertanto, nel caso in cui aveste ricevuto per errore la presente comunicazione, Vi preghiamo di provvedere a distruggerla e di contattare immediatamente i nostri uffici ai numeri su indicati. Grazie. This communication and its attachments, if any, may contain information covered by professional secrecy or data subject to current privacy law, and they are intended only for the use of the individual to whom it is addressed. They may contain information that is legally privileged, confidential or exempt from disclosure. Reading, reproduction, disclosure, and extraction of copies are prohibited unless expressly authorized. If you have received this communication in error, please notify immediately our offices at the contact details specified above and delete the communication without retaining any copies. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170508/f5bb528a/attachment.html From snehsach at rediffmail.com Mon May 8 16:08:57 2017 From: snehsach at rediffmail.com (sachin ) Date: 8 May 2017 12:08:57 -0000 Subject: [Freeswitch-users] =?utf-8?q?Changing_audio_DTLS_state_from_HANDS?= =?utf-8?q?HAKE_to_FAIL_issue?= In-Reply-To: <239F18B3-90D8-41B7-AA63-92C33502E0EC@jerris.com> Message-ID: <1493999540.S.17847.21554.f4-235-140.1494245336.524@webmail.rediffmail.com> Hello Michael,I tried using Sip.js instead of sipml5 and with Linux as the server. I have installed the FreeSWITCH Version 1.9.0+git~20170501T171230Z~e3ef041517~64bit (git e3ef041 2017-05-01 17:12:30Z 64bit)I am getting INCOMPATIBLE  DESTINATION error. Also tried sipml5.. with bot the client I am getting the same error.?Please let me know what I am missing.?Thanks and Regards,?SDFrom: Michael Jerris <mike at jerris.com>Sent: Fri, 05 May 2017 21:22:20To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>Subject: Re: [Freeswitch-users] Changing audio DTLS state from HANDSHAKE to FAIL issuecan you try this using sip.js or something thats confirmed to work when using linux as a server please.  This may just be an issue with sipml5. On May 5, 2017, at 9:53 AM, Brian West <brian at freeswitch.org> wrote:  Clearly that isn't the case, I suspect a define is missing to actually enable ECDSA at this point.  On Fri, May 5, 2017 at 8:47 AM, Gregor Nanger <gregor at infomedia.si> wrote: Brian, isn't this solved in 1.6.16?   FS-10037 [core] Update OpenSSL to version 1.0.2k for Windows  2017-05-05 15:26 GMT+02:00 Brian West <brian at freeswitch.org>: OpenSSL on the windows build needs to be updated.  https://freeswitch.org/jira/browse/FS-9510  On Thu, May 4, 2017 at 2:31 PM, Gregor Nanger <gregor at infomedia.si> wrote: This error is familiar to me, I think so, if I remembered correctly.    I am using windows. I had this error when client was Chrome. I just updated openssl dlls. I think that openssl library is updated on windows build now with version 1.6.17. And this version is also compiled on FS FTP to just download and install it.  2017-05-04 19:15 GMT+02:00 Anthony Minessale <anthony.minessale at gmail.com>: FS 1.5 sounds like a bad plan.Try latest FS 1.6 or master.    On Thu, May 4, 2017 at 12:04 PM, Ítalo Rossi <italo at freeswitch.org> wrote: Old openssl version maybe?Em qui, 4 de mai de 2017 às 11:19, sachin <snehsach at rediffmail.com> escreveu: Hello All,?I am trying to setup the Webrtc call between 2 SIPML5 client. I am using FreeSWITCH Version 1.6.14 on Windows. I am using internal LAN.The clients are getting registered over wss. I have created self signed certificates.  In var.xml I have set the codecs setting as follows  <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMU,PCMA">  <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA">?I am able to establish the call and there is 2 way voice when I call from Sipmpl5 to Linphone and it works. But when I call from Linphone to Firefox (Simpl5) then the call is not getting established and I am getting following error2017-05-04 11:29:04.543434 [ERR] switch_rtp.c:2907 audio Handshake failure 12017-05-04 11:29:04.543434 [INFO] switch_rtp.c:2908 Changing audio DTLS state from HANDSHAKE to FAIL2017-05-04 11:29:04.543434 [NOTICE] switch_rtp.c:2889 Hangup sofia/internal/sip:1001 at df7jal23ls0d.invalid [CS_EXCHANGE_MEDIA] [DESTINATION_OUT_OF_ORDER]My setup is as followsSIPML5 (FireFox browser 53.0)<-----> FS (v1.5)(Windows) <----> LinphoneI am attaching the logs for the reference.fs-logleve9.txt : Debug trace with loglevel =9?fs-sip-trace.txt : Sip tracePlease let me know what could the issue and pointers to resolve the same.?Thanks and RegardsSD_________________________________________________________________________Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.comOfficial FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.comFreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170508/8cf4d5f7/attachment.html From mike at jerris.com Mon May 8 19:14:30 2017 From: mike at jerris.com (Michael Jerris) Date: Mon, 8 May 2017 11:14:30 -0400 Subject: [Freeswitch-users] One way audio after sometime In-Reply-To: References: Message-ID: The latest current release of FreeSWITCH is 1.6.17 > On May 8, 2017, at 2:19 AM, Gaetano Volpe wrote: > > Hi guys, > > I'm running the last stable version of Freeswitch in my environment. > > FreeSWITCH Version 1.6.14~64bit ( 64bit) > > I've a very strange problem with a group of extensions of sales department > > An isdn pri line is provided by a cisco router (2921) through a sip trunk. > > When the calls volume increases, suddenly some phones experience one way audio (the customer hears, sellers don't). When it happesn the only way to get the phone back to work is to reboot it. > > We're running all polycom vvx 300 phones. > > What I've tried so far: > > 1. Change the codec to g.711 > 2. Upgrade freeswitch > 3. Upgrade Polycom firmware > 4. Downgrade Polycom firmware > > I've captured packets with wireshark packets and I can see both audio channels in the call trace. > > So I guessed it could be the phones but it's very strange. What do you think about it? > > Thanks > > Gaetano Volpe > CEO, CTO & Systems Analyst > g.volpe at studiofox.it - Mob.+39.3485260691 > > STUDIO FOX SERVICE di G.Volpe > Via Spineto, 24 - 70038 - Terlizzi (Bari) - Italy > Ph. +39.0803540043 - Fax +39.0803515004 > C.F. VLPGTN83E10L109S - P.Iva IT05803550721 > www.studiofox.it > La presente comunicazione ed i suoi eventuali allegati potrebbero contenere informazioni legate al segreto professionale o dati soggetti alla vigente Legge sulla Privacy e sono comunque strettamente riservati e confidenziali per il destinatario. In mancanza di autorizzazione ? vietata la lettura, la riproduzione, la diffusione e l'estrazione di copie. Pertanto, nel caso in cui aveste ricevuto per errore la presente comunicazione, Vi preghiamo di provvedere a distruggerla e di contattare immediatamente i nostri uffici ai numeri su indicati. Grazie. > > This communication and its attachments, if any, may contain information covered by professional secrecy or data subject to current privacy law, and they are intended only for the use of the individual to whom it is addressed. They may contain information that is legally privileged, confidential or exempt from disclosure. Reading, reproduction, disclosure, and extraction of copies are prohibited unless expressly authorized. If you have received this communication in error, please notify immediately our offices at the contact details specified above and delete the communication without retaining any copies. > Please note, no messages sent to this mailing list should be considered private. All messages to this list will be duplicated and distributed to the list and stored in the list archives. Private consulting services are available via consulting at freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170508/72079dc5/attachment-0001.html From mike at jerris.com Mon May 8 19:16:10 2017 From: mike at jerris.com (Michael Jerris) Date: Mon, 8 May 2017 11:16:10 -0400 Subject: [Freeswitch-users] Changing audio DTLS state from HANDSHAKE to FAIL issue In-Reply-To: <1493992618.S.41103.25639.f4-234-233.1494218754.24193@webmail.rediffmail.com> References: <1493992618.S.41103.25639.f4-234-233.1494218754.24193@webmail.rediffmail.com> Message-ID: <56E7108D-3C8C-42CE-8E43-29B3D2E2A68D@jerris.com> No it is not. It will require modifying the build, and current freeswitch code will not build against openssl 1.1, nor is it necessary. > On May 8, 2017, at 12:45 AM, sachin wrote: > > Hello Gregor, > > I have upgraded the openssl version to 1.1.0e from the following link. But the issue remains the same. > > http://slproweb.com/download/Win64OpenSSL-1_1_0e.exe > > ?Is the above way correct to upgrade the open ssl. > > ?Thanks and Regards, > ?Sachin > > From: Brian West > Sent: Fri, 05 May 2017 19:26:58 > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Changing audio DTLS state from HANDSHAKE to FAIL issue > > Clearly that isn't the case, I suspect a define is missing to actually enable ECDSA at this point. > > On Fri, May 5, 2017 at 8:47 AM, Gregor Nanger wrote: > Brian, isn't this solved in 1.6.16? > > FS-10037 [core] Update OpenSSL to version 1.0.2k for Windows > > 2017-05-05 15:26 GMT+02:00 Brian West : > OpenSSL on the windows build needs to be updated. > > https://freeswitch.org/jira/browse/FS-9510 > > On Thu, May 4, 2017 at 2:31 PM, Gregor Nanger wrote: > This error is familiar to me, I think so, if I remembered correctly. > > I am using windows. I had this error when client was Chrome. I just updated openssl dlls. I think that openssl library is updated on windows build now with version 1.6.17. And this version is also compiled on FS FTP to just download and install it. > > 2017-05-04 19:15 GMT+02:00 Anthony Minessale : > FS 1.5 sounds like a bad plan. > Try latest FS 1.6 or master. > > > On Thu, May 4, 2017 at 12:04 PM, ?talo Rossi wrote: > Old openssl version maybe? > Em qui, 4 de mai de 2017 ?s 11:19, sachin escreveu: > Hello All, > > ?I am trying to setup the Webrtc call between 2 SIPML5 client. I am using FreeSWITCH Version 1.6.14 on Windows. I am using internal LAN. > > The clients are getting registered over wss. I have created self signed certificates. In var.xml I have set the codecs setting as follows > > > > > ?I am able to establish the call and there is 2 way voice when I call from Sipmpl5 to Linphone and it works. But when I call from Linphone to Firefox (Simpl5) then the call is not getting established and I am getting following error > > 2017-05-04 11:29:04.543434 [ERR] switch_rtp.c:2907 audio Handshake failure 1 > 2017-05-04 11:29:04.543434 [INFO] switch_rtp.c:2908 Changing audio DTLS state from HANDSHAKE to FAIL > 2017-05-04 11:29:04.543434 [NOTICE] switch_rtp.c:2889 Hangup sofia/internal/sip:1001 at df7jal23ls0d.invalid [CS_EXCHANGE_MEDIA] [DESTINATION_OUT_OF_ORDER] > > My setup is as follows > > SIPML5 (FireFox browser 53.0)<-----> FS (v1.5)(Windows) <----> Linphone > > I am attaching the logs for the reference. > fs-logleve9.txt : Debug trace with loglevel =9 > ?fs-sip-trace.txt : Sip trace > Please let me know what could the issue and pointers to resolve the same. > > ?Thanks and Regards > SD > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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=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 > > -- > 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 > > -- > Brian West > brian at freeswitch.org > > Twitter: @FreeSWITCH , @briankwest > > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > Book a phone call (CST) > > Allison prompts for FreeSWITCH: > > https://www.gofundme.com/allison-prompts-for-freeswitch > Got Bugs? Report them here ! | Reddit: /r/freeswitch > T: 19184209001 | F: 19184209002 | M:+1918424WEST (9378) > Skype:briankwest > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 > > -- > Brian West > brian at freeswitch.org > > Twitter: @FreeSWITCH , @briankwest > > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > Book a phone call (CST) > > Allison prompts for FreeSWITCH: > > https://www.gofundme.com/allison-prompts-for-freeswitch > Got Bugs? Report them here ! | Reddit: /r/freeswitch > T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) > Skype:briankwest > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170508/448dd4f3/attachment-0001.html From adam.ben.ayoun1 at gmail.com Mon May 8 20:34:41 2017 From: adam.ben.ayoun1 at gmail.com (Adam Ben-Ayoun) Date: Mon, 8 May 2017 19:34:41 +0300 Subject: [Freeswitch-users] Trickle ICE using Verto Message-ID: Hi, We are currently using SIP for WebRTC for our app (web, iOS and Android implementations). ICE gathering can sometimes be slow especially on mobile and we are looking for ways to speed this up. We thought about using trickle ICE but it's not possible/easy using SIP/Freeswitch. My question is if we can do trickle ICE using verto. Thanks, Adam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170508/673f882c/attachment.html From anthony.minessale at gmail.com Mon May 8 21:41:55 2017 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 8 May 2017 12:41:55 -0500 Subject: [Freeswitch-users] Trickle ICE using Verto In-Reply-To: References: Message-ID: Its the same amount of work because they share the same central media stack. On Mon, May 8, 2017 at 11:34 AM, Adam Ben-Ayoun wrote: > Hi, > > We are currently using SIP for WebRTC for our app (web, iOS and Android > implementations). ICE gathering can sometimes be slow especially on mobile > and we are looking for ways to speed this up. We thought about using > trickle ICE but it's not possible/easy using SIP/Freeswitch. My question is > if we can do trickle ICE using verto. > > Thanks, > Adam > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170508/5eea725a/attachment.html From adam.ben.ayoun1 at gmail.com Mon May 8 21:58:35 2017 From: adam.ben.ayoun1 at gmail.com (Adam Ben-Ayoun) Date: Mon, 08 May 2017 17:58:35 +0000 Subject: [Freeswitch-users] Trickle ICE using Verto In-Reply-To: References: Message-ID: In what sense? Does vertical support trickle ice at the moment? On Mon, 8 May 2017 at 10:42 Anthony Minessale wrote: > Its the same amount of work because they share the same central media > stack. > > > On Mon, May 8, 2017 at 11:34 AM, Adam Ben-Ayoun > wrote: > >> Hi, >> >> We are currently using SIP for WebRTC for our app (web, iOS and Android >> implementations). ICE gathering can sometimes be slow especially on mobile >> and we are looking for ways to speed this up. We thought about using >> trickle ICE but it's not possible/easy using SIP/Freeswitch. My question is >> if we can do trickle ICE using verto. >> >> Thanks, >> Adam >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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=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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170508/592ba1b1/attachment.html From krice at freeswitch.org Mon May 8 22:06:10 2017 From: krice at freeswitch.org (Ken Rice) Date: Mon, 8 May 2017 13:06:10 -0500 Subject: [Freeswitch-users] Trickle ICE using Verto In-Reply-To: References: Message-ID: <142801d2c825$c67bc130$53734390$@freeswitch.org> Verto and SIP use the same centralized Media Stack? So no verto does not support trickle ICE, neither does SIP, implementing it for either SIP or Verto would be the same amount of effort? K From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Adam Ben-Ayoun Sent: Monday, May 8, 2017 12:59 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Trickle ICE using Verto In what sense? Does vertical support trickle ice at the moment? On Mon, 8 May 2017 at 10:42 Anthony Minessale > wrote: Its the same amount of work because they share the same central media stack. On Mon, May 8, 2017 at 11:34 AM, Adam Ben-Ayoun > wrote: Hi, We are currently using SIP for WebRTC for our app (web, iOS and Android implementations). ICE gathering can sometimes be slow especially on mobile and we are looking for ways to speed this up. We thought about using trickle ICE but it's not possible/easy using SIP/Freeswitch. My question is if we can do trickle ICE using verto. Thanks, Adam _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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=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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170508/ebafcea4/attachment-0001.html From ksh.sip at gmail.com Mon May 8 22:17:07 2017 From: ksh.sip at gmail.com (Gauri Kshirsagar) Date: Mon, 8 May 2017 23:47:07 +0530 Subject: [Freeswitch-users] video call using H264 Message-ID: Hi, I am new to freeswitch. I installed Freeswitch on CentOS 7 using rpm. I tried making video call using H264 but it failed. show codec on fs_cli does not list H264. What are the configurations required to make a video call ? Rgds, GK -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170508/e0afb5e7/attachment.html From mike at jerris.com Mon May 8 22:38:53 2017 From: mike at jerris.com (Michael Jerris) Date: Mon, 8 May 2017 14:38:53 -0400 Subject: [Freeswitch-users] video call using H264 In-Reply-To: References: Message-ID: Centos does not contain the libraries required for h264 support. Debian 8 does. > On May 8, 2017, at 2:17 PM, Gauri Kshirsagar wrote: > > Hi, > > I am new to freeswitch. I installed Freeswitch on CentOS 7 using rpm. I tried making video call using H264 but it failed. > > show codec on fs_cli does not list H264. What are the configurations required to make a video call ? From piotrek.gregor at gmail.com Mon May 8 23:53:30 2017 From: piotrek.gregor at gmail.com (Piotr Gregor) Date: Mon, 8 May 2017 20:53:30 +0100 Subject: [Freeswitch-users] Realtime kernels - CONFIG_PREEMPT_RT In-Reply-To: References: <6A32EAD3-FC96-46CB-BE70-0921F26CF70F@gmx.net> <0261E3CA-BD15-4F58-A137-0B5E0E4D2FD6@gmx.net> <590F31BA.5070304@telefaks.de> Message-ID: Hi Karl, RT kernels are not about speed, but about guaranteed biggest latency, i.e. bounded latency in the kernel. That highest latency gets significantly improved if RT is set up correctly (which is not so straightforward). In case of VoIP the most latency in playing audio stream comes from network congestion, dropouts, etc., therefore it is external to the kernel and most likely there is not that many degrees of freedom in this space which real time system could improve. Piotr -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170508/c348f0a6/attachment.html From anthony.minessale at gmail.com Tue May 9 00:17:51 2017 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 8 May 2017 15:17:51 -0500 Subject: [Freeswitch-users] Realtime kernels - CONFIG_PREEMPT_RT In-Reply-To: References: <6A32EAD3-FC96-46CB-BE70-0921F26CF70F@gmx.net> <0261E3CA-BD15-4F58-A137-0B5E0E4D2FD6@gmx.net> <590F31BA.5070304@telefaks.de> Message-ID: We do use realtime priorities on the scheduler, however, so some threads can be elevated to the least possible ignore time from the kernel. On Mon, May 8, 2017 at 2:53 PM, Piotr Gregor wrote: > Hi Karl, > > RT kernels are not about speed, but about guaranteed biggest latency, i.e. > bounded latency in the kernel. > That highest latency gets significantly improved if RT is set up correctly > (which is not so straightforward). > In case of VoIP the most latency in playing audio stream comes from > network congestion, dropouts, etc., > therefore it is external to the kernel and most likely there is not that > many degrees of freedom in this space > which real time system could improve. > > Piotr > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170508/de9c6fe0/attachment.html From gregor at infomedia.si Tue May 9 01:29:59 2017 From: gregor at infomedia.si (Gregor Nanger) Date: Mon, 8 May 2017 23:29:59 +0200 Subject: [Freeswitch-users] Changing audio DTLS state from HANDSHAKE to FAIL issue In-Reply-To: <1493992618.S.41103.25639.f4-234-233.1494218754.24193@webmail.rediffmail.com> References: <1493992618.S.41103.25639.f4-234-233.1494218754.24193@webmail.rediffmail.com> Message-ID: And you also copied dlls from openssl instalation to freeswitch root directory? 2017-05-08 6:45 GMT+02:00 sachin : > Hello Gregor, > > I have upgraded the openssl version to 1.1.0e from the following link. But > the issue remains the same. > > http://slproweb.com/download/Win64OpenSSL-1_1_0e.exe > > ?Is the above way correct to upgrade the open ssl. > > ?Thanks and Regards, > ?Sachin > > From: Brian West > Sent: Fri, 05 May 2017 19:26:58 > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Changing audio DTLS state from HANDSHAKE > to FAIL issue > > Clearly that isn't the case, I suspect a define is missing to actually > enable ECDSA at this point. > > On Fri, May 5, 2017 at 8:47 AM, Gregor Nanger > wrote: > >> Brian, isn't this solved in 1.6.16? >> >> >> - FS-10037 >> [core] >> Update OpenSSL to version 1.0.2k for Windows >> >> >> 2017-05-05 15:26 GMT+02:00 Brian West : >> >>> OpenSSL on the windows build needs to be updated. >>> >>> https://freeswitch.org/jira/browse/FS-9510 >>> >>> >>> On Thu, May 4, 2017 at 2:31 PM, Gregor Nanger >>> wrote: >>> >>>> This error is familiar to me, I think so, if I remembered correctly. >>>> >>>> I am using windows. I had this error when client was Chrome. I just >>>> updated openssl dlls. I think that openssl library is updated on windows >>>> build now with version 1.6.17. And this version is also compiled on FS FTP >>>> to just download and install it. >>>> >>>> 2017-05-04 19:15 GMT+02:00 Anthony Minessale < >>>> anthony.minessale at gmail.com>: >>>> >>>>> FS 1.5 sounds like a bad plan. >>>>> Try latest FS 1.6 or master. >>>>> >>>>> >>>>> On Thu, May 4, 2017 at 12:04 PM, ?talo Rossi >>>>> wrote: >>>>>> >>>>>> Old openssl version maybe? >>>>>> Em qui, 4 de mai de 2017 ?s 11:19, sachin >>>>>> escreveu: >>>>>> >>>>>>> Hello All, >>>>>>> >>>>>>> ?I am trying to setup the Webrtc call between 2 SIPML5 client. I am >>>>>>> using FreeSWITCH Version 1.6.14 on Windows. I am using internal LAN. >>>>>>> >>>>>>> The clients are getting registered over wss. I have created self >>>>>>> signed certificates. In var.xml I have set the codecs setting as follows >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> ?I am able to establish the call and there is 2 way voice when I >>>>>>> call from Sipmpl5 to Linphone and it works. But when I call from Linphone >>>>>>> to Firefox (Simpl5) then the call is not getting established and I am >>>>>>> getting following error >>>>>>> >>>>>>> 2017-05-04 11:29:04.543434 [ERR] switch_rtp.c:2907 audio Handshake >>>>>>> failure 1 >>>>>>> 2017-05-04 11:29:04.543434 [INFO] switch_rtp.c:2908 Changing audio >>>>>>> DTLS state from HANDSHAKE to FAIL >>>>>>> 2017-05-04 11:29:04.543434 [NOTICE] switch_rtp.c:2889 Hangup >>>>>>> sofia/internal/sip:1001 at df7jal23ls0d.invalid [CS_EXCHANGE_MEDIA] >>>>>>> [DESTINATION_OUT_OF_ORDER] >>>>>>> >>>>>>> My setup is as follows >>>>>>> >>>>>>> SIPML5 (FireFox browser 53.0)<-----> FS (v1.5)(Windows) <----> >>>>>>> Linphone >>>>>>> >>>>>>> I am attaching the logs for the reference. >>>>>>> fs-logleve9.txt : Debug trace with loglevel =9 >>>>>>> ?fs-sip-trace.txt : Sip trace >>>>>>> Please let me know what could the issue and pointers to resolve the >>>>>>> same. >>>>>>> >>>>>>> ?Thanks and Regards >>>>>>> SD >>>>>>> ____________________________________________________________ >>>>>>> _____________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://confluence.freeswitch.org >>>>>>> >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-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 >>>>> >>>>> >>>>> >>>>> -- >>>>> 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 >>>>> <+1%20919-386-9900==> >>>>> >>>>> 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/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 >>> >>> >>> >>> -- >>> >>> *Brian West* >>> brian at freeswitch.org >>> >>> *Twitter: @FreeSWITCH , @briankwest* >>> >>> http://www.freeswitchbook.com >>> http://www.freeswitchcookbook.com >>> >>> Book a phone call (CST) >>> >>> >>> Allison prompts for FreeSWITCH: >>> >>> *https://www.gofundme.com/allison-prompts-for-freeswitch* >>> >>> >>> Got Bugs? Report them here >>> ! >>> | Reddit: /r/freeswitch >>> >>> >>> *T:* 19184209001 <+1%20918-420-9001==> | *F:* 19184209002 >>> <+1%20918-420-9002==> | *M:*+1918424WEST (9378) >>> *Skype:*briankwest >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 > > > > -- > > *Brian West* > brian at freeswitch.org > > *Twitter: @FreeSWITCH , @briankwest* > > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > Book a phone call (CST) > > > Allison prompts for FreeSWITCH: > > *https://www.gofundme.com/allison-prompts-for-freeswitch* > > > Got Bugs? Report them here > ! > | Reddit: /r/freeswitch > > > *T:*+19184209001 <+1%20918-420-9001> | *F:*+19184209002 > <+1%20918-420-9002> | *M:*+1918424WEST (9378) > *Skype:*briankwest > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170508/08fccc80/attachment-0001.html From gregor at infomedia.si Tue May 9 01:33:39 2017 From: gregor at infomedia.si (Gregor Nanger) Date: Mon, 8 May 2017 23:33:39 +0200 Subject: [Freeswitch-users] Changing audio DTLS state from HANDSHAKE to FAIL issue In-Reply-To: References: <1493992618.S.41103.25639.f4-234-233.1494218754.24193@webmail.rediffmail.com> Message-ID: Overwrite libeay32.dll and ssleay32.dll from openssl installation directory to freeswitch root 2017-05-08 23:29 GMT+02:00 Gregor Nanger : > And you also copied dlls from openssl instalation to freeswitch root > directory? > > 2017-05-08 6:45 GMT+02:00 sachin : > >> Hello Gregor, >> >> I have upgraded the openssl version to 1.1.0e from the following link. >> But the issue remains the same. >> >> http://slproweb.com/download/Win64OpenSSL-1_1_0e.exe >> >> ?Is the above way correct to upgrade the open ssl. >> >> ?Thanks and Regards, >> ?Sachin >> >> From: Brian West >> Sent: Fri, 05 May 2017 19:26:58 >> To: FreeSWITCH Users Help >> Subject: Re: [Freeswitch-users] Changing audio DTLS state from HANDSHAKE >> to FAIL issue >> >> Clearly that isn't the case, I suspect a define is missing to actually >> enable ECDSA at this point. >> >> On Fri, May 5, 2017 at 8:47 AM, Gregor Nanger >> wrote: >> >>> Brian, isn't this solved in 1.6.16? >>> >>> >>> - FS-10037 >>> [core] >>> Update OpenSSL to version 1.0.2k for Windows >>> >>> >>> 2017-05-05 15:26 GMT+02:00 Brian West : >>> >>>> OpenSSL on the windows build needs to be updated. >>>> >>>> https://freeswitch.org/jira/browse/FS-9510 >>>> >>>> >>>> On Thu, May 4, 2017 at 2:31 PM, Gregor Nanger >>>> wrote: >>>> >>>>> This error is familiar to me, I think so, if I remembered correctly. >>>>> >>>>> I am using windows. I had this error when client was Chrome. I just >>>>> updated openssl dlls. I think that openssl library is updated on windows >>>>> build now with version 1.6.17. And this version is also compiled on FS FTP >>>>> to just download and install it. >>>>> >>>>> 2017-05-04 19:15 GMT+02:00 Anthony Minessale < >>>>> anthony.minessale at gmail.com>: >>>>> >>>>>> FS 1.5 sounds like a bad plan. >>>>>> Try latest FS 1.6 or master. >>>>>> >>>>>> >>>>>> On Thu, May 4, 2017 at 12:04 PM, ?talo Rossi >>>>>> wrote: >>>>>>> >>>>>>> Old openssl version maybe? >>>>>>> Em qui, 4 de mai de 2017 ?s 11:19, sachin >>>>>>> escreveu: >>>>>>> >>>>>>>> Hello All, >>>>>>>> >>>>>>>> ?I am trying to setup the Webrtc call between 2 SIPML5 client. I am >>>>>>>> using FreeSWITCH Version 1.6.14 on Windows. I am using internal LAN. >>>>>>>> >>>>>>>> The clients are getting registered over wss. I have created self >>>>>>>> signed certificates. In var.xml I have set the codecs setting as follows >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> ?I am able to establish the call and there is 2 way voice when I >>>>>>>> call from Sipmpl5 to Linphone and it works. But when I call from Linphone >>>>>>>> to Firefox (Simpl5) then the call is not getting established and I am >>>>>>>> getting following error >>>>>>>> >>>>>>>> 2017-05-04 11:29:04.543434 [ERR] switch_rtp.c:2907 audio Handshake >>>>>>>> failure 1 >>>>>>>> 2017-05-04 11:29:04.543434 [INFO] switch_rtp.c:2908 Changing audio >>>>>>>> DTLS state from HANDSHAKE to FAIL >>>>>>>> 2017-05-04 11:29:04.543434 [NOTICE] switch_rtp.c:2889 Hangup >>>>>>>> sofia/internal/sip:1001 at df7jal23ls0d.invalid [CS_EXCHANGE_MEDIA] >>>>>>>> [DESTINATION_OUT_OF_ORDER] >>>>>>>> >>>>>>>> My setup is as follows >>>>>>>> >>>>>>>> SIPML5 (FireFox browser 53.0)<-----> FS (v1.5)(Windows) <----> >>>>>>>> Linphone >>>>>>>> >>>>>>>> I am attaching the logs for the reference. >>>>>>>> fs-logleve9.txt : Debug trace with loglevel =9 >>>>>>>> ?fs-sip-trace.txt : Sip trace >>>>>>>> Please let me know what could the issue and pointers to resolve the >>>>>>>> same. >>>>>>>> >>>>>>>> ?Thanks and Regards >>>>>>>> SD >>>>>>>> ____________________________________________________________ >>>>>>>> _____________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting at freeswitch.org >>>>>>>> http://www.freeswitchsolutions.com >>>>>>>> >>>>>>>> >>>>>>>> Official FreeSWITCH Sites >>>>>>>> http://www.freeswitch.org >>>>>>>> http://confluence.freeswitch.org >>>>>>>> >>>>>>>> http://www.cluecon.com >>>>>>>> >>>>>>>> FreeSWITCH-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 >>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> 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 >>>>>> <+1%20919-386-9900==> >>>>>> >>>>>> 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/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 >>>> >>>> >>>> >>>> -- >>>> >>>> *Brian West* >>>> brian at freeswitch.org >>>> >>>> *Twitter: @FreeSWITCH , @briankwest* >>>> >>>> http://www.freeswitchbook.com >>>> http://www.freeswitchcookbook.com >>>> >>>> Book a phone call (CST) >>>> >>>> >>>> Allison prompts for FreeSWITCH: >>>> >>>> *https://www.gofundme.com/allison-prompts-for-freeswitch* >>>> >>>> >>>> Got Bugs? Report them here >>>> ! >>>> | Reddit: /r/freeswitch >>>> >>>> >>>> *T:* 19184209001 <+1%20918-420-9001==> | *F:* 19184209002 >>>> <+1%20918-420-9002==> | *M:*+1918424WEST (9378) >>>> *Skype:*briankwest >>>> >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-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 >> >> >> >> -- >> >> *Brian West* >> brian at freeswitch.org >> >> *Twitter: @FreeSWITCH , @briankwest* >> >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> Book a phone call (CST) >> >> >> Allison prompts for FreeSWITCH: >> >> *https://www.gofundme.com/allison-prompts-for-freeswitch* >> >> >> Got Bugs? Report them here >> ! >> | Reddit: /r/freeswitch >> >> >> *T:*+19184209001 <+1%20918-420-9001> | *F:*+19184209002 >> <+1%20918-420-9002> | *M:*+1918424WEST (9378) >> *Skype:*briankwest >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170508/bcb23358/attachment-0001.html From anthony.minessale at gmail.com Tue May 9 01:39:40 2017 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 8 May 2017 16:39:40 -0500 Subject: [Freeswitch-users] Noisy logs (switch_rtp.c: 1086 missed 1) In-Reply-To: References: Message-ID: Left over debug line. Update to latest code to remove it or comment it out in your local copy until the next time you update. On Mon, May 8, 2017 at 7:03 AM, Allan Kristensen wrote: > Hello, > > I'm getting thousand of these every hour on my Freeswitch log: > > May 8 08:09:32 sip1b media-switch-container[18466]: 2017-05-08 > 08:09:24.807956 [WARNING] switch_rtp.c:1086 missed 1 > May 8 08:09:32 sip1b media-switch-container[18466]: 2017-05-08 > 08:09:25.787957 [WARNING] switch_rtp.c:1086 missed 1 > : > > I can see it's related to calls coming from an Asterisk server but the > INVITE's look fine to me and calls are active. > The customer says there are no problems, but it's making my log monitor > unhappy (they are warnings after all). > > I've looked into the code and I can see it's something to do with handling > of ICE, but why it appears is a little mysterious to me. > (I using "apply-candidate-acl = wan.auto", so nothing fancy there). > > Any clues? > > /Allan > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170508/55e712a7/attachment.html From mike at jerris.com Tue May 9 01:48:18 2017 From: mike at jerris.com (Michael Jerris) Date: Mon, 8 May 2017 17:48:18 -0400 Subject: [Freeswitch-users] Changing audio DTLS state from HANDSHAKE to FAIL issue In-Reply-To: References: <1493992618.S.41103.25639.f4-234-233.1494218754.24193@webmail.rediffmail.com> Message-ID: This is not enough, and will not work for sure. Updating to 1.1 is more involved, and will require code changes to freeswitch code as well. What you are suggesting here will almost for sure cause things to crash. Furthermore, 1.0.2 is more than enough to address this issue. I?ve put some comments on the Jira related to this of the details and testing i need done, we can continue the conversation there. I just want to be completely clear, updating to 1.1 as described here should not be done, and will only cause more issues. Lets continue this conversation on the Jira for those interested. https://freeswitch.org/jira/browse/FS-9510 Mike > On May 8, 2017, at 5:33 PM, Gregor Nanger wrote: > > Overwrite libeay32.dll and ssleay32.dll from openssl installation directory to freeswitch root > > 2017-05-08 23:29 GMT+02:00 Gregor Nanger >: > And you also copied dlls from openssl instalation to freeswitch root directory? > > 2017-05-08 6:45 GMT+02:00 sachin >: > Hello Gregor, > > I have upgraded the openssl version to 1.1.0e from the following link. But the issue remains the same. > > http://slproweb.com/download/Win64OpenSSL-1_1_0e.exe > > ?Is the above way correct to upgrade the open ssl. > > ?Thanks and Regards, > ?Sachin > > From: Brian West > > Sent: Fri, 05 May 2017 19:26:58 > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] Changing audio DTLS state from HANDSHAKE to FAIL issue > > Clearly that isn't the case, I suspect a define is missing to actually enable ECDSA at this point. > > On Fri, May 5, 2017 at 8:47 AM, Gregor Nanger > wrote: > Brian, isn't this solved in 1.6.16? > > FS-10037 [core] Update OpenSSL to version 1.0.2k for Windows > > 2017-05-05 15:26 GMT+02:00 Brian West >: > OpenSSL on the windows build needs to be updated. > > https://freeswitch.org/jira/browse/FS-9510 > > On Thu, May 4, 2017 at 2:31 PM, Gregor Nanger > wrote: > This error is familiar to me, I think so, if I remembered correctly. > > I am using windows. I had this error when client was Chrome. I just updated openssl dlls. I think that openssl library is updated on windows build now with version 1.6.17. And this version is also compiled on FS FTP to just download and install it. > > 2017-05-04 19:15 GMT+02:00 Anthony Minessale >: > FS 1.5 sounds like a bad plan. > Try latest FS 1.6 or master. > > > On Thu, May 4, 2017 at 12:04 PM, ?talo Rossi > wrote: > Old openssl version maybe? > Em qui, 4 de mai de 2017 ?s 11:19, sachin > escreveu: > Hello All, > > ?I am trying to setup the Webrtc call between 2 SIPML5 client. I am using FreeSWITCH Version 1.6.14 on Windows. I am using internal LAN. > > The clients are getting registered over wss. I have created self signed certificates. In var.xml I have set the codecs setting as follows > > > > > ?I am able to establish the call and there is 2 way voice when I call from Sipmpl5 to Linphone and it works. But when I call from Linphone to Firefox (Simpl5) then the call is not getting established and I am getting following error > > 2017-05-04 11:29:04.543434 [ERR] switch_rtp.c:2907 audio Handshake failure 1 > 2017-05-04 11:29:04.543434 [INFO] switch_rtp.c:2908 Changing audio DTLS state from HANDSHAKE to FAIL > 2017-05-04 11:29:04.543434 [NOTICE] switch_rtp.c:2889 Hangup sofia/internal/sip:1001 at df7jal 23ls0d.invalid [CS_EXCHANGE_MEDIA] [DESTINATION_OUT_OF_ORDER] > > My setup is as follows > > SIPML5 (FireFox browser 53.0)<-----> FS (v1.5)(Windows) <----> Linphone > > I am attaching the logs for the reference. > fs-logleve9.txt : Debug trace with loglevel =9 > ?fs-sip-trace.txt : Sip trace > Please let me know what could the issue and pointers to resolve the same. > > ?Thanks and Regards > SD -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170508/695d6e0f/attachment-0001.html From gregor at infomedia.si Tue May 9 01:59:21 2017 From: gregor at infomedia.si (Gregor Nanger) Date: Mon, 8 May 2017 23:59:21 +0200 Subject: [Freeswitch-users] Changing audio DTLS state from HANDSHAKE to FAIL issue In-Reply-To: References: <1493992618.S.41103.25639.f4-234-233.1494218754.24193@webmail.rediffmail.com> Message-ID: I belive you are right, Michael. Guess that moving to 1.1 is more complicated. I had same error copied dlls of 1.02g to freeswitch root and it worked. But as you said, 1.1 needs internal changes in FS? Will continue conversation on jira, if I can be of any help. Sachin, just suggesting to try to install openssl 1.02g and copy dlls. Maybe it will not work for you , but you have nothing to lose. 2017-05-08 23:48 GMT+02:00 Michael Jerris : > This is not enough, and will not work for sure. Updating to 1.1 is more > involved, and will require code changes to freeswitch code as well. What > you are suggesting here will almost for sure cause things to crash. > Furthermore, 1.0.2 is more than enough to address this issue. I?ve put > some comments on the Jira related to this of the details and testing i need > done, we can continue the conversation there. I just want to be completely > clear, updating to 1.1 as described here should not be done, and will only > cause more issues. Lets continue this conversation on the Jira for those > interested. > > https://freeswitch.org/jira/browse/FS-9510 > > Mike > > On May 8, 2017, at 5:33 PM, Gregor Nanger wrote: > > Overwrite libeay32.dll and ssleay32.dll from openssl installation > directory to freeswitch root > > 2017-05-08 23:29 GMT+02:00 Gregor Nanger : > >> And you also copied dlls from openssl instalation to freeswitch root >> directory? >> >> 2017-05-08 6:45 GMT+02:00 sachin : >> >>> Hello Gregor, >>> >>> I have upgraded the openssl version to 1.1.0e from the following link. >>> But the issue remains the same. >>> >>> http://slproweb.com/download/Win64OpenSSL-1_1_0e.exe >>> >>> ?Is the above way correct to upgrade the open ssl. >>> >>> ?Thanks and Regards, >>> ?Sachin >>> >>> From: Brian West >>> Sent: Fri, 05 May 2017 19:26:58 >>> To: FreeSWITCH Users Help >>> Subject: Re: [Freeswitch-users] Changing audio DTLS state from HANDSHAKE >>> to FAIL issue >>> >>> Clearly that isn't the case, I suspect a define is missing to actually >>> enable ECDSA at this point. >>> >>> On Fri, May 5, 2017 at 8:47 AM, Gregor Nanger w >>> rote: >>> >>>> Brian, isn't this solved in 1.6.16? >>>> >>>> >>>> - FS-10037 >>>> [core] >>>> Update OpenSSL to version 1.0.2k for Windows >>>> >>>> >>>> 2017-05-05 15:26 GMT+02:00 Brian West : >>>> >>>>> OpenSSL on the windows build needs to be updated. >>>>> >>>>> https://freeswitch.org/jira/browse/FS-9510 >>>>> >>>>> >>>>> On Thu, May 4, 2017 at 2:31 PM, Gregor Nanger w >>>>> rote: >>>>> >>>>>> This error is familiar to me, I think so, if I remembered correctly. >>>>>> >>>>>> I am using windows. I had this error when client was Chrome. I just >>>>>> updated openssl dlls. I think that openssl library is updated on windows >>>>>> build now with version 1.6.17. And this version is also compiled on FS FTP >>>>>> to just download and install it. >>>>>> >>>>>> 2017-05-04 19:15 GMT+02:00 Anthony Minessale >>>>> gmail.com>: >>>>>> >>>>>>> FS 1.5 sounds like a bad plan. >>>>>>> Try latest FS 1.6 or master. >>>>>>> >>>>>>> >>>>>>> On Thu, May 4, 2017 at 12:04 PM, ?talo Rossi >>>>>>> wrote: >>>>>>>> >>>>>>>> Old openssl version maybe? >>>>>>>> Em qui, 4 de mai de 2017 ?s 11:19, sachin >>>>>>>> escreveu: >>>>>>>> >>>>>>>>> Hello All, >>>>>>>>> >>>>>>>>> ?I am trying to setup the Webrtc call between 2 SIPML5 client. I >>>>>>>>> am using FreeSWITCH Version 1.6.14 on Windows. I am using internal LAN. >>>>>>>>> >>>>>>>>> The clients are getting registered over wss. I have created self >>>>>>>>> signed certificates. In var.xml I have set the codecs setting as follows >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> ?I am able to establish the call and there is 2 way voice when I >>>>>>>>> call from Sipmpl5 to Linphone and it works. But when I call from Linphone >>>>>>>>> to Firefox (Simpl5) then the call is not getting established and I am >>>>>>>>> getting following error >>>>>>>>> >>>>>>>>> 2017-05-04 11:29:04.543434 [ERR] switch_rtp.c:2907 audio Handshake >>>>>>>>> failure 1 >>>>>>>>> 2017-05-04 11:29:04.543434 [INFO] switch_rtp.c:2908 Changing audio >>>>>>>>> DTLS state from HANDSHAKE to FAIL >>>>>>>>> 2017-05-04 11:29:04.543434 [NOTICE] switch_rtp.c:2889 Hangup >>>>>>>>> sofia/internal/sip:1001 at df7jal23ls0d.invalid [CS_EXCHANGE_MEDIA] >>>>>>>>> [DESTINATION_OUT_OF_ORDER] >>>>>>>>> >>>>>>>>> My setup is as follows >>>>>>>>> >>>>>>>>> SIPML5 (FireFox browser 53.0)<-----> FS (v1.5)(Windows) <----> >>>>>>>>> Linphone >>>>>>>>> >>>>>>>>> I am attaching the logs for the reference. >>>>>>>>> fs-logleve9.txt : Debug trace with loglevel =9 >>>>>>>>> ?fs-sip-trace.txt : Sip trace >>>>>>>>> Please let me know what could the issue and pointers to resolve >>>>>>>>> the same. >>>>>>>>> >>>>>>>>> ?Thanks and Regards >>>>>>>>> SD >>>>>>>>> >>>>>>>> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170508/c0aefbbf/attachment.html From mike at jerris.com Tue May 9 02:57:07 2017 From: mike at jerris.com (Michael Jerris) Date: Mon, 8 May 2017 18:57:07 -0400 Subject: [Freeswitch-users] Changing audio DTLS state from HANDSHAKE to FAIL issue In-Reply-To: References: <1493992618.S.41103.25639.f4-234-233.1494218754.24193@webmail.rediffmail.com> Message-ID: <7F42D59E-35EA-460F-97E5-5A752309B5F6@jerris.com> DO NOT copy DLL?s around. This will break things. We can continue this conversation on Jira. > On May 8, 2017, at 5:59 PM, Gregor Nanger wrote: > > I belive you are right, Michael. Guess that moving to 1.1 is more complicated. I had same error copied dlls of 1.02g to freeswitch root and it worked. But as you said, 1.1 needs internal changes in FS? Will continue conversation on jira, if I can be of any help. > > Sachin, just suggesting to try to install openssl 1.02g and copy dlls. Maybe it will not work for you , but you have nothing to lose. > > 2017-05-08 23:48 GMT+02:00 Michael Jerris >: > This is not enough, and will not work for sure. Updating to 1.1 is more involved, and will require code changes to freeswitch code as well. What you are suggesting here will almost for sure cause things to crash. Furthermore, 1.0.2 is more than enough to address this issue. I?ve put some comments on the Jira related to this of the details and testing i need done, we can continue the conversation there. I just want to be completely clear, updating to 1.1 as described here should not be done, and will only cause more issues. Lets continue this conversation on the Jira for those interested. > > https://freeswitch.org/jira/browse/FS-9510 > > Mike > >> On May 8, 2017, at 5:33 PM, Gregor Nanger > wrote: >> >> Overwrite libeay32.dll and ssleay32.dll from openssl installation directory to freeswitch root >> >> 2017-05-08 23:29 GMT+02:00 Gregor Nanger >: >> And you also copied dlls from openssl instalation to freeswitch root directory? >> >> 2017-05-08 6:45 GMT+02:00 sachin >: >> Hello Gregor, >> >> I have upgraded the openssl version to 1.1.0e from the following link. But the issue remains the same. >> >> http://slproweb.com/download/Win64OpenSSL-1_1_0e.exe >> >> ?Is the above way correct to upgrade the open ssl. >> >> ?Thanks and Regards, >> ?Sachin >> >> From: Brian West > >> Sent: Fri, 05 May 2017 19:26:58 >> To: FreeSWITCH Users Help > >> Subject: Re: [Freeswitch-users] Changing audio DTLS state from HANDSHAKE to FAIL issue >> >> Clearly that isn't the case, I suspect a define is missing to actually enable ECDSA at this point. >> >> On Fri, May 5, 2017 at 8:47 AM, Gregor Nanger > wrote: >> Brian, isn't this solved in 1.6.16? >> >> FS-10037 [core] Update OpenSSL to version 1.0.2k for Windows >> >> 2017-05-05 15:26 GMT+02:00 Brian West >: >> OpenSSL on the windows build needs to be updated. >> >> https://freeswitch.org/jira/browse/FS-9510 >> >> On Thu, May 4, 2017 at 2:31 PM, Gregor Nanger > wrote: >> This error is familiar to me, I think so, if I remembered correctly. >> >> I am using windows. I had this error when client was Chrome. I just updated openssl dlls. I think that openssl library is updated on windows build now with version 1.6.17. And this version is also compiled on FS FTP to just download and install it. >> >> 2017-05-04 19:15 GMT+02:00 Anthony Minessale >: >> FS 1.5 sounds like a bad plan. >> Try latest FS 1.6 or master. >> >> >> On Thu, May 4, 2017 at 12:04 PM, ?talo Rossi > wrote: >> Old openssl version maybe? >> Em qui, 4 de mai de 2017 ?s 11:19, sachin > escreveu: >> Hello All, >> >> ?I am trying to setup the Webrtc call between 2 SIPML5 client. I am using FreeSWITCH Version 1.6.14 on Windows. I am using internal LAN. >> >> The clients are getting registered over wss. I have created self signed certificates. In var.xml I have set the codecs setting as follows >> >> >> >> >> ?I am able to establish the call and there is 2 way voice when I call from Sipmpl5 to Linphone and it works. But when I call from Linphone to Firefox (Simpl5) then the call is not getting established and I am getting following error >> >> 2017-05-04 11:29:04.543434 [ERR] switch_rtp.c:2907 audio Handshake failure 1 >> 2017-05-04 11:29:04.543434 [INFO] switch_rtp.c:2908 Changing audio DTLS state from HANDSHAKE to FAIL >> 2017-05-04 11:29:04.543434 [NOTICE] switch_rtp.c:2889 Hangup sofia/internal/sip:1001 at df7jal <>23ls0d.invalid <> [CS_EXCHANGE_MEDIA] [DESTINATION_OUT_OF_ORDER] >> >> My setup is as follows >> >> SIPML5 (FireFox browser 53.0)<-----> FS (v1.5)(Windows) <----> Linphone >> >> I am attaching the logs for the reference. >> fs-logleve9.txt : Debug trace with loglevel =9 >> ?fs-sip-trace.txt : Sip trace >> Please let me know what could the issue and pointers to resolve the same. >> >> ?Thanks and Regards >> SD > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170508/d9b79351/attachment-0001.html From ak at hejdu.dk Tue May 9 11:48:28 2017 From: ak at hejdu.dk (Allan Kristensen) Date: Tue, 09 May 2017 07:48:28 +0000 Subject: [Freeswitch-users] Noisy logs (switch_rtp.c: 1086 missed 1) In-Reply-To: References: Message-ID: Hello Anthony, Thanks, it will save us all some disk space ;-) /Allan On Mon, May 8, 2017 at 11:44 PM Anthony Minessale < anthony.minessale at gmail.com> wrote: > Left over debug line. > Update to latest code to remove it or comment it out in your local copy > until the next time you update. > > > On Mon, May 8, 2017 at 7:03 AM, Allan Kristensen wrote: > >> Hello, >> >> I'm getting thousand of these every hour on my Freeswitch log: >> >> May 8 08:09:32 sip1b media-switch-container[18466]: 2017-05-08 >> 08:09:24.807956 [WARNING] switch_rtp.c:1086 missed 1 >> May 8 08:09:32 sip1b media-switch-container[18466]: 2017-05-08 >> 08:09:25.787957 [WARNING] switch_rtp.c:1086 missed 1 >> : >> >> I can see it's related to calls coming from an Asterisk server but the >> INVITE's look fine to me and calls are active. >> The customer says there are no problems, but it's making my log monitor >> unhappy (they are warnings after all). >> >> I've looked into the code and I can see it's something to do with >> handling of ICE, but why it appears is a little mysterious to me. >> (I using "apply-candidate-acl = wan.auto", so nothing fancy there). >> >> Any clues? >> >> /Allan >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 <(919)%20386-9900> > > 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170509/f501751b/attachment.html From bogdan at opensips.org Tue May 9 17:18:43 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 9 May 2017 16:18:43 +0300 Subject: [Freeswitch-users] [OpenSIPS-Users] TLS SIP packet tracing and visualization In-Reply-To: References: Message-ID: Thank you Giovanni, that is a useful tool - we will document it in the OpenSIPS TLS tutorial, so other can benefit ;) Many thanks, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit May 2017 Amsterdam http://www.opensips.org/events/Summit-2017Amsterdam.html On 05/02/2017 05:52 PM, Giovanni Maruzzelli wrote: > For a cut and paste ready version, that has the correct carriage > returns (mangled by mail), check it in FreeSWITCH documentation: > > https://freeswitch.org/confluence/display/FREESWITCH/Packet+Capture#PacketCapture-TLSwithsharka > > -giovanni > > On 2 May 2017 at 16:26, Giovanni Maruzzelli > wrote: > > Hello fellows, > > after some experimentation with various tools, I come out with a > little shell tool that maybe can be useful to you too. > > It can only work with non-forward secrecy ciphers, obviously, and > only if is started before the client do the initial TLS handshake > (eg, just restart the client). Forward secrecy cannot be decrypted > after fact, so don't waste effort. > > An example of ciphers that can be decrypted are the "AES256-SHA" > openssl cipher group. You can use ssldump to check what cipher is > used by serverhello. > > Enjoy, make it better, and share it :) > > > #!/bin/bash > # brought to you by Giovanni Maruzzelli > # > SERVERIP="192.168.1.150" > SERVERPORT="5061" > PRIVKEY="/etc/certs/privkey.pem" > STDERR2DEVNULL=" 2>/dev/null " > REGEX="notyet" > > if [ -z "$1" ]; then > REGEX="\\\.*" > else > REGEX="$1" > fi > FILTER="ssl.app_data and sip matches" > FILTER2="$FILTER \"$REGEX\"" > FILTER3="'$FILTER2'" > ARGUMENT="-i 1 -Y $FILTER3 -E header=y -T fields -e frame.number > -e frame.time -e frame.time_delta_displayed -e ip.src -e ip.dst -e > sip.Status-Line -e sip.Request-Line -e sip.msg_hdr -l -d > tcp.port\=\=5061,sip -o \"ssl.keys_list: > $SERVERIP,$SERVERPORT,sip,$PRIVKEY\" $STDERR2DEVNULL | sed -u > 's/\t/\n/g' | sed -u '/^$/d' | sed -u > 's/^[0-9]*$/\n==&==============================/g'" > > echo "" > echo "NB: if it do not works, edit script so that > STDERR2DEVNULL=\" \" and try again" > echo "" > echo "NB: remember to quote and escape match patterns, using > triple slash" > echo " eg, for matching 1010 at pbx.example.com > , use \"1010 at pbx.example.com > \"" > echo " eg, for matching anything, use \"\\\\\\.*\"" > echo " eg, for matching *98, use \"\\\\\\*98\"" > echo "USAGE: $0 \"\\\\\\*98 at pbx.example.com > \"" > echo "" > > > case "$1" in > -help|--help|?) > exit 0 > ;; > *) > echo "THIS TIME WE'RE DOING:" > echo "tshark $ARGUMENT" > echo "" > bash -c "tshark $ARGUMENT" > ;; > esac > > > > -- > > Sincerely, > > Giovanni Maruzzelli > OpenTelecom.IT > cell: +39 347 266 56 18 > > > > > -- > > Sincerely, > > Giovanni Maruzzelli > OpenTelecom.IT > cell: +39 347 266 56 18 > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170509/435e9ca6/attachment.html From virtualguard2015 at gmail.com Tue May 9 05:21:29 2017 From: virtualguard2015 at gmail.com (Ravi sanyal) Date: Tue, 9 May 2017 13:21:29 +1200 Subject: [Freeswitch-users] [ERR] mod_sofia.c:2491 CODEC NEGOTIATION ERROR. Message-ID: I'm sorry for pasting the whole code in the email. Here's a much more succinct version. pastebin of code is included below. I'm new to using freeswitch and need to set up so that i can use a webrtc data channel to connect two clients together. I'm using SIPJS. I followed the guide to setting up webrtc on freeswitch end, however to begin with it gave a 488 error and INCOMPATIBLE_DESTINATION. Finding out about this led me to codec negotation (the logs showed that when sending the invite from client A produced a CODEC NEGOTATION ERROR) Then I followed a codec negotation guide on the confluence page, however I'm new to freeswitch. The guide talked about outbound and inbound, however it then talked about a profile for outbound. I have an internal and external profile, and was setting the variables there. Having set inbound-late-negotation and inbound-zrtp-passthru and not finding the others in the profile, i restarted freeswitch to get this error: (on both internal and external profile) i'll give you the whole sip trace as it seems important. It would also be helpful to know what the internal profile is doing in terms of webrtc. the guide says that internal is only used internally. As in the local network. Im not connected to freeswitch locally. what is the purpose of the internal profile then? It was at points getting 503 service unavailable but it's gone back to 488. I'm not sure what the problem is. https://pastebin.freeswitch.org/view/ca875188 Nathan -- *Virtual Guard Ltd* *info at virtualguard.co.nz * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170509/41dec246/attachment-0001.html From anibal.carvalho at gmail.com Tue May 9 17:57:43 2017 From: anibal.carvalho at gmail.com (=?UTF-8?Q?An=C3=ADbal_C=C3=A9sar_Aguiar_de_Carvalho?=) Date: Tue, 9 May 2017 10:57:43 -0300 Subject: [Freeswitch-users] No events after a hangup Message-ID: Hi everyone. I have a scenario with the following flow: - Freeswitch is configured to forward connections to a determined number to a localhost port and my software is connected at this socket. - A call is made by a first user to this number. - This call is connected and answered by our software. - A second channel is originated to a second user, with the origination_caller_id_number of the first user. At this point if the second user answers or rejects the call (manually or due to a connection timeout) then the originate event, answer or hangup and others events are sent by Freeswitch, If the first user hangs up, no event is sent whatsoever. Is this how Freeswitch is supposed to work, or is it a bug? 'event plain ALL' is used at the beginning, so it is not a filter issue from my side. Any help is appreciated. Thanks, -- An?bal Carvalho. anibal.carvalho at gmail.com anibal.carvalho at vulcanet.com.br -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170509/1d6c1d54/attachment.html From gmaruzz at gmail.com Tue May 9 19:10:39 2017 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Tue, 9 May 2017 17:10:39 +0200 Subject: [Freeswitch-users] [OpenSIPS-Users] TLS SIP packet tracing and visualization In-Reply-To: References: Message-ID: On 9 May 2017 at 15:18, Bogdan-Andrei Iancu wrote: > Thank you Giovanni, that is a useful tool - we will document it in the > OpenSIPS TLS tutorial, so other can benefit ;) > > Glad about it! Be sure to get it from https://freeswitch.org/confluence/display/FREESWITCH/ Packet+Capture#PacketCapture-TLSwithsharka , is the latest version with a couple fixes. -giovanni > Many thanks, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > OpenSIPS Summit May 2017 Amsterdam > http://www.opensips.org/events/Summit-2017Amsterdam.html > > On 05/02/2017 05:52 PM, Giovanni Maruzzelli wrote: > > For a cut and paste ready version, that has the correct carriage returns > (mangled by mail), check it in FreeSWITCH documentation: > > https://freeswitch.org/confluence/display/FREESWITCH/ > Packet+Capture#PacketCapture-TLSwithsharka > > -giovanni > > On 2 May 2017 at 16:26, Giovanni Maruzzelli wrote: > >> Hello fellows, >> >> after some experimentation with various tools, I come out with a little >> shell tool that maybe can be useful to you too. >> >> It can only work with non-forward secrecy ciphers, obviously, and only if >> is started before the client do the initial TLS handshake (eg, just restart >> the client). Forward secrecy cannot be decrypted after fact, so don't waste >> effort. >> >> An example of ciphers that can be decrypted are the "AES256-SHA" openssl >> cipher group. You can use ssldump to check what cipher is used by >> serverhello. >> >> Enjoy, make it better, and share it :) >> >> >> #!/bin/bash >> # brought to you by Giovanni Maruzzelli >> # >> SERVERIP="192.168.1.150" >> SERVERPORT="5061" >> PRIVKEY="/etc/certs/privkey.pem" >> STDERR2DEVNULL=" 2>/dev/null " >> REGEX="notyet" >> >> if [ -z "$1" ]; then >> REGEX="\\\.*" >> else >> REGEX="$1" >> fi >> FILTER="ssl.app_data and sip matches" >> FILTER2="$FILTER \"$REGEX\"" >> FILTER3="'$FILTER2'" >> ARGUMENT="-i 1 -Y $FILTER3 -E header=y -T fields -e frame.number -e >> frame.time -e frame.time_delta_displayed -e ip.src -e ip.dst -e >> sip.Status-Line -e sip.Request-Line -e sip.msg_hdr -l -d >> tcp.port\=\=5061,sip -o \"ssl.keys_list: $SERVERIP,$SERVERPORT,sip,$PRIVKEY\" >> $STDERR2DEVNULL | sed -u 's/\t/\n/g' | sed -u '/^$/d' | sed -u >> 's/^[0-9]*$/\n==&==============================/g'" >> >> echo "" >> echo "NB: if it do not works, edit script so that STDERR2DEVNULL=\" \" >> and try again" >> echo "" >> echo "NB: remember to quote and escape match patterns, using triple slash" >> echo " eg, for matching 1010 at pbx.example.com, use \" >> 1010 at pbx.example.com\"" >> echo " eg, for matching anything, use \"\\\\\\.*\"" >> echo " eg, for matching *98, use \"\\\\\\*98\"" >> echo "USAGE: $0 \"\\\\\\*98 at pbx.example.com\"" >> echo "" >> >> >> case "$1" in >> -help|--help|?) >> exit 0 >> ;; >> *) >> echo "THIS TIME WE'RE DOING:" >> echo "tshark $ARGUMENT" >> echo "" >> bash -c "tshark $ARGUMENT" >> ;; >> esac >> >> >> >> -- >> >> Sincerely, >> >> Giovanni Maruzzelli >> OpenTelecom.IT >> cell: +39 347 266 56 18 >> > > > > -- > > Sincerely, > > Giovanni Maruzzelli > OpenTelecom.IT > cell: +39 347 266 56 18 > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170509/4e706831/attachment.html From lesley.pervis at gmail.com Tue May 9 22:39:25 2017 From: lesley.pervis at gmail.com (Lesley Pervis) Date: Tue, 9 May 2017 12:39:25 -0600 Subject: [Freeswitch-users] Python-ESL: ESLconnection.execute() blocked when handling channel_park under thread In-Reply-To: References: Message-ID: 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. I looked at doing something similar in Python in the 1.2 days and after a couple of roadblocks like this, I took the advice of the wiki and the developers and developed what I needed in Lua. It worked out very well. Python gets some love in Confluence, so I assume people use it, though. I know for a fact that you can start a Lua event listener by executing a luarun command via Python ESL API. Maybe it would also work in this case to run a Python script in the interpreter managed by mod_python. Let FS manage thread state. On Tue, Apr 11, 2017 at 1:03 AM, Mohd Kamal Bin Mustafa < kamal.mustafa at gmail.com> wrote: > I have this python code:- > > import time > import json > import logging > import threading > > from freeswitchESL import ESL > > class EventThread(threading.Thread): > def __init__(self, target, args): > super(EventThread, self).__init__() > self.target = target > self.args = args > > def run(self): > print("Running target") > self.target(*self.args) > > def on_channel_park(e): > uuid = e.getHeader('Unique-ID') > print("Ringing") > conn.execute('ring_ready ', '', uuid) > conn.execute('sleep', '5500', uuid) > conn.execute('answer', '', uuid) > print("answered, playing media ...") > media_url = 'http://www.noiseaddicts.com/samples_1w72b820/17.mp3' > resp = conn.execute("playback", media_url, uuid) > print(resp.getHeader("Reply-Text")) > print("playback done") > print(resp) > > if __name__ == '__main__': > conn = ESL.ESLconnection('127.0.0.1', '8021', 'mypass') > conn.events('plain', 'all') > while True: > e = conn.recvEvent() > uuid = e.getHeader('Unique-ID') > events_message = json.loads(e.serialize('json')) > event_name = events_message['Event-Name'].lower() > method_name = 'on_%s' % event_name > print(method_name) > > if event_name == 'channel_park': > print("spawning thread") > t = EventThread(on_channel_park, args=(e,)) > t.start() > > if event_name == 'playback_stop': > conn.execute("hangup", "", uuid) > print("Hanging up") > > Above, the code will stuck at conn.execute('ring_ready ', '', uuid) in > on_channel_park(). It never went pass through until I hangup the call. > In freeswitch log, it stopped at executing park(). > > But if I moved ring_ready, sleep, answer outside of the thread it will > work, i.e I can hear a playback:- > > def on_channel_park(e): > uuid = e.getHeader('Unique-ID') > print("answered, playing media ...") > media_url = 'http://www.noiseaddicts.com/samples_1w72b820/17.mp3' > resp = conn.execute("playback", media_url, uuid) > print(resp.getHeader("Reply-Text")) > print("playback done") > print(resp) > > if __name__ == '__main__': > conn = ESL.ESLconnection('127.0.0.1', '8021', 'mypass') > conn.events('plain', 'all') > while True: > e = conn.recvEvent() > uuid = e.getHeader('Unique-ID') > events_message = json.loads(e.serialize('json')) > event_name = events_message['Event-Name'].lower() > method_name = 'on_%s' % event_name > print(method_name) > > if event_name == 'channel_park': > print("Ringing") > conn.execute('ring_ready ', '', uuid) > conn.execute('sleep', '5500', uuid) > conn.execute('answer', '', uuid) > print("spawning thread") > t = EventThread(on_channel_park, args=(e,)) > t.start() > > Any idea why this happen ? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170509/721a49d0/attachment-0001.html From zrothy at monmouth.com Wed May 10 01:26:38 2017 From: zrothy at monmouth.com (Zach Rothy) Date: Tue, 09 May 2017 17:26:38 -0400 Subject: [Freeswitch-users] Audio Bridge to Asterisk from Freeswitch video conference Message-ID: Good Afternoon, I currently have a video conference which users connect to via the Verto library. I was looking to enhance this conference by also allowing users to join audio-wise to the video conference by making a bridge/call to my asterisk server which hosts audio conference rooms. Currently I have the conference_set_auto_outcall making the call via sip to the asterisk server, but ran into a couple of issues after asterisk and the freeswitch servers establish the call. The first issue, though it may be on purpose, is that when I use the ghost flag for the member it seems RTP is never delivered to my verto endpoint from the sip call to the asterisk as well as the other way around. Though I clearly see freeswitch is recognizing energy detection from my asterisk conference, as I see the talking flag appear but no audio is heard from it on the verto client. It is the same the other way around, where I will talk via a mic on my verto client, but never hear it on the asterisk side of things. Though for asterisk it does not even seem to be sending audio to the asterisk server. I'd prefer to use the ghost flag as I do with screen sharing so that the audio bridge does not count as a user for the video conference, of course I can always get around this by just bumping up the max users by one too if the user decides to use the bridge. The second issue might be more of a misconfiguration issue, but I am a bit stuck on where to check. For other times when I don't have the ghost or any flag set for the outbound bridge call, things work as expected. I can transmit and hear audio from both ends to the other one without any issue. Though even though Freeswitch knows the bridged call has no video since has_video is set to false in conference xml_list, it seems to still try to use that as the presenter / floor occasionally. During these times I get the following in fs_cli constantly 2017-05-09 17:10:05.742387 [WARNING] switch_core_media.c:11459 sofia/internal/352650 at asterisk-server has no video codec wher asterisk server is actually the IP/dns host name of the asterisk server I develop with. Ideally if I were not to use the ghost flag I'd want it to treat this bridged call as audio only and not constantly spew that warning message. Some ideas I haven't tested yet as I am not too familiar with freeswitch besides verto/mod_conference a bit, was to see if I could have it join the conference with an audio only profile maybe, or somehow set the SIP not to try offering video maybe. Any ideas? Here is the profile and dialplan I was using to test Conference profile (returned from mod_xml_curl):
Dialplan(Returned from mod_xml_curl):
Thanks in advance, Zach From zrothy at monmouth.com Wed May 10 01:29:21 2017 From: zrothy at monmouth.com (Zach Rothy) Date: Tue, 09 May 2017 17:29:21 -0400 Subject: [Freeswitch-users] Audio Bridge to Asterisk from Freeswitch video conference Message-ID: Good Afternoon, I forgot to mention I am currently running Freeswitch 1.6.12 64bit on Centos 7 from the Freeswitch RPM repo. Thanks in advance, Zach From brian at freeswitch.org Wed May 10 01:55:08 2017 From: brian at freeswitch.org (Brian West) Date: Tue, 9 May 2017 16:55:08 -0500 Subject: [Freeswitch-users] Audio Bridge to Asterisk from Freeswitch video conference In-Reply-To: References: Message-ID: You should probably give it a shot on newest release which is 1.6.17. On Tue, May 9, 2017 at 4:29 PM, Zach Rothy wrote: > Good Afternoon, > > I forgot to mention I am currently running Freeswitch 1.6.12 64bit on > Centos 7 from the Freeswitch RPM repo. > > Thanks in advance, > Zach > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Book a phone call (CST) Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170509/c451ebbc/attachment.html From mike at jerris.com Wed May 10 02:41:46 2017 From: mike at jerris.com (Michael Jerris) Date: Tue, 9 May 2017 18:41:46 -0400 Subject: [Freeswitch-users] Audio Bridge to Asterisk from Freeswitch video conference In-Reply-To: References: Message-ID: Why would you want to bridge another conference in, instead of just running the single conference? Doing so will allow you full access to all the controls for individual members. You could still do this back to back with freeswitch and asterisk by changing asterisk dial plan to bridge to the conference room on freeswitch for each member. > On May 9, 2017, at 5:55 PM, Brian West wrote: > > You should probably give it a shot on newest release which is 1.6.17. > > On Tue, May 9, 2017 at 4:29 PM, Zach Rothy > wrote: > Good Afternoon, > > I forgot to mention I am currently running Freeswitch 1.6.12 64bit on > Centos 7 from the Freeswitch RPM repo. > > Thanks in advance, > Zach > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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.org > Twitter: @FreeSWITCH , @briankwest > > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > Book a phone call (CST) > > Allison prompts for FreeSWITCH: > > https://www.gofundme.com/allison-prompts-for-freeswitch > Got Bugs? Report them here ! | Reddit: /r/freeswitch > T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) > Skype:briankwest > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170509/a9a30498/attachment-0001.html From vladislaus at gmail.com Wed May 10 03:03:09 2017 From: vladislaus at gmail.com (Andres Gomez) Date: Tue, 9 May 2017 18:03:09 -0500 Subject: [Freeswitch-users] BFCP screen sharing Message-ID: Hi Friends can I use bfcp screen sharing in videoconferences? How can i use ? Regards Carlos. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170509/86399b7b/attachment.html From krice at freeswitch.org Wed May 10 03:08:12 2017 From: krice at freeswitch.org (Ken Rice) Date: Tue, 9 May 2017 18:08:12 -0500 Subject: [Freeswitch-users] BFCP screen sharing In-Reply-To: References: Message-ID: <009001d2c919$22a093c0$67e1bb40$@freeswitch.org> FreeSWITCH does not support BFCP at this time? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Andres Gomez Sent: Tuesday, May 9, 2017 6:03 PM To: FreeSWITCH Users Help Subject: [Freeswitch-users] BFCP screen sharing Hi Friends can I use bfcp screen sharing in videoconferences? How can i use ? Regards Carlos. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170509/b5ca82e6/attachment.html From deepikay at iiitd.ac.in Wed May 10 11:02:49 2017 From: deepikay at iiitd.ac.in (Deepika Yadav) Date: Wed, 10 May 2017 12:32:49 +0530 Subject: [Freeswitch-users] Freeswitch PRI support In-Reply-To: <27a9d063.ad3b.15bd85e9762.Coremail.xxxman2008@126.com> References: <27a9d063.ad3b.15bd85e9762.Coremail.xxxman2008@126.com> Message-ID: Hi, I searched some gateways online. The ones available at ebay are cheap, the following link shows a gateway supporting 32 SIMS for $859 : http://www.ebay.com/itm/GoIP-32-32-Port-VoIP-GSM-Gateway- with-32-External-Antenna-/291811535942 However, when I contacted a company called "Matrix Telecom Solution", they gave the quote of $4150 for similar number of SIM support. On asking them to compare the two gateways, they said that the one available at ebay is a Chinese gateway about which they are apprehensive for the quality, warranty and working. Regards, Deepika On Fri, May 5, 2017 at 4:58 PM, Raymond wrote: > > HI, > > Regarding this. i have asked some question of module "GSMOpen". > > In fact ,i want to make Freeswitch box as "GSM VOIP Gateway" by using > "gsmopen". But...seems no one help me . :-(( . If it worked , can save > more money than hardware gsm voip gateway ( GOIP ). > > Raymond > > > > At 2017-05-04 17:24:13, "Giovanni Maruzzelli" wrote: > > For PRI you use Sangoma or Patton. > > But, why don't you use an hardware gateway sip<->gsm? > > It would save you very big money. > > Check on ebay and google, there are many of them, you put SIMs inside, and > you are good to go. > > -giovanni > > > sent from mobile > cell: +39 347 266 56 18 > Giovanni Maruzzelli > OpenTelecom.IT > > Il 04 mag 2017 11:13, "Deepika Yadav" ha scritto: > >> Hi, >> >> I am using Freeswitch in an application that initiates conference calls >> amongst 20 users in cellular network (mobile phones in GSM network). >> Currently, for the VOIP-GSM gateway, we are using the service from a >> company called Doorvaani. >> >> But, since, the cost of call estimates to be high and we cannot debug the >> call drops; we are thinking to buy our own PRI card. >> >> I am seeking recommendation on following points: >> >> 1. Which card should I buy that is most easily configurable with the >> Freeswitch i.e. company and type. >> 2. Reference on how should I start to make Freeswitch configure with >> the PRI card and start sending and receiving calls. For the current gateway >> service in use, I simply put the authentication credentials for the >> corresponding VOIP line offered by the company Doorvaani in the external >> SIP Profile. In case of PRI, what all changes are needed? >> >> Regards, >> Deepika >> >> -- >> Regards >> Deepika >> https://www.iiitd.edu.in/~deepikay/ >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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://www.iiitd.edu.in/~deepikay/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170510/6b21f6d3/attachment.html From bipin at xbipin.com Wed May 10 12:38:08 2017 From: bipin at xbipin.com (Bipin Patel) Date: Wed, 10 May 2017 12:38:08 +0400 Subject: [Freeswitch-users] Freeswitch PRI support In-Reply-To: References: <27a9d063.ad3b.15bd85e9762.Coremail.xxxman2008@126.com> Message-ID: <6fc1c046-45f9-cdbe-2896-a077dbcc5dad@xbipin.com> hi, we use matrix boxes but 4 port ones, they work quiet well Regards, Bipin ------------------------------------------------------------------------ -------- Original Message -------- Subject: Re: [Freeswitch-users] Freeswitch PRI support From: Deepika Yadav To: FreeSWITCH Users Help Date: 5/10/2017, 11:02:49 AM > Hi, > > I searched some gateways online. The ones available at ebay are cheap, > the following link shows a gateway supporting 32 SIMS for $859 : > > http://www.ebay.com/itm/GoIP-32-32-Port-VoIP-GSM-Gateway-with-32-External-Antenna-/291811535942 > > > However, when I contacted a company called "Matrix Telecom Solution", > they gave the quote of $4150 for similar number of SIM support. On > asking them to compare the two gateways, they said that the one > available at ebay is a Chinese gateway about which they are > apprehensive for the quality, warranty and working. > > Regards, > Deepika > > > On Fri, May 5, 2017 at 4:58 PM, Raymond > wrote: > > > HI, > > Regarding this. i have asked some question of module "GSMOpen". > > In fact ,i want to make Freeswitch box as "GSM VOIP Gateway" by > using "gsmopen". But...seems no one help me . :-(( . If it > worked , can save more money than hardware gsm voip gateway ( GOIP ). > > Raymond > > > > At 2017-05-04 17:24:13, "Giovanni Maruzzelli" > wrote: > > For PRI you use Sangoma or Patton. > > But, why don't you use an hardware gateway sip<->gsm? > > It would save you very big money. > > Check on ebay and google, there are many of them, you put SIMs > inside, and you are good to go. > > -giovanni > > > sent from mobile > cell: +39 347 266 56 18 > Giovanni Maruzzelli > OpenTelecom.IT > > Il 04 mag 2017 11:13, "Deepika Yadav" > ha scritto: > > Hi, > > I am using Freeswitch in an application that initiates > conference calls amongst 20 users in cellular network > (mobile phones in GSM network). Currently, for the > VOIP-GSM gateway, we are using the service from a company > called Doorvaani. > > But, since, the cost of call estimates to be high and we > cannot debug the call drops; we are thinking to buy our > own PRI card. > > I am seeking recommendation on following points: > > 1. Which card should I buy that is most easily > configurable with the Freeswitch i.e. company and type. > 2. Reference on how should I start to make Freeswitch > configure with the PRI card and start sending and > receiving calls. For the current gateway service in > use, I simply put the authentication credentials for > the corresponding VOIP line offered by the company > Doorvaani in the external SIP Profile. In case of PRI, > what all changes are needed? > > Regards, > Deepika > > -- > Regards > Deepika > https://www.iiitd.edu.in/~deepikay/ > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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://www.iiitd.edu.in/~deepikay/ > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170510/014207c5/attachment-0001.html From david.villasmil.work at gmail.com Wed May 10 13:35:53 2017 From: david.villasmil.work at gmail.com (David Villasmil) Date: Wed, 10 May 2017 09:35:53 +0000 Subject: [Freeswitch-users] Freeswitch PRI support In-Reply-To: <6fc1c046-45f9-cdbe-2896-a077dbcc5dad@xbipin.com> References: <27a9d063.ad3b.15bd85e9762.Coremail.xxxman2008@126.com> <6fc1c046-45f9-cdbe-2896-a077dbcc5dad@xbipin.com> Message-ID: There's another chinese brand that has a 32 port, I've used it, works pretty well, i think it's "GoIP" or something. On Wed, May 10, 2017 at 10:39 AM Bipin Patel wrote: > hi, > > we use matrix boxes but 4 port ones, they work quiet well > > > Regards, > Bipin > > > ------------------------------ > -------- Original Message -------- > Subject: Re: [Freeswitch-users] Freeswitch PRI support > From: Deepika Yadav > To: FreeSWITCH Users Help > > Date: 5/10/2017, 11:02:49 AM > > Hi, > > I searched some gateways online. The ones available at ebay are cheap, the > following link shows a gateway supporting 32 SIMS for $859 : > > > http://www.ebay.com/itm/GoIP-32-32-Port-VoIP-GSM-Gateway-with-32-External-Antenna-/291811535942 > > However, when I contacted a company called "Matrix Telecom Solution", they > gave the quote of $4150 for similar number of SIM support. On asking them > to compare the two gateways, they said that the one available at ebay is a > Chinese gateway about which they are apprehensive for the quality, warranty > and working. > > Regards, > Deepika > > > On Fri, May 5, 2017 at 4:58 PM, Raymond wrote: > >> >> HI, >> >> Regarding this. i have asked some question of module "GSMOpen". >> >> In fact ,i want to make Freeswitch box as "GSM VOIP Gateway" by using >> "gsmopen". But...seems no one help me . :-(( . If it worked , can save >> more money than hardware gsm voip gateway ( GOIP ). >> >> Raymond >> >> >> >> At 2017-05-04 17:24:13, "Giovanni Maruzzelli" wrote: >> >> For PRI you use Sangoma or Patton. >> >> But, why don't you use an hardware gateway sip<->gsm? >> >> It would save you very big money. >> >> Check on ebay and google, there are many of them, you put SIMs inside, >> and you are good to go. >> >> -giovanni >> >> >> sent from mobile >> cell: +39 347 266 56 18 >> Giovanni Maruzzelli >> OpenTelecom.IT >> >> Il 04 mag 2017 11:13, "Deepika Yadav" ha scritto: >> >>> Hi, >>> >>> I am using Freeswitch in an application that initiates conference calls >>> amongst 20 users in cellular network (mobile phones in GSM network). >>> Currently, for the VOIP-GSM gateway, we are using the service from a >>> company called Doorvaani. >>> >>> But, since, the cost of call estimates to be high and we cannot debug >>> the call drops; we are thinking to buy our own PRI card. >>> >>> I am seeking recommendation on following points: >>> >>> 1. Which card should I buy that is most easily configurable with the >>> Freeswitch i.e. company and type. >>> 2. Reference on how should I start to make Freeswitch configure with >>> the PRI card and start sending and receiving calls. For the current gateway >>> service in use, I simply put the authentication credentials for the >>> corresponding VOIP line offered by the company Doorvaani in the external >>> SIP Profile. In case of PRI, what all changes are needed? >>> >>> Regards, >>> Deepika >>> >>> -- >>> Regards >>> Deepika >>> https://www.iiitd.edu.in/~deepikay/ >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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://www.iiitd.edu.in/~deepikay/ > > > _________________________________________________________________________ > 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170510/9789a25a/attachment.html From colton.conor at gmail.com Wed May 10 17:38:51 2017 From: colton.conor at gmail.com (Colton Conor) Date: Wed, 10 May 2017 08:38:51 -0500 Subject: [Freeswitch-users] OPUS Codec In-Reply-To: References: <20170503071638.GA14349@mail.marktcontact.com> Message-ID: Anyone know on Polycom VVX's using OPUS, what is the sample rate? 8Khz or 48Khz or somewhere inbetween? I looked a at call through VoIP Monitor, and it said OPUS48 as the codec. Buy since "the OPUS specification says that the SDP must always specify 48 kHz and 2 channels" according to Kristian Kielhofner", I assume it just sadys OPUS 48 because of this. Like I mentioned, I though that G.722 between two Polycom's sounded better than OPUS to my suprise. I assume this is because G.722 is at 16 kHZ, and Polycom's opus is potentially at 8kHZ. The softswitch we are using (Netsapiens) does not do any trasconding to my knowledge, so the payload has to be whatever Polycom encodes/decodes at. Is there a way to tell what kHz the Opus codec is running at on a Polcom VVX, or by looked at a packet capture? On Sat, May 6, 2017 at 1:26 PM, Kristian Kielhofner wrote: > FreeSWITCH can limit Opus to a configured sample rate but the OPUS > specification says that the SDP must always specify 48 kHz and 2 > channels. > > OPUS is a combination of SILK and CELT, with the original PLC+FEC > coming from SILK. There are effectively three modes in OPUS - SILK, > CELT, and hybrid. It's not clear to me how well PLC+FEC works in the > hybrid and CELT modes used at higher sample rates. I haven't looked at > it specifically (or tested it) but my suspicion is that PLC+FEC works > much better the closer you get to "pure" SILK mode, which is > essentially what limiting the sample rate to 8 kHz does. > > On Sat, May 6, 2017 at 10:21 AM, Colin Morelli > wrote: > > Transcoding != resampling. Often times you will have to do both to > convert > > audio streams, where transcoding is the act of converting between the two > > audio codecs' data format, and resampling is changing the sampling rate > of > > the audio (8khz in your case). Opus is a completely different audio > codec > > than G.711. Even if they're both at 8 khz, you'll have to transcode > between > > the two, though you may be able to avoid the cost of resampling the > audio to > > a different rate. > > > > Opus can handle sampling rates from 8khz-48khz. Sampling rates make a > huge > > difference to audio quality (they'll effectively define the range of > audio > > frequencies you can capture). If you're running opus at a constant 8khz, > the > > reason G.722 sounds better is almost certainly because it's using a 16khz > > sampling rate. Most likely if you run opus at 16khz, it'll sound similar > to > > G.722. If you have the capacity (and the devices support it), pushing > opus > > to super-wideband or full band (24khz/48khz sampling rates, respectively) > > will yield much better audio quality. > > > > My guess (admittedly it's just an educated assumption), is that most > WebRTC > > companies are using fullband opus in VBR mode simply because this is the > > default in WebRTC. Without mangling the SDP, this is what you're going to > > get. > > > > Best, > > Colin > > > > On Sat, May 6, 2017 at 10:41 AM, Colton Conor > > wrote: > >> > >> So what does today support FEC/PLC today? > >> > >> We tested two Polycom VVX phones with the opus codec, and overall > thought > >> that G.722 sounded much better than OPUS for some reason. > >> > >> So using Opus at 8 KHz, does that mean that there will be less or no > >> transcoding to G.711 since its also in 8 KHz? > >> > >> What are most of the web only WebRTC companies using as far as OPUS > goes? > >> What bitrate and KHz? > >> > >> On Fri, May 5, 2017 at 8:33 AM, Brian West > wrote: > >>> > >>> None of the hardware phones do FEC/PLC last I tested, Sonus has OPUS > >>> support, but again no FEC/PLC support. > >>> > >>> /b > >>> > >>> > >>> On Wed, May 3, 2017 at 2:16 AM, EL wrote: > >>>> > >>>> > >>>> Yealink is supporting OPUS on several other models since firmware > >>>> V81: > >>>> > >>>> Quote: > >>>> "We will support opus on the standard V81 of > >>>> SIP-T40P/T23P/T23G/T2 1(P) E2/T19(P) E2." > >>>> > >>>> Source: > >>>> http://forum.yealink.com/forum/showthread.php?tid= > 29650&pid=51262&mode=threaded > >>>> > >>>> I can confirm OPUS implementation on the 'T21P E2' model. > >>>> > >>>> -- > >>>> EL > >>>> > >>>> > >>>> ____________________________________________________________ > _____________ > >>>> Professional FreeSWITCH Consulting Services: > >>>> consulting at freeswitch.org > >>>> http://www.freeswitchsolutions.com > >>>> > >>>> Official FreeSWITCH Sites > >>>> http://www.freeswitch.org > >>>> http://confluence.freeswitch.org > >>>> http://www.cluecon.com > >>>> > >>>> FreeSWITCH-users mailing list > >>>> FreeSWITCH-users at 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.org > >>> > >>> Twitter: @FreeSWITCH , @briankwest > >>> > >>> http://www.freeswitchbook.com > >>> http://www.freeswitchcookbook.com > >>> > >>> Book a phone call (CST) > >>> > >>> Allison prompts for FreeSWITCH: > >>> > >>> https://www.gofundme.com/allison-prompts-for-freeswitch > >>> > >>> Got Bugs? Report them here! | Reddit: /r/freeswitch > >>> > >>> T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) > >>> Skype:briankwest > >>> > >>> > >>> ____________________________________________________________ > _____________ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting at freeswitch.org > >>> http://www.freeswitchsolutions.com > >>> > >>> Official FreeSWITCH Sites > >>> http://www.freeswitch.org > >>> http://confluence.freeswitch.org > >>> http://www.cluecon.com > >>> > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at 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 > > > > -- > Kristian Kielhofner > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170510/ce33bcde/attachment-0001.html From zrothy at monmouth.com Wed May 10 17:56:13 2017 From: zrothy at monmouth.com (Zach Rothy) Date: Wed, 10 May 2017 09:56:13 -0400 Subject: [Freeswitch-users] Audio Bridge to Asterisk from Freeswitch video conference In-Reply-To: References: Message-ID: <2a49ba951b77ef81aa5ccd6030a974f3@monmouth.com> > You should probably give it a shot on newest release which is 1.6.17. I'll check out the newer version of freeswitch to see if it solves the issue. > Why would you want to bridge another conference in, instead of just > running the single conference? Doing so will allow you full access to > all the controls for individual members. You could still do this back > to back with freeswitch and asterisk by changing asterisk dial plan to > bridge to the conference room on freeswitch for each member. I have it set up so Freeswitch sends it to Asterisk instead of the other way around, as the Asterisk audio bridges are a completely separate feature. Bridging Freeswitch to Asterisk gives them the option of dialing into a video conference for the audio portion without having to change infrastructure a decent amount so that selectively some numbers go to Freeswitch and others Asterisk. I don't think it would be approved anyway to do that, partially due to infrastructure /routing changes for multiple servers, as well as the way video conference is billed versus the Asterisk conference rooms. From virtualguard2015 at gmail.com Wed May 10 04:26:28 2017 From: virtualguard2015 at gmail.com (Ravi sanyal) Date: Wed, 10 May 2017 12:26:28 +1200 Subject: [Freeswitch-users] Setting up webrtc certificates Message-ID: Hello, I'm following the guide on confluence to setup webrtc and testing the various ports for certificate information. I noticed that i need to copy the ca.crt file into the certs folder, however, i used letsencrypt over the self-signed program that was being used for the example. What is the equivalent of this file in letsencrypt? Nathan -- *Virtual Guard Ltd* *info at virtualguard.co.nz * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170510/11572e4e/attachment.html From virtualguard2015 at gmail.com Wed May 10 06:47:24 2017 From: virtualguard2015 at gmail.com (Ravi sanyal) Date: Wed, 10 May 2017 14:47:24 +1200 Subject: [Freeswitch-users] [NOTICE] sofia.c: [INCOMPATIBLE DESTINATION] (setting up NAPTR records?) Message-ID: This is a follow up to the previous problem. I was able to fix the certificates and stop the codec error, however it's still getting INCOMPATIBLE DESTINATION and hanging up as a result. the debug log looks like this: 2017-05-10 14:40:55.164862 [DEBUG] sofia.c:7257 Remote SDP: v=0 o=- 7420685792921982190 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE data a=msid-semantic: WMS m=application 55446 DTLS/SCTP 5000 c=IN IP4 111.69.178.179 a=candidate:1520298561 1 udp 2113937151 192.168.178.32 55446 typ host generation 0 network-cost 50 a=candidate:2664932878 1 udp 2113939711 2406:e007:2f57:1:1d0a:7773:e5e:f3d4 55447 typ host generation 0 network-cost 50 a=candidate:842163049 1 udp 1677729535 111.69.178.179 55446 typ srflx raddr 192.168.178.32 rport 55446 generation 0 network-cost 50 a=ice-ufrag:cCqv a=ice-pwd:m1PW5eEuaLQiFndNoj8Fm9q5 a=fingerprint:sha-256 AB:05:44:B9:4E:AE:4E:FA:E1:74:DE:0B:8C:41:B4:68:42:96:48:B4:C8:46:5A:20:F9:3B:7A:A8:AC:84:64:8E a=setup:actpass a=mid:data a=sctpmap:5000 webrtc-datachannel 1024 2017-05-10 14:40:55.164862 [NOTICE] sofia.c:7728 Hangup sofia/internal/ 1001 at chat.fleetchat.co.nz [CS_NEW] [INCOMPATIBLE_DESTINATION] The sip trace looks like this: SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/WSS 21d9a8odj6h4.invalid;branch=z9hG4bK3506982;received=111.69.178.179;rport=57055 Max-Forwards: 70 From: ;tag=400sjnc8e6 To: ;tag=Bta6SDNmvt28j Call-ID: 6c5ktogfdljrhk3odh8c CSeq: 7347 INVITE User-Agent: FreeSWITCH-mod_sofia/1.9.0-380-c66a012~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" Content-Length: 0 Remote-Party-ID: "1002" ;party=calling;privacy=off;screen=no paste bin of extended sip trace: https://pastebin.freeswitch.org/view/304d4f82 I think it's something to do with SCTP, but knowing nothing of this standard, I am not sure if this is the problem i'm facing. I'm guessing it may be not getting the required information between, since if it is trying to send through sctp ports, my server is not currently set up for that. So how do you set up sctp on freeswitch? i have an article im looking at: https://wiki.freeswitch.org/wiki/SIP_TLS which mentions NAPTR and SRV records. Do you need to set up a DNS server to make NAPTR records? Nathan -- *Virtual Guard Ltd* *info at virtualguard.co.nz * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170510/3cfc1ca2/attachment.html From colton.conor at gmail.com Wed May 10 18:04:20 2017 From: colton.conor at gmail.com (Colton Conor) Date: Wed, 10 May 2017 09:04:20 -0500 Subject: [Freeswitch-users] Freeswitch PRI support In-Reply-To: References: <27a9d063.ad3b.15bd85e9762.Coremail.xxxman2008@126.com> <6fc1c046-45f9-cdbe-2896-a077dbcc5dad@xbipin.com> Message-ID: So how does this GSM stuff work in the USA? Would I got buy a cell phone plan from someone like AT&T or T-Mobile, and then insert the SIM into these devices. The device would then covert the audio from GSM to SIP, and then I could use it like a SIP trunk? What is the advantage of doing this over just buying a pure SIP trunk from an internet provider? On Wed, May 10, 2017 at 4:35 AM, David Villasmil < david.villasmil.work at gmail.com> wrote: > There's another chinese brand that has a 32 port, I've used it, works > pretty well, i think it's "GoIP" or something. > > On Wed, May 10, 2017 at 10:39 AM Bipin Patel wrote: > >> hi, >> >> we use matrix boxes but 4 port ones, they work quiet well >> >> >> Regards, >> Bipin >> >> >> ------------------------------ >> -------- Original Message -------- >> Subject: Re: [Freeswitch-users] Freeswitch PRI support >> From: Deepika Yadav >> To: FreeSWITCH Users Help >> >> Date: 5/10/2017, 11:02:49 AM >> >> Hi, >> >> I searched some gateways online. The ones available at ebay are cheap, >> the following link shows a gateway supporting 32 SIMS for $859 : >> >> http://www.ebay.com/itm/GoIP-32-32-Port-VoIP-GSM-Gateway- >> with-32-External-Antenna-/291811535942 >> >> However, when I contacted a company called "Matrix Telecom Solution", >> they gave the quote of $4150 for similar number of SIM support. On asking >> them to compare the two gateways, they said that the one available at ebay >> is a Chinese gateway about which they are apprehensive for the quality, >> warranty and working. >> >> Regards, >> Deepika >> >> >> On Fri, May 5, 2017 at 4:58 PM, Raymond wrote: >> >>> >>> HI, >>> >>> Regarding this. i have asked some question of module "GSMOpen". >>> >>> In fact ,i want to make Freeswitch box as "GSM VOIP Gateway" by using >>> "gsmopen". But...seems no one help me . :-(( . If it worked , can save >>> more money than hardware gsm voip gateway ( GOIP ). >>> >>> Raymond >>> >>> >>> >>> At 2017-05-04 17:24:13, "Giovanni Maruzzelli" wrote: >>> >>> For PRI you use Sangoma or Patton. >>> >>> But, why don't you use an hardware gateway sip<->gsm? >>> >>> It would save you very big money. >>> >>> Check on ebay and google, there are many of them, you put SIMs inside, >>> and you are good to go. >>> >>> -giovanni >>> >>> >>> sent from mobile >>> cell: +39 347 266 56 18 >>> Giovanni Maruzzelli >>> OpenTelecom.IT >>> >>> Il 04 mag 2017 11:13, "Deepika Yadav" ha scritto: >>> >>>> Hi, >>>> >>>> I am using Freeswitch in an application that initiates conference calls >>>> amongst 20 users in cellular network (mobile phones in GSM network). >>>> Currently, for the VOIP-GSM gateway, we are using the service from a >>>> company called Doorvaani. >>>> >>>> But, since, the cost of call estimates to be high and we cannot debug >>>> the call drops; we are thinking to buy our own PRI card. >>>> >>>> I am seeking recommendation on following points: >>>> >>>> 1. Which card should I buy that is most easily configurable with >>>> the Freeswitch i.e. company and type. >>>> 2. Reference on how should I start to make Freeswitch configure >>>> with the PRI card and start sending and receiving calls. For the current >>>> gateway service in use, I simply put the authentication credentials for the >>>> corresponding VOIP line offered by the company Doorvaani in the external >>>> SIP Profile. In case of PRI, what all changes are needed? >>>> >>>> Regards, >>>> Deepika >>>> >>>> -- >>>> Regards >>>> Deepika >>>> https://www.iiitd.edu.in/~deepikay/ >>>> >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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://www.iiitd.edu.in/~deepikay/ >> >> >> _________________________________________________________________________ >> 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170510/dd928a69/attachment-0001.html From rbetancor at gmail.com Wed May 10 18:12:21 2017 From: rbetancor at gmail.com (=?UTF-8?Q?Ra=C3=BAl_Alexis_Betancor_Santana?=) Date: Wed, 10 May 2017 15:12:21 +0100 Subject: [Freeswitch-users] Freeswitch PRI support In-Reply-To: References: <27a9d063.ad3b.15bd85e9762.Coremail.xxxman2008@126.com> <6fc1c046-45f9-cdbe-2896-a077dbcc5dad@xbipin.com> Message-ID: The only real advantage over a regular SIP trunk from any provider, it's that you could select a plan than MAYBE allow you some sort of flat-rate and not per-minute use plans ... but it doesn't worth the effort and money you have to spend to get that to generate revenue ... also most of celullar carriers withh put some restrictions on the usage of the flat-rate, as all of them have 'fair use' policies. At least on Spain, it's ilegal to resell that kind of traffic and I asume its the same in most of the world. 2017-05-10 15:04 GMT+01:00 Colton Conor : > So how does this GSM stuff work in the USA? Would I got buy a cell phone > plan from someone like AT&T or T-Mobile, and then insert the SIM into these > devices. The device would then covert the audio from GSM to SIP, and then I > could use it like a SIP trunk? > > What is the advantage of doing this over just buying a pure SIP trunk from > an internet provider? > > > > On Wed, May 10, 2017 at 4:35 AM, David Villasmil < > david.villasmil.work at gmail.com> wrote: > >> There's another chinese brand that has a 32 port, I've used it, works >> pretty well, i think it's "GoIP" or something. >> >> On Wed, May 10, 2017 at 10:39 AM Bipin Patel wrote: >> >>> hi, >>> >>> we use matrix boxes but 4 port ones, they work quiet well >>> >>> >>> Regards, >>> Bipin >>> >>> >>> ------------------------------ >>> -------- Original Message -------- >>> Subject: Re: [Freeswitch-users] Freeswitch PRI support >>> From: Deepika Yadav >>> To: FreeSWITCH Users Help >>> >>> Date: 5/10/2017, 11:02:49 AM >>> >>> Hi, >>> >>> I searched some gateways online. The ones available at ebay are cheap, >>> the following link shows a gateway supporting 32 SIMS for $859 : >>> >>> http://www.ebay.com/itm/GoIP-32-32-Port-VoIP-GSM-Gateway-wit >>> h-32-External-Antenna-/291811535942 >>> >>> However, when I contacted a company called "Matrix Telecom Solution", >>> they gave the quote of $4150 for similar number of SIM support. On asking >>> them to compare the two gateways, they said that the one available at ebay >>> is a Chinese gateway about which they are apprehensive for the quality, >>> warranty and working. >>> >>> Regards, >>> Deepika >>> >>> >>> On Fri, May 5, 2017 at 4:58 PM, Raymond wrote: >>> >>>> >>>> HI, >>>> >>>> Regarding this. i have asked some question of module "GSMOpen". >>>> >>>> In fact ,i want to make Freeswitch box as "GSM VOIP Gateway" by using >>>> "gsmopen". But...seems no one help me . :-(( . If it worked , can save >>>> more money than hardware gsm voip gateway ( GOIP ). >>>> >>>> Raymond >>>> >>>> >>>> >>>> At 2017-05-04 17:24:13, "Giovanni Maruzzelli" >>>> wrote: >>>> >>>> For PRI you use Sangoma or Patton. >>>> >>>> But, why don't you use an hardware gateway sip<->gsm? >>>> >>>> It would save you very big money. >>>> >>>> Check on ebay and google, there are many of them, you put SIMs inside, >>>> and you are good to go. >>>> >>>> -giovanni >>>> >>>> >>>> sent from mobile >>>> cell: +39 347 266 56 18 >>>> Giovanni Maruzzelli >>>> OpenTelecom.IT >>>> >>>> Il 04 mag 2017 11:13, "Deepika Yadav" ha >>>> scritto: >>>> >>>>> Hi, >>>>> >>>>> I am using Freeswitch in an application that initiates conference >>>>> calls amongst 20 users in cellular network (mobile phones in GSM network). >>>>> Currently, for the VOIP-GSM gateway, we are using the service from a >>>>> company called Doorvaani. >>>>> >>>>> But, since, the cost of call estimates to be high and we cannot debug >>>>> the call drops; we are thinking to buy our own PRI card. >>>>> >>>>> I am seeking recommendation on following points: >>>>> >>>>> 1. Which card should I buy that is most easily configurable with >>>>> the Freeswitch i.e. company and type. >>>>> 2. Reference on how should I start to make Freeswitch configure >>>>> with the PRI card and start sending and receiving calls. For the current >>>>> gateway service in use, I simply put the authentication credentials for the >>>>> corresponding VOIP line offered by the company Doorvaani in the external >>>>> SIP Profile. In case of PRI, what all changes are needed? >>>>> >>>>> Regards, >>>>> Deepika >>>>> >>>>> -- >>>>> Regards >>>>> Deepika >>>>> https://www.iiitd.edu.in/~deepikay/ >>>>> >>>>> ____________________________________________________________ >>>>> _____________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at 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://www.iiitd.edu.in/~deepikay/ >>> >>> >>> _________________________________________________________________________ >>> 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170510/39e0808b/attachment-0001.html From david.villasmil.work at gmail.com Wed May 10 19:06:26 2017 From: david.villasmil.work at gmail.com (David Villasmil) Date: Wed, 10 May 2017 17:06:26 +0200 Subject: [Freeswitch-users] Freeswitch PRI support In-Reply-To: References: <27a9d063.ad3b.15bd85e9762.Coremail.xxxman2008@126.com> <6fc1c046-45f9-cdbe-2896-a077dbcc5dad@xbipin.com> Message-ID: I think it's illegal everywhere... but value? there's lots and lots of values, it only depends on where you are doing it... :) ? Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 On Wed, May 10, 2017 at 4:12 PM, Ra?l Alexis Betancor Santana < rbetancor at gmail.com> wrote: > The only real advantage over a regular SIP trunk from any provider, it's > that you could select a plan than MAYBE allow you some sort of flat-rate > and not per-minute use plans ... but it doesn't worth the effort and money > you have to spend to get that to generate revenue ... also most of celullar > carriers withh put some restrictions on the usage of the flat-rate, as all > of them have 'fair use' policies. > At least on Spain, it's ilegal to resell that kind of traffic and I asume > its the same in most of the world. > > 2017-05-10 15:04 GMT+01:00 Colton Conor : > >> So how does this GSM stuff work in the USA? Would I got buy a cell phone >> plan from someone like AT&T or T-Mobile, and then insert the SIM into these >> devices. The device would then covert the audio from GSM to SIP, and then I >> could use it like a SIP trunk? >> >> What is the advantage of doing this over just buying a pure SIP trunk >> from an internet provider? >> >> >> >> On Wed, May 10, 2017 at 4:35 AM, David Villasmil < >> david.villasmil.work at gmail.com> wrote: >> >>> There's another chinese brand that has a 32 port, I've used it, works >>> pretty well, i think it's "GoIP" or something. >>> >>> On Wed, May 10, 2017 at 10:39 AM Bipin Patel wrote: >>> >>>> hi, >>>> >>>> we use matrix boxes but 4 port ones, they work quiet well >>>> >>>> >>>> Regards, >>>> Bipin >>>> >>>> >>>> ------------------------------ >>>> -------- Original Message -------- >>>> Subject: Re: [Freeswitch-users] Freeswitch PRI support >>>> From: Deepika Yadav >>>> To: FreeSWITCH Users Help >>>> >>>> Date: 5/10/2017, 11:02:49 AM >>>> >>>> Hi, >>>> >>>> I searched some gateways online. The ones available at ebay are cheap, >>>> the following link shows a gateway supporting 32 SIMS for $859 : >>>> >>>> http://www.ebay.com/itm/GoIP-32-32-Port-VoIP-GSM-Gateway-wit >>>> h-32-External-Antenna-/291811535942 >>>> >>>> However, when I contacted a company called "Matrix Telecom Solution", >>>> they gave the quote of $4150 for similar number of SIM support. On asking >>>> them to compare the two gateways, they said that the one available at ebay >>>> is a Chinese gateway about which they are apprehensive for the quality, >>>> warranty and working. >>>> >>>> Regards, >>>> Deepika >>>> >>>> >>>> On Fri, May 5, 2017 at 4:58 PM, Raymond wrote: >>>> >>>>> >>>>> HI, >>>>> >>>>> Regarding this. i have asked some question of module "GSMOpen". >>>>> >>>>> In fact ,i want to make Freeswitch box as "GSM VOIP Gateway" by >>>>> using "gsmopen". But...seems no one help me . :-(( . If it worked , can >>>>> save more money than hardware gsm voip gateway ( GOIP ). >>>>> >>>>> Raymond >>>>> >>>>> >>>>> >>>>> At 2017-05-04 17:24:13, "Giovanni Maruzzelli" >>>>> wrote: >>>>> >>>>> For PRI you use Sangoma or Patton. >>>>> >>>>> But, why don't you use an hardware gateway sip<->gsm? >>>>> >>>>> It would save you very big money. >>>>> >>>>> Check on ebay and google, there are many of them, you put SIMs inside, >>>>> and you are good to go. >>>>> >>>>> -giovanni >>>>> >>>>> >>>>> sent from mobile >>>>> cell: +39 347 266 56 18 >>>>> Giovanni Maruzzelli >>>>> OpenTelecom.IT >>>>> >>>>> Il 04 mag 2017 11:13, "Deepika Yadav" ha >>>>> scritto: >>>>> >>>>>> Hi, >>>>>> >>>>>> I am using Freeswitch in an application that initiates conference >>>>>> calls amongst 20 users in cellular network (mobile phones in GSM network). >>>>>> Currently, for the VOIP-GSM gateway, we are using the service from a >>>>>> company called Doorvaani. >>>>>> >>>>>> But, since, the cost of call estimates to be high and we cannot debug >>>>>> the call drops; we are thinking to buy our own PRI card. >>>>>> >>>>>> I am seeking recommendation on following points: >>>>>> >>>>>> 1. Which card should I buy that is most easily configurable with >>>>>> the Freeswitch i.e. company and type. >>>>>> 2. Reference on how should I start to make Freeswitch configure >>>>>> with the PRI card and start sending and receiving calls. For the current >>>>>> gateway service in use, I simply put the authentication credentials for the >>>>>> corresponding VOIP line offered by the company Doorvaani in the external >>>>>> SIP Profile. In case of PRI, what all changes are needed? >>>>>> >>>>>> Regards, >>>>>> Deepika >>>>>> >>>>>> -- >>>>>> Regards >>>>>> Deepika >>>>>> https://www.iiitd.edu.in/~deepikay/ >>>>>> >>>>>> ____________________________________________________________ >>>>>> _____________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-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 >>>>> >>>> >>>> >>>> >>>> -- >>>> Regards >>>> Deepika >>>> https://www.iiitd.edu.in/~deepikay/ >>>> >>>> >>>> _________________________________________________________________________ >>>> 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/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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170510/2effa087/attachment-0001.html From dragos.oancea at vonage.com Wed May 10 19:35:46 2017 From: dragos.oancea at vonage.com (Oancea, Dragos) Date: Wed, 10 May 2017 16:35:46 +0100 Subject: [Freeswitch-users] OPUS Codec In-Reply-To: References: <20170503071638.GA14349@mail.marktcontact.com> Message-ID: Better check with "opus_debug on" on the FS CLI. Eg: 2017-05-10 16:28:05.724001 [DEBUG] mod_opus.c:453 decode: opus_frames [1] samples [960] audio bandwidth [WIDEBAND] bytes [75] FEC[yes] channels[1] Look at the audio bandwidth of the decoded packets . If the audio bandwitdh is WIDEBAND, then the sample rate is 16 khz. You cannot find this out by looking at the pcap , the RTP timestamp is incremented as described by the SDP profile , so always at 48 khz (which is 960 for 20 ms ptime ) . Cheers, Dragos On Wed, May 10, 2017 at 2:38 PM, Colton Conor wrote: > Anyone know on Polycom VVX's using OPUS, what is the sample rate? 8Khz or > 48Khz or somewhere inbetween? > > I looked a at call through VoIP Monitor, and it said OPUS48 as the codec. > Buy since "the OPUS > specification says that the SDP must always specify 48 kHz and 2 channels" > according to Kristian Kielhofner", I assume it just sadys OPUS 48 because > of this. > > Like I mentioned, I though that G.722 between two Polycom's sounded better > than OPUS to my suprise. I assume this is because G.722 is at 16 kHZ, and > Polycom's opus is potentially at 8kHZ. > The softswitch we are using (Netsapiens) does not do any trasconding to my > knowledge, so the payload has to be whatever Polycom encodes/decodes at. > > Is there a way to tell what kHz the Opus codec is running at on a Polcom > VVX, or by looked at a packet capture? > > > On Sat, May 6, 2017 at 1:26 PM, Kristian Kielhofner > wrote: > >> FreeSWITCH can limit Opus to a configured sample rate but the OPUS >> specification says that the SDP must always specify 48 kHz and 2 >> channels. >> >> OPUS is a combination of SILK and CELT, with the original PLC+FEC >> coming from SILK. There are effectively three modes in OPUS - SILK, >> CELT, and hybrid. It's not clear to me how well PLC+FEC works in the >> hybrid and CELT modes used at higher sample rates. I haven't looked at >> it specifically (or tested it) but my suspicion is that PLC+FEC works >> much better the closer you get to "pure" SILK mode, which is >> essentially what limiting the sample rate to 8 kHz does. >> >> On Sat, May 6, 2017 at 10:21 AM, Colin Morelli >> wrote: >> > Transcoding != resampling. Often times you will have to do both to >> convert >> > audio streams, where transcoding is the act of converting between the >> two >> > audio codecs' data format, and resampling is changing the sampling rate >> of >> > the audio (8khz in your case). Opus is a completely different audio >> codec >> > than G.711. Even if they're both at 8 khz, you'll have to transcode >> between >> > the two, though you may be able to avoid the cost of resampling the >> audio to >> > a different rate. >> > >> > Opus can handle sampling rates from 8khz-48khz. Sampling rates make a >> huge >> > difference to audio quality (they'll effectively define the range of >> audio >> > frequencies you can capture). If you're running opus at a constant >> 8khz, the >> > reason G.722 sounds better is almost certainly because it's using a >> 16khz >> > sampling rate. Most likely if you run opus at 16khz, it'll sound >> similar to >> > G.722. If you have the capacity (and the devices support it), pushing >> opus >> > to super-wideband or full band (24khz/48khz sampling rates, >> respectively) >> > will yield much better audio quality. >> > >> > My guess (admittedly it's just an educated assumption), is that most >> WebRTC >> > companies are using fullband opus in VBR mode simply because this is the >> > default in WebRTC. Without mangling the SDP, this is what you're going >> to >> > get. >> > >> > Best, >> > Colin >> > >> > On Sat, May 6, 2017 at 10:41 AM, Colton Conor >> > wrote: >> >> >> >> So what does today support FEC/PLC today? >> >> >> >> We tested two Polycom VVX phones with the opus codec, and overall >> thought >> >> that G.722 sounded much better than OPUS for some reason. >> >> >> >> So using Opus at 8 KHz, does that mean that there will be less or no >> >> transcoding to G.711 since its also in 8 KHz? >> >> >> >> What are most of the web only WebRTC companies using as far as OPUS >> goes? >> >> What bitrate and KHz? >> >> >> >> On Fri, May 5, 2017 at 8:33 AM, Brian West >> wrote: >> >>> >> >>> None of the hardware phones do FEC/PLC last I tested, Sonus has OPUS >> >>> support, but again no FEC/PLC support. >> >>> >> >>> /b >> >>> >> >>> >> >>> On Wed, May 3, 2017 at 2:16 AM, EL wrote: >> >>>> >> >>>> >> >>>> Yealink is supporting OPUS on several other models since firmware >> >>>> V81: >> >>>> >> >>>> Quote: >> >>>> "We will support opus on the standard V81 of >> >>>> SIP-T40P/T23P/T23G/T2 1(P) E2/T19(P) E2." >> >>>> >> >>>> Source: >> >>>> http://forum.yealink.com/forum/showthread.php?tid=29650&pid= >> 51262&mode=threaded >> >>>> >> >>>> I can confirm OPUS implementation on the 'T21P E2' model. >> >>>> >> >>>> -- >> >>>> EL >> >>>> >> >>>> >> >>>> ____________________________________________________________ >> _____________ >> >>>> Professional FreeSWITCH Consulting Services: >> >>>> consulting at freeswitch.org >> >>>> http://www.freeswitchsolutions.com >> >>>> >> >>>> Official FreeSWITCH Sites >> >>>> http://www.freeswitch.org >> >>>> http://confluence.freeswitch.org >> >>>> http://www.cluecon.com >> >>>> >> >>>> FreeSWITCH-users mailing list >> >>>> FreeSWITCH-users at 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.org >> >>> >> >>> Twitter: @FreeSWITCH , @briankwest >> >>> >> >>> http://www.freeswitchbook.com >> >>> http://www.freeswitchcookbook.com >> >>> >> >>> Book a phone call (CST) >> >>> >> >>> Allison prompts for FreeSWITCH: >> >>> >> >>> https://www.gofundme.com/allison-prompts-for-freeswitch >> >>> >> >>> Got Bugs? Report them here! | Reddit: /r/freeswitch >> >>> >> >>> T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) >> >>> Skype:briankwest >> >>> >> >>> >> >>> ____________________________________________________________ >> _____________ >> >>> Professional FreeSWITCH Consulting Services: >> >>> consulting at freeswitch.org >> >>> http://www.freeswitchsolutions.com >> >>> >> >>> Official FreeSWITCH Sites >> >>> http://www.freeswitch.org >> >>> http://confluence.freeswitch.org >> >>> http://www.cluecon.com >> >>> >> >>> FreeSWITCH-users mailing list >> >>> FreeSWITCH-users at 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 >> >> >> >> -- >> Kristian Kielhofner >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170510/a5b663dd/attachment-0001.html From kris at kriskinc.com Wed May 10 20:40:23 2017 From: kris at kriskinc.com (Kristian Kielhofner) Date: Wed, 10 May 2017 11:40:23 -0500 Subject: [Freeswitch-users] OPUS Codec In-Reply-To: References: <20170503071638.GA14349@mail.marktcontact.com> Message-ID: Just because "according to Kristian Kielhofner" sounds strange to me; how about according to RFC 7587 :)? https://tools.ietf.org/html/rfc7587#section-7 "The media subtype ("opus") goes in SDP "a=rtpmap" as the encoding name. The RTP clock rate in "a=rtpmap" MUST be 48000, and the number of channels MUST be 2." On Wed, May 10, 2017 at 8:38 AM, Colton Conor wrote: > Anyone know on Polycom VVX's using OPUS, what is the sample rate? 8Khz or > 48Khz or somewhere inbetween? > > I looked a at call through VoIP Monitor, and it said OPUS48 as the codec. > Buy since "the OPUS > specification says that the SDP must always specify 48 kHz and 2 channels" > according to Kristian Kielhofner", I assume it just sadys OPUS 48 because of > this. > > Like I mentioned, I though that G.722 between two Polycom's sounded better > than OPUS to my suprise. I assume this is because G.722 is at 16 kHZ, and > Polycom's opus is potentially at 8kHZ. > The softswitch we are using (Netsapiens) does not do any trasconding to my > knowledge, so the payload has to be whatever Polycom encodes/decodes at. > > Is there a way to tell what kHz the Opus codec is running at on a Polcom > VVX, or by looked at a packet capture? > > > On Sat, May 6, 2017 at 1:26 PM, Kristian Kielhofner > wrote: >> >> FreeSWITCH can limit Opus to a configured sample rate but the OPUS >> specification says that the SDP must always specify 48 kHz and 2 >> channels. >> >> OPUS is a combination of SILK and CELT, with the original PLC+FEC >> coming from SILK. There are effectively three modes in OPUS - SILK, >> CELT, and hybrid. It's not clear to me how well PLC+FEC works in the >> hybrid and CELT modes used at higher sample rates. I haven't looked at >> it specifically (or tested it) but my suspicion is that PLC+FEC works >> much better the closer you get to "pure" SILK mode, which is >> essentially what limiting the sample rate to 8 kHz does. >> >> On Sat, May 6, 2017 at 10:21 AM, Colin Morelli >> wrote: >> > Transcoding != resampling. Often times you will have to do both to >> > convert >> > audio streams, where transcoding is the act of converting between the >> > two >> > audio codecs' data format, and resampling is changing the sampling rate >> > of >> > the audio (8khz in your case). Opus is a completely different audio >> > codec >> > than G.711. Even if they're both at 8 khz, you'll have to transcode >> > between >> > the two, though you may be able to avoid the cost of resampling the >> > audio to >> > a different rate. >> > >> > Opus can handle sampling rates from 8khz-48khz. Sampling rates make a >> > huge >> > difference to audio quality (they'll effectively define the range of >> > audio >> > frequencies you can capture). If you're running opus at a constant 8khz, >> > the >> > reason G.722 sounds better is almost certainly because it's using a >> > 16khz >> > sampling rate. Most likely if you run opus at 16khz, it'll sound similar >> > to >> > G.722. If you have the capacity (and the devices support it), pushing >> > opus >> > to super-wideband or full band (24khz/48khz sampling rates, >> > respectively) >> > will yield much better audio quality. >> > >> > My guess (admittedly it's just an educated assumption), is that most >> > WebRTC >> > companies are using fullband opus in VBR mode simply because this is the >> > default in WebRTC. Without mangling the SDP, this is what you're going >> > to >> > get. >> > >> > Best, >> > Colin >> > >> > On Sat, May 6, 2017 at 10:41 AM, Colton Conor >> > wrote: >> >> >> >> So what does today support FEC/PLC today? >> >> >> >> We tested two Polycom VVX phones with the opus codec, and overall >> >> thought >> >> that G.722 sounded much better than OPUS for some reason. >> >> >> >> So using Opus at 8 KHz, does that mean that there will be less or no >> >> transcoding to G.711 since its also in 8 KHz? >> >> >> >> What are most of the web only WebRTC companies using as far as OPUS >> >> goes? >> >> What bitrate and KHz? >> >> >> >> On Fri, May 5, 2017 at 8:33 AM, Brian West >> >> wrote: >> >>> >> >>> None of the hardware phones do FEC/PLC last I tested, Sonus has OPUS >> >>> support, but again no FEC/PLC support. >> >>> >> >>> /b >> >>> >> >>> >> >>> On Wed, May 3, 2017 at 2:16 AM, EL wrote: >> >>>> >> >>>> >> >>>> Yealink is supporting OPUS on several other models since firmware >> >>>> V81: >> >>>> >> >>>> Quote: >> >>>> "We will support opus on the standard V81 of >> >>>> SIP-T40P/T23P/T23G/T2 1(P) E2/T19(P) E2." >> >>>> >> >>>> Source: >> >>>> >> >>>> http://forum.yealink.com/forum/showthread.php?tid=29650&pid=51262&mode=threaded >> >>>> >> >>>> I can confirm OPUS implementation on the 'T21P E2' model. >> >>>> >> >>>> -- >> >>>> EL >> >>>> >> >>>> >> >>>> >> >>>> _________________________________________________________________________ >> >>>> Professional FreeSWITCH Consulting Services: >> >>>> consulting at freeswitch.org >> >>>> http://www.freeswitchsolutions.com >> >>>> >> >>>> Official FreeSWITCH Sites >> >>>> http://www.freeswitch.org >> >>>> http://confluence.freeswitch.org >> >>>> http://www.cluecon.com >> >>>> >> >>>> FreeSWITCH-users mailing list >> >>>> FreeSWITCH-users at 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.org >> >>> >> >>> Twitter: @FreeSWITCH , @briankwest >> >>> >> >>> http://www.freeswitchbook.com >> >>> http://www.freeswitchcookbook.com >> >>> >> >>> Book a phone call (CST) >> >>> >> >>> Allison prompts for FreeSWITCH: >> >>> >> >>> https://www.gofundme.com/allison-prompts-for-freeswitch >> >>> >> >>> Got Bugs? Report them here! | Reddit: /r/freeswitch >> >>> >> >>> T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) >> >>> Skype:briankwest >> >>> >> >>> >> >>> >> >>> _________________________________________________________________________ >> >>> Professional FreeSWITCH Consulting Services: >> >>> consulting at freeswitch.org >> >>> http://www.freeswitchsolutions.com >> >>> >> >>> Official FreeSWITCH Sites >> >>> http://www.freeswitch.org >> >>> http://confluence.freeswitch.org >> >>> http://www.cluecon.com >> >>> >> >>> FreeSWITCH-users mailing list >> >>> FreeSWITCH-users at 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 >> >> >> >> -- >> Kristian Kielhofner >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 -- Kristian Kielhofner From mike at jerris.com Wed May 10 20:55:13 2017 From: mike at jerris.com (Michael Jerris) Date: Wed, 10 May 2017 12:55:13 -0400 Subject: [Freeswitch-users] OPUS Codec In-Reply-To: References: <20170503071638.GA14349@mail.marktcontact.com> Message-ID: > On May 10, 2017, at 12:40 PM, Kristian Kielhofner wrote: > > Just because "according to Kristian Kielhofner" sounds strange to me; > how about according to RFC 7587 :)? > > https://tools.ietf.org/html/rfc7587#section-7 > > "The media subtype ("opus") goes in SDP "a=rtpmap" as the encoding > name. The RTP clock rate in "a=rtpmap" MUST be 48000, and the number > of channels MUST be 2.? - According to Kristian Kielhofner and RFC 7587?. > > On Wed, May 10, 2017 at 8:38 AM, Colton Conor wrote: >> Anyone know on Polycom VVX's using OPUS, what is the sample rate? 8Khz or >> 48Khz or somewhere inbetween? >> >> I looked a at call through VoIP Monitor, and it said OPUS48 as the codec. >> Buy since "the OPUS >> specification says that the SDP must always specify 48 kHz and 2 channels" >> according to Kristian Kielhofner", I assume it just sadys OPUS 48 because of >> this. >> >> Like I mentioned, I though that G.722 between two Polycom's sounded better >> than OPUS to my suprise. I assume this is because G.722 is at 16 kHZ, and >> Polycom's opus is potentially at 8kHZ. >> The softswitch we are using (Netsapiens) does not do any trasconding to my >> knowledge, so the payload has to be whatever Polycom encodes/decodes at. >> >> Is there a way to tell what kHz the Opus codec is running at on a Polcom >> VVX, or by looked at a packet capture? >> >> >> On Sat, May 6, 2017 at 1:26 PM, Kristian Kielhofner >> wrote: >>> >>> FreeSWITCH can limit Opus to a configured sample rate but the OPUS >>> specification says that the SDP must always specify 48 kHz and 2 >>> channels. >>> >>> OPUS is a combination of SILK and CELT, with the original PLC+FEC >>> coming from SILK. There are effectively three modes in OPUS - SILK, >>> CELT, and hybrid. It's not clear to me how well PLC+FEC works in the >>> hybrid and CELT modes used at higher sample rates. I haven't looked at >>> it specifically (or tested it) but my suspicion is that PLC+FEC works >>> much better the closer you get to "pure" SILK mode, which is >>> essentially what limiting the sample rate to 8 kHz does. >>> >>> On Sat, May 6, 2017 at 10:21 AM, Colin Morelli >>> wrote: >>>> Transcoding != resampling. Often times you will have to do both to >>>> convert >>>> audio streams, where transcoding is the act of converting between the >>>> two >>>> audio codecs' data format, and resampling is changing the sampling rate >>>> of >>>> the audio (8khz in your case). Opus is a completely different audio >>>> codec >>>> than G.711. Even if they're both at 8 khz, you'll have to transcode >>>> between >>>> the two, though you may be able to avoid the cost of resampling the >>>> audio to >>>> a different rate. >>>> >>>> Opus can handle sampling rates from 8khz-48khz. Sampling rates make a >>>> huge >>>> difference to audio quality (they'll effectively define the range of >>>> audio >>>> frequencies you can capture). If you're running opus at a constant 8khz, >>>> the >>>> reason G.722 sounds better is almost certainly because it's using a >>>> 16khz >>>> sampling rate. Most likely if you run opus at 16khz, it'll sound similar >>>> to >>>> G.722. If you have the capacity (and the devices support it), pushing >>>> opus >>>> to super-wideband or full band (24khz/48khz sampling rates, >>>> respectively) >>>> will yield much better audio quality. >>>> >>>> My guess (admittedly it's just an educated assumption), is that most >>>> WebRTC >>>> companies are using fullband opus in VBR mode simply because this is the >>>> default in WebRTC. Without mangling the SDP, this is what you're going >>>> to >>>> get. >>>> >>>> Best, >>>> Colin >>>> >>>> On Sat, May 6, 2017 at 10:41 AM, Colton Conor >>>> wrote: >>>>> >>>>> So what does today support FEC/PLC today? >>>>> >>>>> We tested two Polycom VVX phones with the opus codec, and overall >>>>> thought >>>>> that G.722 sounded much better than OPUS for some reason. >>>>> >>>>> So using Opus at 8 KHz, does that mean that there will be less or no >>>>> transcoding to G.711 since its also in 8 KHz? >>>>> >>>>> What are most of the web only WebRTC companies using as far as OPUS >>>>> goes? >>>>> What bitrate and KHz? >>>>> >>>>> On Fri, May 5, 2017 at 8:33 AM, Brian West >>>>> wrote: >>>>>> >>>>>> None of the hardware phones do FEC/PLC last I tested, Sonus has OPUS >>>>>> support, but again no FEC/PLC support. >>>>>> >>>>>> /b >>>>>> >>>>>> >>>>>> On Wed, May 3, 2017 at 2:16 AM, EL wrote: >>>>>>> >>>>>>> >>>>>>> Yealink is supporting OPUS on several other models since firmware >>>>>>> V81: >>>>>>> >>>>>>> Quote: >>>>>>> "We will support opus on the standard V81 of >>>>>>> SIP-T40P/T23P/T23G/T2 1(P) E2/T19(P) E2." >>>>>>> >>>>>>> Source: >>>>>>> >>>>>>> http://forum.yealink.com/forum/showthread.php?tid=29650&pid=51262&mode=threaded >>>>>>> >>>>>>> I can confirm OPUS implementation on the 'T21P E2' model. From zrothy at monmouth.com Wed May 10 21:42:20 2017 From: zrothy at monmouth.com (Zach Rothy) Date: Wed, 10 May 2017 13:42:20 -0400 Subject: [Freeswitch-users] Audio Bridge to Asterisk from Freeswitch video conference Message-ID: <83deee6696f9df53c2ed6dcf11f17ed4@monmouth.com> So I tested with Freeswitch 1.6.17 as Brian suggested earlier, and it seems that the errors and warnings I saw still occur. To summarize the original post I made, I have my Freeswitch server which supplies the server for video conferences using Verto. I have it bridging to an Asterisk server (testing with Asterisk 13) to allow PSTN calls to be bridged to Freeswitch via an Asterisk MeetMe conference room. Freeswitch gets its dialplan for Verto via mod_xml_curl and sends the call to Asterisk via the conference auto outcall application. When I set the flags for this bridged call to ghost it seems Freeswitch is receiving the RTP, as I see verto say "Talking" properly, but I never hear anything from my verto client connected to the video conference room. This does not happen when the ghost flag is removed, as I can speak from either end and hear it on the other one. Additionally when the bridged call initially joins the conference, it seems to be being placed as the presenter even though I have the flags so that without video it can't be on the canvas. Such I get the following warning flooding the console: 2017-05-10 12:51:57.505225 [WARNING] switch_core_media.c:11774 sofia/internal/352650 at asterskdev has no video codec This warning stops both when the floor is not trying to be set to the audio bridge, or when the floor is locked to another presenter. It recognizes properly you can't set the vid floor to that bridged call in fs_cli as it says in console: 2017-05-10 13:33:18.125230 [DEBUG] conference_event.c:110 conf 3525236 CMD mod_verto [vid-floor] 0002 2017-05-10 13:33:18.125230 [ERR] conference_api.c:1770 Channel sofia/internal/352650 at hasterisk-dev does not have video capability! 2017-05-10 13:33:18.125230 [ALERT] conference_event.c:316 RES [3525236 vid-floor 0002 force][] I've included both the profile and dialplan returned by my web application for mod_xml_curl. Dialplan:
Dialplan(Returned from mod_xml_curl):
From hunterj91 at hotmail.com Wed May 10 22:34:45 2017 From: hunterj91 at hotmail.com (Jonathan Hunter) Date: Wed, 10 May 2017 18:34:45 +0000 Subject: [Freeswitch-users] voicemail_ivr examples when checking voicemail Message-ID: Hey guys, Sorry for the noise! I am looking to use voicemail_ivr so I can ensure my own phrases are played and I can avoid giving the user options to record just one voicemail message (as they have the option to upload a file from our portal too). Can this be achieved using voicemail_ivr and if so are there any examples as confluence is good but doesnt have many configuration examples if you want to miss out phrases etc. Cheers Jon -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170510/70f868ea/attachment-0001.html From mike at jerris.com Wed May 10 22:49:26 2017 From: mike at jerris.com (Michael Jerris) Date: Wed, 10 May 2017 14:49:26 -0400 Subject: [Freeswitch-users] voicemail_ivr examples when checking voicemail In-Reply-To: References: Message-ID: <719512DD-9FEF-45BB-9E25-058B33BD278D@jerris.com> > On May 10, 2017, at 2:34 PM, Jonathan Hunter wrote: > > Hey guys, > > Sorry for the noise! > > I am looking to use voicemail_ivr so I can ensure my own phrases are played and I can avoid giving the user options to record just one voicemail message (as they have the option to upload a file from our portal too). You can do all this with regular voicemail app by modifying the phrases involved and options I think. No need to do your own completely. Take a look at the phrase configs used for voicemail and voicemail config file and let us know where it might be an issue. > > Can this be achieved using voicemail_ivr and if so are there any examples as confluence is good but doesnt have many configuration examples if you want to miss out phrases etc. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170510/64420edc/attachment.html From vladislaus at gmail.com Wed May 10 23:55:06 2017 From: vladislaus at gmail.com (Andres Gomez) Date: Wed, 10 May 2017 14:55:06 -0500 Subject: [Freeswitch-users] BFCP screen sharing In-Reply-To: <009001d2c919$22a093c0$67e1bb40$@freeswitch.org> References: <009001d2c919$22a093c0$67e1bb40$@freeswitch.org> Message-ID: Hi Ken. Libre library support BFCP protocol. Is there any integration between libre library and freeswitch? I see this https://freeswitch.org/stash/projects/SD/repos/libre/browse/src/bfcp/bfcp.h Thanks a lot Carlos On Tue, May 9, 2017 at 6:08 PM, Ken Rice wrote: > FreeSWITCH does not support BFCP at this time? > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Andres Gomez > *Sent:* Tuesday, May 9, 2017 6:03 PM > *To:* FreeSWITCH Users Help > *Subject:* [Freeswitch-users] BFCP screen sharing > > > > Hi Friends > > can I use bfcp screen sharing in videoconferences? > > How can i use ? > > Regards > > Carlos. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170510/b3011440/attachment.html From krice at freeswitch.org Wed May 10 23:59:05 2017 From: krice at freeswitch.org (Ken Rice) Date: Wed, 10 May 2017 14:59:05 -0500 Subject: [Freeswitch-users] BFCP screen sharing In-Reply-To: References: <009001d2c919$22a093c0$67e1bb40$@freeswitch.org> Message-ID: <068c01d2c9c7$e1a668e0$a4f33aa0$@freeswitch.org> not at this time? BFCP is currently not on the roadmap any time soon?. If this is something you might want to look at sponsoring I would contact consulting at freeswitch.org and speak with The FS Team about it. K From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Andres Gomez Sent: Wednesday, May 10, 2017 2:55 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] BFCP screen sharing Hi Ken. Libre library support BFCP protocol. Is there any integration between libre library and freeswitch? I see this https://freeswitch.org/stash/projects/SD/repos/libre/browse/src/bfcp/bfcp.h Thanks a lot Carlos On Tue, May 9, 2017 at 6:08 PM, Ken Rice > wrote: FreeSWITCH does not support BFCP at this time? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org ] On Behalf Of Andres Gomez Sent: Tuesday, May 9, 2017 6:03 PM To: FreeSWITCH Users Help > Subject: [Freeswitch-users] BFCP screen sharing Hi Friends can I use bfcp screen sharing in videoconferences? How can i use ? Regards Carlos. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170510/80f5a03d/attachment.html From vladislaus at gmail.com Thu May 11 02:30:35 2017 From: vladislaus at gmail.com (Andres Gomez) Date: Wed, 10 May 2017 17:30:35 -0500 Subject: [Freeswitch-users] BFCP screen sharing In-Reply-To: <068c01d2c9c7$e1a668e0$a4f33aa0$@freeswitch.org> References: <009001d2c919$22a093c0$67e1bb40$@freeswitch.org> <068c01d2c9c7$e1a668e0$a4f33aa0$@freeswitch.org> Message-ID: Perfect Ken, I'll contact. Regards On Wed, May 10, 2017 at 2:59 PM, Ken Rice wrote: > not at this time? BFCP is currently not on the roadmap any time soon?. If > this is something you might want to look at sponsoring I would contact > consulting at freeswitch.org and speak with The FS Team about it. > > > > K > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Andres Gomez > *Sent:* Wednesday, May 10, 2017 2:55 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] BFCP screen sharing > > > > Hi Ken. > > > > Libre library support BFCP protocol. Is there any integration between > libre library and freeswitch? > > > > I see this https://freeswitch.org/stash/projects/SD/repos/libre/ > browse/src/bfcp/bfcp.h > > > > Thanks a lot > > > > > > Carlos > > > > On Tue, May 9, 2017 at 6:08 PM, Ken Rice wrote: > > FreeSWITCH does not support BFCP at this time? > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Andres Gomez > *Sent:* Tuesday, May 9, 2017 6:03 PM > *To:* FreeSWITCH Users Help > *Subject:* [Freeswitch-users] BFCP screen sharing > > > > Hi Friends > > can I use bfcp screen sharing in videoconferences? > > How can i use ? > > Regards > > Carlos. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170510/79256cb1/attachment-0001.html From thetsinling at outlook.com Thu May 11 07:02:51 2017 From: thetsinling at outlook.com (bob. chen) Date: Thu, 11 May 2017 03:02:51 +0000 Subject: [Freeswitch-users] Setting up webrtc certificates In-Reply-To: References: Message-ID: https://freeswitch.org/confluence/display/FREESWITCH/WebRTC#WebRTC-InstallCertificates From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ravi sanyal Sent: Wednesday, May 10, 2017 8:26 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Setting up webrtc certificates Hello, I'm following the guide on confluence to setup webrtc and testing the various ports for certificate information. I noticed that i need to copy the ca.crt file into the certs folder, however, i used letsencrypt over the self-signed program that was being used for the example. What is the equivalent of this file in letsencrypt? Nathan -- Virtual Guard Ltd info at virtualguard.co.nz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170511/6caae90c/attachment.html From richard at treeboxsolutions.com Thu May 11 17:22:58 2017 From: richard at treeboxsolutions.com (Richard Chan) Date: Thu, 11 May 2017 21:22:58 +0800 Subject: [Freeswitch-users] Why two ilbc(ilbc2) RPMs in CentOS 7 repos Message-ID: Hi, I was wondering what the purpose is of having two RPMs for ilbc, named ilbc and ilbc2. Does the build system check for both? In Fedora (not sure for CentOS) there is an incompatible RPM so perhaps to avoid a package name clash? -- Richard Chan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170511/31752f4b/attachment.html From colton.conor at gmail.com Thu May 11 18:35:44 2017 From: colton.conor at gmail.com (Colton Conor) Date: Thu, 11 May 2017 09:35:44 -0500 Subject: [Freeswitch-users] Freeswitch PRI support In-Reply-To: References: <27a9d063.ad3b.15bd85e9762.Coremail.xxxman2008@126.com> <6fc1c046-45f9-cdbe-2896-a077dbcc5dad@xbipin.com> Message-ID: What is the value? Mobile voice plans are expensive (realtive to sip trunks). Also, call quality is not even G711 in most cases. I could see the use case is you wanted a mobile backup with GSM for voice in the case your hardline internet were down, but even then I would think it would be cheaper and more benefical to get a mobile data backup plan, and continue to use the SIP trunks over the internet. I honestly had now clue these large GSM banks existed. Anyone used them in North America? On Wed, May 10, 2017 at 10:06 AM, David Villasmil < david.villasmil.work at gmail.com> wrote: > I think it's illegal everywhere... but value? there's lots and lots of > values, it only depends on where you are doing it... :) > ? > > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 <+34%20669%2044%2083%2037> > > On Wed, May 10, 2017 at 4:12 PM, Ra?l Alexis Betancor Santana < > rbetancor at gmail.com> wrote: > >> The only real advantage over a regular SIP trunk from any provider, it's >> that you could select a plan than MAYBE allow you some sort of flat-rate >> and not per-minute use plans ... but it doesn't worth the effort and money >> you have to spend to get that to generate revenue ... also most of celullar >> carriers withh put some restrictions on the usage of the flat-rate, as all >> of them have 'fair use' policies. >> At least on Spain, it's ilegal to resell that kind of traffic and I asume >> its the same in most of the world. >> >> 2017-05-10 15:04 GMT+01:00 Colton Conor : >> >>> So how does this GSM stuff work in the USA? Would I got buy a cell phone >>> plan from someone like AT&T or T-Mobile, and then insert the SIM into these >>> devices. The device would then covert the audio from GSM to SIP, and then I >>> could use it like a SIP trunk? >>> >>> What is the advantage of doing this over just buying a pure SIP trunk >>> from an internet provider? >>> >>> >>> >>> On Wed, May 10, 2017 at 4:35 AM, David Villasmil < >>> david.villasmil.work at gmail.com> wrote: >>> >>>> There's another chinese brand that has a 32 port, I've used it, works >>>> pretty well, i think it's "GoIP" or something. >>>> >>>> On Wed, May 10, 2017 at 10:39 AM Bipin Patel wrote: >>>> >>>>> hi, >>>>> >>>>> we use matrix boxes but 4 port ones, they work quiet well >>>>> >>>>> >>>>> Regards, >>>>> Bipin >>>>> >>>>> >>>>> ------------------------------ >>>>> -------- Original Message -------- >>>>> Subject: Re: [Freeswitch-users] Freeswitch PRI support >>>>> From: Deepika Yadav >>>>> To: FreeSWITCH Users Help >>>>> >>>>> Date: 5/10/2017, 11:02:49 AM >>>>> >>>>> Hi, >>>>> >>>>> I searched some gateways online. The ones available at ebay are cheap, >>>>> the following link shows a gateway supporting 32 SIMS for $859 : >>>>> >>>>> http://www.ebay.com/itm/GoIP-32-32-Port-VoIP-GSM-Gateway-wit >>>>> h-32-External-Antenna-/291811535942 >>>>> >>>>> However, when I contacted a company called "Matrix Telecom Solution", >>>>> they gave the quote of $4150 for similar number of SIM support. On asking >>>>> them to compare the two gateways, they said that the one available at ebay >>>>> is a Chinese gateway about which they are apprehensive for the quality, >>>>> warranty and working. >>>>> >>>>> Regards, >>>>> Deepika >>>>> >>>>> >>>>> On Fri, May 5, 2017 at 4:58 PM, Raymond wrote: >>>>> >>>>>> >>>>>> HI, >>>>>> >>>>>> Regarding this. i have asked some question of module "GSMOpen". >>>>>> >>>>>> In fact ,i want to make Freeswitch box as "GSM VOIP Gateway" by >>>>>> using "gsmopen". But...seems no one help me . :-(( . If it worked , can >>>>>> save more money than hardware gsm voip gateway ( GOIP ). >>>>>> >>>>>> Raymond >>>>>> >>>>>> >>>>>> >>>>>> At 2017-05-04 17:24:13, "Giovanni Maruzzelli" >>>>>> wrote: >>>>>> >>>>>> For PRI you use Sangoma or Patton. >>>>>> >>>>>> But, why don't you use an hardware gateway sip<->gsm? >>>>>> >>>>>> It would save you very big money. >>>>>> >>>>>> Check on ebay and google, there are many of them, you put SIMs >>>>>> inside, and you are good to go. >>>>>> >>>>>> -giovanni >>>>>> >>>>>> >>>>>> sent from mobile >>>>>> cell: +39 347 266 56 18 >>>>>> Giovanni Maruzzelli >>>>>> OpenTelecom.IT >>>>>> >>>>>> Il 04 mag 2017 11:13, "Deepika Yadav" ha >>>>>> scritto: >>>>>> >>>>>>> Hi, >>>>>>> >>>>>>> I am using Freeswitch in an application that initiates conference >>>>>>> calls amongst 20 users in cellular network (mobile phones in GSM network). >>>>>>> Currently, for the VOIP-GSM gateway, we are using the service from a >>>>>>> company called Doorvaani. >>>>>>> >>>>>>> But, since, the cost of call estimates to be high and we cannot >>>>>>> debug the call drops; we are thinking to buy our own PRI card. >>>>>>> >>>>>>> I am seeking recommendation on following points: >>>>>>> >>>>>>> 1. Which card should I buy that is most easily configurable with >>>>>>> the Freeswitch i.e. company and type. >>>>>>> 2. Reference on how should I start to make Freeswitch configure >>>>>>> with the PRI card and start sending and receiving calls. For the current >>>>>>> gateway service in use, I simply put the authentication credentials for the >>>>>>> corresponding VOIP line offered by the company Doorvaani in the external >>>>>>> SIP Profile. In case of PRI, what all changes are needed? >>>>>>> >>>>>>> Regards, >>>>>>> Deepika >>>>>>> >>>>>>> -- >>>>>>> Regards >>>>>>> Deepika >>>>>>> https://www.iiitd.edu.in/~deepikay/ >>>>>>> >>>>>>> ____________________________________________________________ >>>>>>> _____________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://confluence.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-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 >>>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> Regards >>>>> Deepika >>>>> https://www.iiitd.edu.in/~deepikay/ >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> 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/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 >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170511/eab00fc3/attachment-0001.html From mike at jerris.com Thu May 11 18:39:43 2017 From: mike at jerris.com (Michael Jerris) Date: Thu, 11 May 2017 14:39:43 +0000 Subject: [Freeswitch-users] Why two ilbc(ilbc2) RPMs in CentOS 7 repos In-Reply-To: References: Message-ID: ilbc2 is the right one, configure checks for both, it is to avoid the name collision. not sure why they would both be there, ilbc probably shouldn't be. On Thu, May 11, 2017 at 9:27 AM Richard Chan wrote: > Hi, > > I was wondering what the purpose is of having two RPMs for ilbc, named > ilbc and ilbc2. > > Does the build system check for both? In Fedora (not sure for CentOS) > there is an incompatible RPM so perhaps to avoid a package name clash? > > -- > 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170511/b7860e1b/attachment.html From dig1234 at gmail.com Wed May 10 23:38:00 2017 From: dig1234 at gmail.com (Daniel Greenwald) Date: Wed, 10 May 2017 15:38:00 -0400 Subject: [Freeswitch-users] [OpenSIPS-Users] TLS SIP packet tracing and visualization In-Reply-To: References: Message-ID: Thanks for this script! Theoretically it is possible to see TLS SIP traffic with freeswitch sending HEP to Homer. But there seems to be a bug in FS that only sends one side of SIP conversation (ie the FS side, not inbound messages).. On Tue, May 9, 2017 at 11:10 AM, Giovanni Maruzzelli wrote: > On 9 May 2017 at 15:18, Bogdan-Andrei Iancu wrote: > >> Thank you Giovanni, that is a useful tool - we will document it in the >> OpenSIPS TLS tutorial, so other can benefit ;) >> >> > > Glad about it! > Be sure to get it from https://freeswitch.org/conflue > nce/display/FREESWITCH/Packet+Capture#PacketCapture-TLSwithsharka , is > the latest version with a couple fixes. > > -giovanni > > > > >> Many thanks, >> >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> >> OpenSIPS Summit May 2017 Amsterdam >> http://www.opensips.org/events/Summit-2017Amsterdam.html >> >> On 05/02/2017 05:52 PM, Giovanni Maruzzelli wrote: >> >> For a cut and paste ready version, that has the correct carriage returns >> (mangled by mail), check it in FreeSWITCH documentation: >> >> https://freeswitch.org/confluence/display/FREESWITCH/Packet+ >> Capture#PacketCapture-TLSwithsharka >> >> -giovanni >> >> On 2 May 2017 at 16:26, Giovanni Maruzzelli wrote: >> >>> Hello fellows, >>> >>> after some experimentation with various tools, I come out with a little >>> shell tool that maybe can be useful to you too. >>> >>> It can only work with non-forward secrecy ciphers, obviously, and only >>> if is started before the client do the initial TLS handshake (eg, just >>> restart the client). Forward secrecy cannot be decrypted after fact, so >>> don't waste effort. >>> >>> An example of ciphers that can be decrypted are the "AES256-SHA" openssl >>> cipher group. You can use ssldump to check what cipher is used by >>> serverhello. >>> >>> Enjoy, make it better, and share it :) >>> >>> >>> #!/bin/bash >>> # brought to you by Giovanni Maruzzelli >>> # >>> SERVERIP="192.168.1.150" >>> SERVERPORT="5061" >>> PRIVKEY="/etc/certs/privkey.pem" >>> STDERR2DEVNULL=" 2>/dev/null " >>> REGEX="notyet" >>> >>> if [ -z "$1" ]; then >>> REGEX="\\\.*" >>> else >>> REGEX="$1" >>> fi >>> FILTER="ssl.app_data and sip matches" >>> FILTER2="$FILTER \"$REGEX\"" >>> FILTER3="'$FILTER2'" >>> ARGUMENT="-i 1 -Y $FILTER3 -E header=y -T fields -e frame.number -e >>> frame.time -e frame.time_delta_displayed -e ip.src -e ip.dst -e >>> sip.Status-Line -e sip.Request-Line -e sip.msg_hdr -l -d >>> tcp.port\=\=5061,sip -o \"ssl.keys_list: $SERVERIP,$SERVERPORT,sip,$PRIVKEY\" >>> $STDERR2DEVNULL | sed -u 's/\t/\n/g' | sed -u '/^$/d' | sed -u >>> 's/^[0-9]*$/\n==&==============================/g'" >>> >>> echo "" >>> echo "NB: if it do not works, edit script so that STDERR2DEVNULL=\" \" >>> and try again" >>> echo "" >>> echo "NB: remember to quote and escape match patterns, using triple >>> slash" >>> echo " eg, for matching 1010 at pbx.example.com, use \" >>> 1010 at pbx.example.com\"" >>> echo " eg, for matching anything, use \"\\\\\\.*\"" >>> echo " eg, for matching *98, use \"\\\\\\*98\"" >>> echo "USAGE: $0 \"\\\\\\*98 at pbx.example.com\"" >>> echo "" >>> >>> >>> case "$1" in >>> -help|--help|?) >>> exit 0 >>> ;; >>> *) >>> echo "THIS TIME WE'RE DOING:" >>> echo "tshark $ARGUMENT" >>> echo "" >>> bash -c "tshark $ARGUMENT" >>> ;; >>> esac >>> >>> >>> >>> -- >>> >>> Sincerely, >>> >>> Giovanni Maruzzelli >>> OpenTelecom.IT >>> cell: +39 347 266 56 18 >>> >> >> >> >> -- >> >> Sincerely, >> >> Giovanni Maruzzelli >> OpenTelecom.IT >> cell: +39 347 266 56 18 >> >> >> _______________________________________________ >> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> > > > -- > > Sincerely, > > Giovanni Maruzzelli > OpenTelecom.IT > cell: +39 347 266 56 18 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170510/07b5aa89/attachment.html From petrpwp at gmail.com Wed May 10 18:19:24 2017 From: petrpwp at gmail.com (Pete Procenko) Date: Wed, 10 May 2017 17:19:24 +0300 Subject: [Freeswitch-users] Setting up webrtc certificates In-Reply-To: References: Message-ID: Hi! https://www.identrust.com/certificates/trustid/root-download-x3.html this is a cert content, it should be between -----BEGIN CERTIFICATE----- content from https://www.identrust.com/certificates/trustid/root- download-x3.html here -----END CERTIFICATE----- 2017-05-10 3:26 GMT+03:00 Ravi sanyal : > Hello, > > I'm following the guide on confluence to setup webrtc and testing the > various ports for certificate information. I noticed that i need to copy > the ca.crt file into the certs folder, however, i used letsencrypt over the > self-signed program that was being used for the example. What is the > equivalent of this file in letsencrypt? > > Nathan > > -- > > *Virtual Guard Ltd* > *info at virtualguard.co.nz * > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170510/6690ddbf/attachment-0001.html From paranoya at paranoya.org Thu May 11 18:59:55 2017 From: paranoya at paranoya.org (Jakub Wojewoda) Date: Thu, 11 May 2017 16:59:55 +0200 Subject: [Freeswitch-users] Passing variables directly in mod_lcr Message-ID: Anyone knows how to import custom variable in mod_lcr? I've got custom query with custom variable "custom_var" in lcr.xml, and export it in export_fields. I can see it in ${lcr_auto_route} as[custom_variable=xxxx ...]gateway when I call . But I want to see it as ${custom_var}. Any ideas? Regards, Jakub Wojewoda -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170511/b4e29075/attachment.html From deepikay at iiitd.ac.in Thu May 11 19:31:21 2017 From: deepikay at iiitd.ac.in (Deepika Yadav) Date: Thu, 11 May 2017 21:01:21 +0530 Subject: [Freeswitch-users] Freeswitch PRI support In-Reply-To: References: <27a9d063.ad3b.15bd85e9762.Coremail.xxxman2008@126.com> <6fc1c046-45f9-cdbe-2896-a077dbcc5dad@xbipin.com> Message-ID: This is for the usecase of community heath worker in rural areas of India who only have access to feature phones i.e. no Internet, no data plan. Regards, Deepika On Thu, May 11, 2017 at 8:05 PM, Colton Conor wrote: > What is the value? Mobile voice plans are expensive (realtive to sip > trunks). Also, call quality is not even G711 in most cases. > > I could see the use case is you wanted a mobile backup with GSM for voice > in the case your hardline internet were down, but even then I would think > it would be cheaper and more benefical to get a mobile data backup plan, > and continue to use the SIP trunks over the internet. > > I honestly had now clue these large GSM banks existed. > > Anyone used them in North America? > > On Wed, May 10, 2017 at 10:06 AM, David Villasmil < > david.villasmil.work at gmail.com> wrote: > >> I think it's illegal everywhere... but value? there's lots and lots of >> values, it only depends on where you are doing it... :) >> ? >> >> Regards, >> >> David Villasmil >> email: david.villasmil.work at gmail.com >> phone: +34669448337 <+34%20669%2044%2083%2037> >> >> On Wed, May 10, 2017 at 4:12 PM, Ra?l Alexis Betancor Santana < >> rbetancor at gmail.com> wrote: >> >>> The only real advantage over a regular SIP trunk from any provider, it's >>> that you could select a plan than MAYBE allow you some sort of flat-rate >>> and not per-minute use plans ... but it doesn't worth the effort and money >>> you have to spend to get that to generate revenue ... also most of celullar >>> carriers withh put some restrictions on the usage of the flat-rate, as all >>> of them have 'fair use' policies. >>> At least on Spain, it's ilegal to resell that kind of traffic and I >>> asume its the same in most of the world. >>> >>> 2017-05-10 15:04 GMT+01:00 Colton Conor : >>> >>>> So how does this GSM stuff work in the USA? Would I got buy a cell >>>> phone plan from someone like AT&T or T-Mobile, and then insert the SIM into >>>> these devices. The device would then covert the audio from GSM to SIP, and >>>> then I could use it like a SIP trunk? >>>> >>>> What is the advantage of doing this over just buying a pure SIP trunk >>>> from an internet provider? >>>> >>>> >>>> >>>> On Wed, May 10, 2017 at 4:35 AM, David Villasmil < >>>> david.villasmil.work at gmail.com> wrote: >>>> >>>>> There's another chinese brand that has a 32 port, I've used it, works >>>>> pretty well, i think it's "GoIP" or something. >>>>> >>>>> On Wed, May 10, 2017 at 10:39 AM Bipin Patel wrote: >>>>> >>>>>> hi, >>>>>> >>>>>> we use matrix boxes but 4 port ones, they work quiet well >>>>>> >>>>>> >>>>>> Regards, >>>>>> Bipin >>>>>> >>>>>> >>>>>> ------------------------------ >>>>>> -------- Original Message -------- >>>>>> Subject: Re: [Freeswitch-users] Freeswitch PRI support >>>>>> From: Deepika Yadav >>>>>> To: FreeSWITCH Users Help >>>>>> >>>>>> Date: 5/10/2017, 11:02:49 AM >>>>>> >>>>>> Hi, >>>>>> >>>>>> I searched some gateways online. The ones available at ebay are >>>>>> cheap, the following link shows a gateway supporting 32 SIMS for $859 : >>>>>> >>>>>> http://www.ebay.com/itm/GoIP-32-32-Port-VoIP-GSM-Gateway-wit >>>>>> h-32-External-Antenna-/291811535942 >>>>>> >>>>>> However, when I contacted a company called "Matrix Telecom Solution", >>>>>> they gave the quote of $4150 for similar number of SIM support. On asking >>>>>> them to compare the two gateways, they said that the one available at ebay >>>>>> is a Chinese gateway about which they are apprehensive for the quality, >>>>>> warranty and working. >>>>>> >>>>>> Regards, >>>>>> Deepika >>>>>> >>>>>> >>>>>> On Fri, May 5, 2017 at 4:58 PM, Raymond wrote: >>>>>> >>>>>>> >>>>>>> HI, >>>>>>> >>>>>>> Regarding this. i have asked some question of module "GSMOpen". >>>>>>> >>>>>>> In fact ,i want to make Freeswitch box as "GSM VOIP Gateway" by >>>>>>> using "gsmopen". But...seems no one help me . :-(( . If it worked , can >>>>>>> save more money than hardware gsm voip gateway ( GOIP ). >>>>>>> >>>>>>> Raymond >>>>>>> >>>>>>> >>>>>>> >>>>>>> At 2017-05-04 17:24:13, "Giovanni Maruzzelli" >>>>>>> wrote: >>>>>>> >>>>>>> For PRI you use Sangoma or Patton. >>>>>>> >>>>>>> But, why don't you use an hardware gateway sip<->gsm? >>>>>>> >>>>>>> It would save you very big money. >>>>>>> >>>>>>> Check on ebay and google, there are many of them, you put SIMs >>>>>>> inside, and you are good to go. >>>>>>> >>>>>>> -giovanni >>>>>>> >>>>>>> >>>>>>> sent from mobile >>>>>>> cell: +39 347 266 56 18 >>>>>>> Giovanni Maruzzelli >>>>>>> OpenTelecom.IT >>>>>>> >>>>>>> Il 04 mag 2017 11:13, "Deepika Yadav" ha >>>>>>> scritto: >>>>>>> >>>>>>>> Hi, >>>>>>>> >>>>>>>> I am using Freeswitch in an application that initiates conference >>>>>>>> calls amongst 20 users in cellular network (mobile phones in GSM network). >>>>>>>> Currently, for the VOIP-GSM gateway, we are using the service from a >>>>>>>> company called Doorvaani. >>>>>>>> >>>>>>>> But, since, the cost of call estimates to be high and we cannot >>>>>>>> debug the call drops; we are thinking to buy our own PRI card. >>>>>>>> >>>>>>>> I am seeking recommendation on following points: >>>>>>>> >>>>>>>> 1. Which card should I buy that is most easily configurable >>>>>>>> with the Freeswitch i.e. company and type. >>>>>>>> 2. Reference on how should I start to make Freeswitch configure >>>>>>>> with the PRI card and start sending and receiving calls. For the current >>>>>>>> gateway service in use, I simply put the authentication credentials for the >>>>>>>> corresponding VOIP line offered by the company Doorvaani in the external >>>>>>>> SIP Profile. In case of PRI, what all changes are needed? >>>>>>>> >>>>>>>> Regards, >>>>>>>> Deepika >>>>>>>> >>>>>>>> -- >>>>>>>> Regards >>>>>>>> Deepika >>>>>>>> https://www.iiitd.edu.in/~deepikay/ >>>>>>>> >>>>>>>> ____________________________________________________________ >>>>>>>> _____________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting at freeswitch.org >>>>>>>> http://www.freeswitchsolutions.com >>>>>>>> >>>>>>>> Official FreeSWITCH Sites >>>>>>>> http://www.freeswitch.org >>>>>>>> http://confluence.freeswitch.org >>>>>>>> http://www.cluecon.com >>>>>>>> >>>>>>>> FreeSWITCH-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 >>>>>>> >>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> Regards >>>>>> Deepika >>>>>> https://www.iiitd.edu.in/~deepikay/ >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> 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/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/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 > -- Regards Deepika https://www.iiitd.edu.in/~deepikay/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170511/9932aab0/attachment-0001.html From gmaruzz at gmail.com Thu May 11 19:43:31 2017 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Thu, 11 May 2017 17:43:31 +0200 Subject: [Freeswitch-users] [OpenSIPS-Users] TLS SIP packet tracing and visualization In-Reply-To: References: Message-ID: On 10 May 2017 at 21:38, Daniel Greenwald wrote: > Thanks for this script! > Theoretically it is possible to see TLS SIP traffic with freeswitch > sending HEP to Homer. But there seems to be a bug in FS that only sends one > side of SIP conversation (ie the FS side, not inbound messages).. > > Daniel, have you had this problem yourself, or just heard about? Have you opened a jira issue for it? If you encountered this problem, please report it in Jira. Please never report bugs in mailing list only, they will be lost and forgot. Jira is how we manage bugs ;) -giovanni > On Tue, May 9, 2017 at 11:10 AM, Giovanni Maruzzelli > wrote: > >> On 9 May 2017 at 15:18, Bogdan-Andrei Iancu wrote: >> >>> Thank you Giovanni, that is a useful tool - we will document it in the >>> OpenSIPS TLS tutorial, so other can benefit ;) >>> >>> >> >> Glad about it! >> Be sure to get it from https://freeswitch.org/conflue >> nce/display/FREESWITCH/Packet+Capture#PacketCapture-TLSwithsharka , is >> the latest version with a couple fixes. >> >> -giovanni >> >> >> >> >>> Many thanks, >>> >>> Bogdan-Andrei Iancu >>> OpenSIPS Founder and Developer >>> http://www.opensips-solutions.com >>> >>> OpenSIPS Summit May 2017 Amsterdam >>> http://www.opensips.org/events/Summit-2017Amsterdam.html >>> >>> On 05/02/2017 05:52 PM, Giovanni Maruzzelli wrote: >>> >>> For a cut and paste ready version, that has the correct carriage returns >>> (mangled by mail), check it in FreeSWITCH documentation: >>> >>> https://freeswitch.org/confluence/display/FREESWITCH/Packet+ >>> Capture#PacketCapture-TLSwithsharka >>> >>> -giovanni >>> >>> On 2 May 2017 at 16:26, Giovanni Maruzzelli wrote: >>> >>>> Hello fellows, >>>> >>>> after some experimentation with various tools, I come out with a little >>>> shell tool that maybe can be useful to you too. >>>> >>>> It can only work with non-forward secrecy ciphers, obviously, and only >>>> if is started before the client do the initial TLS handshake (eg, just >>>> restart the client). Forward secrecy cannot be decrypted after fact, so >>>> don't waste effort. >>>> >>>> An example of ciphers that can be decrypted are the "AES256-SHA" >>>> openssl cipher group. You can use ssldump to check what cipher is used by >>>> serverhello. >>>> >>>> Enjoy, make it better, and share it :) >>>> >>>> >>>> #!/bin/bash >>>> # brought to you by Giovanni Maruzzelli >>>> # >>>> SERVERIP="192.168.1.150" >>>> SERVERPORT="5061" >>>> PRIVKEY="/etc/certs/privkey.pem" >>>> STDERR2DEVNULL=" 2>/dev/null " >>>> REGEX="notyet" >>>> >>>> if [ -z "$1" ]; then >>>> REGEX="\\\.*" >>>> else >>>> REGEX="$1" >>>> fi >>>> FILTER="ssl.app_data and sip matches" >>>> FILTER2="$FILTER \"$REGEX\"" >>>> FILTER3="'$FILTER2'" >>>> ARGUMENT="-i 1 -Y $FILTER3 -E header=y -T fields -e frame.number -e >>>> frame.time -e frame.time_delta_displayed -e ip.src -e ip.dst -e >>>> sip.Status-Line -e sip.Request-Line -e sip.msg_hdr -l -d >>>> tcp.port\=\=5061,sip -o \"ssl.keys_list: $SERVERIP,$SERVERPORT,sip,$PRIVKEY\" >>>> $STDERR2DEVNULL | sed -u 's/\t/\n/g' | sed -u '/^$/d' | sed -u >>>> 's/^[0-9]*$/\n==&==============================/g'" >>>> >>>> echo "" >>>> echo "NB: if it do not works, edit script so that STDERR2DEVNULL=\" \" >>>> and try again" >>>> echo "" >>>> echo "NB: remember to quote and escape match patterns, using triple >>>> slash" >>>> echo " eg, for matching 1010 at pbx.example.com, use \" >>>> 1010 at pbx.example.com\"" >>>> echo " eg, for matching anything, use \"\\\\\\.*\"" >>>> echo " eg, for matching *98, use \"\\\\\\*98\"" >>>> echo "USAGE: $0 \"\\\\\\*98 at pbx.example.com\"" >>>> echo "" >>>> >>>> >>>> case "$1" in >>>> -help|--help|?) >>>> exit 0 >>>> ;; >>>> *) >>>> echo "THIS TIME WE'RE DOING:" >>>> echo "tshark $ARGUMENT" >>>> echo "" >>>> bash -c "tshark $ARGUMENT" >>>> ;; >>>> esac >>>> >>>> >>>> >>>> -- >>>> >>>> Sincerely, >>>> >>>> Giovanni Maruzzelli >>>> OpenTelecom.IT >>>> cell: +39 347 266 56 18 >>>> >>> >>> >>> >>> -- >>> >>> Sincerely, >>> >>> Giovanni Maruzzelli >>> OpenTelecom.IT >>> cell: +39 347 266 56 18 >>> >>> >>> _______________________________________________ >>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >> >> >> -- >> >> Sincerely, >> >> Giovanni Maruzzelli >> OpenTelecom.IT >> cell: +39 347 266 56 18 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170511/6a4e6e52/attachment.html From jalsot at gmail.com Thu May 11 19:45:53 2017 From: jalsot at gmail.com (Tamas Jalsovszky) Date: Thu, 11 May 2017 17:45:53 +0200 Subject: [Freeswitch-users] [OpenSIPS-Users] TLS SIP packet tracing and visualization In-Reply-To: References: Message-ID: Does HEP send verto signaling info to Homer too? On 10 May 2017 at 21:38, Daniel Greenwald wrote: > Thanks for this script! > Theoretically it is possible to see TLS SIP traffic with freeswitch > sending HEP to Homer. But there seems to be a bug in FS that only sends one > side of SIP conversation (ie the FS side, not inbound messages).. > > On Tue, May 9, 2017 at 11:10 AM, Giovanni Maruzzelli > wrote: > >> On 9 May 2017 at 15:18, Bogdan-Andrei Iancu wrote: >> >>> Thank you Giovanni, that is a useful tool - we will document it in the >>> OpenSIPS TLS tutorial, so other can benefit ;) >>> >>> >> >> Glad about it! >> Be sure to get it from https://freeswitch.org/conflue >> nce/display/FREESWITCH/Packet+Capture#PacketCapture-TLSwithsharka , is >> the latest version with a couple fixes. >> >> -giovanni >> >> >> >> >>> Many thanks, >>> >>> Bogdan-Andrei Iancu >>> OpenSIPS Founder and Developer >>> http://www.opensips-solutions.com >>> >>> OpenSIPS Summit May 2017 Amsterdam >>> http://www.opensips.org/events/Summit-2017Amsterdam.html >>> >>> On 05/02/2017 05:52 PM, Giovanni Maruzzelli wrote: >>> >>> For a cut and paste ready version, that has the correct carriage returns >>> (mangled by mail), check it in FreeSWITCH documentation: >>> >>> https://freeswitch.org/confluence/display/FREESWITCH/Packet+ >>> Capture#PacketCapture-TLSwithsharka >>> >>> -giovanni >>> >>> On 2 May 2017 at 16:26, Giovanni Maruzzelli wrote: >>> >>>> Hello fellows, >>>> >>>> after some experimentation with various tools, I come out with a little >>>> shell tool that maybe can be useful to you too. >>>> >>>> It can only work with non-forward secrecy ciphers, obviously, and only >>>> if is started before the client do the initial TLS handshake (eg, just >>>> restart the client). Forward secrecy cannot be decrypted after fact, so >>>> don't waste effort. >>>> >>>> An example of ciphers that can be decrypted are the "AES256-SHA" >>>> openssl cipher group. You can use ssldump to check what cipher is used by >>>> serverhello. >>>> >>>> Enjoy, make it better, and share it :) >>>> >>>> >>>> #!/bin/bash >>>> # brought to you by Giovanni Maruzzelli >>>> # >>>> SERVERIP="192.168.1.150" >>>> SERVERPORT="5061" >>>> PRIVKEY="/etc/certs/privkey.pem" >>>> STDERR2DEVNULL=" 2>/dev/null " >>>> REGEX="notyet" >>>> >>>> if [ -z "$1" ]; then >>>> REGEX="\\\.*" >>>> else >>>> REGEX="$1" >>>> fi >>>> FILTER="ssl.app_data and sip matches" >>>> FILTER2="$FILTER \"$REGEX\"" >>>> FILTER3="'$FILTER2'" >>>> ARGUMENT="-i 1 -Y $FILTER3 -E header=y -T fields -e frame.number -e >>>> frame.time -e frame.time_delta_displayed -e ip.src -e ip.dst -e >>>> sip.Status-Line -e sip.Request-Line -e sip.msg_hdr -l -d >>>> tcp.port\=\=5061,sip -o \"ssl.keys_list: $SERVERIP,$SERVERPORT,sip,$PRIVKEY\" >>>> $STDERR2DEVNULL | sed -u 's/\t/\n/g' | sed -u '/^$/d' | sed -u >>>> 's/^[0-9]*$/\n==&==============================/g'" >>>> >>>> echo "" >>>> echo "NB: if it do not works, edit script so that STDERR2DEVNULL=\" \" >>>> and try again" >>>> echo "" >>>> echo "NB: remember to quote and escape match patterns, using triple >>>> slash" >>>> echo " eg, for matching 1010 at pbx.example.com, use \" >>>> 1010 at pbx.example.com\"" >>>> echo " eg, for matching anything, use \"\\\\\\.*\"" >>>> echo " eg, for matching *98, use \"\\\\\\*98\"" >>>> echo "USAGE: $0 \"\\\\\\*98 at pbx.example.com\"" >>>> echo "" >>>> >>>> >>>> case "$1" in >>>> -help|--help|?) >>>> exit 0 >>>> ;; >>>> *) >>>> echo "THIS TIME WE'RE DOING:" >>>> echo "tshark $ARGUMENT" >>>> echo "" >>>> bash -c "tshark $ARGUMENT" >>>> ;; >>>> esac >>>> >>>> >>>> >>>> -- >>>> >>>> Sincerely, >>>> >>>> Giovanni Maruzzelli >>>> OpenTelecom.IT >>>> cell: +39 347 266 56 18 >>>> >>> >>> >>> >>> -- >>> >>> Sincerely, >>> >>> Giovanni Maruzzelli >>> OpenTelecom.IT >>> cell: +39 347 266 56 18 >>> >>> >>> _______________________________________________ >>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >> >> >> -- >> >> Sincerely, >> >> Giovanni Maruzzelli >> OpenTelecom.IT >> cell: +39 347 266 56 18 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170511/67347d89/attachment-0001.html From krice at freeswitch.org Thu May 11 20:04:50 2017 From: krice at freeswitch.org (Ken Rice) Date: Thu, 11 May 2017 11:04:50 -0500 Subject: [Freeswitch-users] Setting up webrtc certificates In-Reply-To: References: Message-ID: <0aae01d2ca70$52444740$f6ccd5c0$@freeswitch.org> When using LetsEncrypt and certbot, its as simple as export FQDN=my.fqdn.com; cat /etc/letsencrypt/live/$FQDN/cert.pem /etc/letsencrypt/live/$FQDN/privkey.pem \ /etc/letsencrypt/live/$FQDN/chain.pem > /usr/local/freeswitch/certs/wss.pem This works if you have installed from source, If you have installed from packages change /usr/local/freeswitch? bit to /etc/freeswitch/tls/wss.pem From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ravi sanyal Sent: Tuesday, May 9, 2017 7:26 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Setting up webrtc certificates Hello, I'm following the guide on confluence to setup webrtc and testing the various ports for certificate information. I noticed that i need to copy the ca.crt file into the certs folder, however, i used letsencrypt over the self-signed program that was being used for the example. What is the equivalent of this file in letsencrypt? Nathan -- Virtual Guard Ltd info at virtualguard.co.nz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170511/2d2d1af5/attachment.html From richard at treeboxsolutions.com Thu May 11 20:36:41 2017 From: richard at treeboxsolutions.com (Richard Chan) Date: Fri, 12 May 2017 00:36:41 +0800 Subject: [Freeswitch-users] Why two ilbc(ilbc2) RPMs in CentOS 7 repos In-Reply-To: References: Message-ID: Hi Michael, shall I file a "bug" against this? Thanks Richard On Thu, May 11, 2017 at 10:39 PM, Michael Jerris wrote: > ilbc2 is the right one, configure checks for both, it is to avoid the name > collision. not sure why they would > both be there, ilbc probably shouldn't be. > > On Thu, May 11, 2017 at 9:27 AM Richard Chan > wrote: > >> Hi, >> >> I was wondering what the purpose is of having two RPMs for ilbc, named >> ilbc and ilbc2. >> >> Does the build system check for both? In Fedora (not sure for CentOS) >> there is an incompatible RPM so perhaps to avoid a package name clash? >> >> -- >> 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Richard Chan Chief Architect TreeBox Solutions Pte Ltd 1 Commonwealth Lane #03-01 Singapore 149544 Tel: 6570 3725 http://www.treeboxsolutions.com Co.Reg.No. 201100585R -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170512/ec686750/attachment.html From cjbujold at accra.ca Thu May 11 23:41:54 2017 From: cjbujold at accra.ca (Charles Bujold) Date: Thu, 11 May 2017 16:41:54 -0300 Subject: [Freeswitch-users] Freeswitch Packet Fragmentation issue Message-ID: <000201d2ca8e$a53caeb0$efb60c10$@accra.ca> Upgraded to the latest Freeswitch and now experiencing an issue with packet sizes being bigger than the 1500 allowed. The sip INVITE is 1577 bytes. Tried compress-header but it is not removing enough bytes. At the end of the Invoice there is the X-FS-Support and Remote-Party-ID which I tried to remove but the best I can do is shrink it. Also, the Contact section seems to duplicate the same information, is there a way to cut that down? How can I reduce the fragmentation? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170511/ec5a0286/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 74651 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170511/ec5a0286/attachment-0001.jpe From anthony.minessale at gmail.com Thu May 11 23:57:08 2017 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 11 May 2017 14:57:08 -0500 Subject: [Freeswitch-users] Freeswitch Packet Fragmentation issue In-Reply-To: <000201d2ca8e$a53caeb0$efb60c10$@accra.ca> References: <000201d2ca8e$a53caeb0$efb60c10$@accra.ca> Message-ID: Its probably the SDP. You can use less codecs otherwise the SPEC says you have to use TCP for packets over the MTU On Thu, May 11, 2017 at 2:41 PM, Charles Bujold wrote: > Upgraded to the latest Freeswitch and now experiencing an issue with > packet sizes being bigger than the 1500 allowed. > > > > The sip INVITE is 1577 bytes. Tried compress-header but it is not > removing enough bytes. At the end of the Invoice there is the X-FS-Support > and Remote-Party-ID which I tried to remove but the best I can do is shrink > it. Also, the Contact section seems to duplicate the same information, is > there a way to cut that down? > > > > How can I reduce the fragmentation? > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170511/c839dca8/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image003.jpg Type: image/jpeg Size: 74651 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170511/c839dca8/attachment-0001.jpg From cjbujold at accra.ca Fri May 12 00:22:03 2017 From: cjbujold at accra.ca (Charles Bujold) Date: Thu, 11 May 2017 17:22:03 -0300 Subject: [Freeswitch-users] Freeswitch Packet Fragmentation issue Message-ID: <000e01d2ca94$40e47410$c2ad5c30$@accra.ca> The only codec is PCMU, nothing else listed. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170511/dc37423c/attachment.html From anthony.minessale at gmail.com Fri May 12 00:28:36 2017 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 11 May 2017 15:28:36 -0500 Subject: [Freeswitch-users] Freeswitch Packet Fragmentation issue In-Reply-To: <000e01d2ca94$40e47410$c2ad5c30$@accra.ca> References: <000e01d2ca94$40e47410$c2ad5c30$@accra.ca> Message-ID: Its not possible to tell from an image of wireshark parsing it. A basic INVITE with only PCMU would be well under 1500 bytes. On Thu, May 11, 2017 at 3:22 PM, Charles Bujold wrote: > The only codec is PCMU, nothing else listed. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170511/9f99d10e/attachment.html From lwahlmeier at gmail.com Fri May 12 01:48:00 2017 From: lwahlmeier at gmail.com (Luke Wahlmeier) Date: Thu, 11 May 2017 15:48:00 -0600 Subject: [Freeswitch-users] WSS SSL errors "decryption failed or bad record mac" under load Message-ID: I keep semi-regularly running into issues using the wss transport when using dtls/strp/ice. This is on the latest 1.6.17~34~0fc0946 on Debian jessie, but I am pretty sure it was happening on the last couple releases as well. It seems like something bad/wrong happens to the encrypted data going over the websocket coming from freeswitch when more then 1 websocket connection are going and so far ice/srtp/dtls also seem to be needed in the invite to duplicate it. I have tried many different languages and network/ssl stacks and keep running into this. It is always on data coming in from freeswitch on the websocket connection, and its very very random. Sometimes I will get it 20 times in a row, other times it takes thousands of connections/sessions before it happen. It also, obviously, completely goes away if I use plain ws instead wss. Here are the errors: python: SSLError: [SSL: DECRYPTION_FAILED_OR_BAD_RECORD_MAC] decryption failed or bad record mac (_ssl.c:1750) c/c++ (stunnel4): SSL_read: 1408F119: error:1408F119:SSL routines:SSL3_GET_RECORD:decryption failed or bad record mac Java: java.lang.IllegalArgumentException: Bad arguments at javax.crypto.Mac.update(Mac.java:509) at sun.security.ssl.MAC.compute(MAC.java:135) at sun.security.ssl.InputRecord.checkMacTags(InputRecord.java:265) at sun.security.ssl.InputRecord.decrypt(InputRecord.java:216) at sun.security.ssl.EngineInputRecord.decrypt(EngineInputRecord.java:177) at sun.security.ssl.SSLEngineImpl.readRecord(SSLEngineImpl.java:974) at sun.security.ssl.SSLEngineImpl.readNetRecord(SSLEngineImpl.java:907) at sun.security.ssl.SSLEngineImpl.unwrap(SSLEngineImpl.java:781) at javax.net.ssl.SSLEngine.unwrap(SSLEngine.java:624) Attached are a simple python script to do the load, my dialplan and sip_profile. The python script can take a few runs before it see the error, and I know its not completing the sip or rtp, but even if it does this still happens. I have also looked at libsofia-sip-ua/tport/ws.c and I dont see anything obvious. I am getting setup to build v1.6 head and test this any guidance on ways I can trouble shoot this better or requests for more info are very welcome. Thanks Luke -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170511/df7de8f5/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: wss-dialplan.xml Type: text/xml Size: 258 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170511/df7de8f5/attachment.xml -------------- next part -------------- A non-text attachment was scrubbed... Name: wss-profile.xml Type: text/xml Size: 3570 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170511/df7de8f5/attachment-0001.xml -------------- next part -------------- A non-text attachment was scrubbed... Name: load.py Type: text/x-python Size: 2580 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170511/df7de8f5/attachment.py From karl at xtronics.com Fri May 12 01:54:17 2017 From: karl at xtronics.com (Karl Schmidt) Date: Thu, 11 May 2017 16:54:17 -0500 Subject: [Freeswitch-users] Debian Stretch In-Reply-To: <000201d2ca8e$a53caeb0$efb60c10$@accra.ca> References: <000201d2ca8e$a53caeb0$efb60c10$@accra.ca> Message-ID: There is this error: W: GPG error: http://files.freeswitch.org/repo/deb/debian jessie InRelease: The following signatures couldn't be verified because the public key is not available: NO_PUBKEY D76EDC7725E010CF And adding the key from http://files.freeswitch.org/repo/deb/debian/freeswitch_archive_g0.pub isn't helpful in Stretch - needs a new type of key.. exploits etc. ,.,.,. Also stretch is almost out so it it time to add it to the repository. 135 - bugs left.. https://bugs.debian.org/release-critical/ And on that note - you probably want testing - or a testing symlink to sid if it is one and the same thing. -------------------------------------------------------------------------------- Karl Schmidt EMail Karl at xtronics.com 3209 West 9th Street Ph (785) 841-3089 Lawrence, KS 66049 Nothing defines humans better than their willingness to do irrational things in the pursuit of phenomenally unlikely payoffs. This is the principle behind lotteries, dating, and religion. --Scott Adams -------------------------------------------------------------------------------- From brian at freeswitch.org Fri May 12 01:59:36 2017 From: brian at freeswitch.org (Brian West) Date: Thu, 11 May 2017 16:59:36 -0500 Subject: [Freeswitch-users] Debian Stretch In-Reply-To: References: <000201d2ca8e$a53caeb0$efb60c10$@accra.ca> Message-ID: Also we recommend downloading the key via https, I think I went thru and updated all those instructions. Maybe you can file a JIRA and outline any change we need to make and help solve this. Thanks, /b On Thu, May 11, 2017 at 4:54 PM, Karl Schmidt wrote: > > There is this error: > > W: GPG error: http://files.freeswitch.org/repo/deb/debian jessie > InRelease: The following signatures > couldn't be verified because the public key is not available: NO_PUBKEY > D76EDC7725E010CF > > And adding the key from > http://files.freeswitch.org/repo/deb/debian/freeswitch_archive_g0.pub > > isn't helpful in Stretch - needs a new type of key.. exploits etc. > > ,.,.,. > > Also stretch is almost out so it it time to add it to the repository. > > 135 - bugs left.. > https://bugs.debian.org/release-critical/ > > And on that note - you probably want testing - or a testing symlink to sid > if it is one and the same > thing. > > > > > > > > > > > > ------------------------------------------------------------ > -------------------- > Karl Schmidt EMail Karl at xtronics.com > 3209 West 9th Street Ph (785) 841-3089 > Lawrence, KS 66049 > > Nothing defines humans better than their > willingness to do irrational things in the pursuit of > phenomenally unlikely payoffs. This is the > principle behind lotteries, dating, and religion. > --Scott Adams > > ------------------------------------------------------------ > -------------------- > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Book a phone call (CST) Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170511/3727f90b/attachment.html From mike at jerris.com Fri May 12 02:20:42 2017 From: mike at jerris.com (Michael Jerris) Date: Thu, 11 May 2017 18:20:42 -0400 Subject: [Freeswitch-users] WSS SSL errors "decryption failed or bad record mac" under load In-Reply-To: References: Message-ID: if you can reproduce this reliably, i?d try master as well. Unless this is a bug in openssl, i can?t imagine how dtls would come into play in something like this. > On May 11, 2017, at 5:48 PM, Luke Wahlmeier wrote: > > I keep semi-regularly running into issues using the wss transport when using dtls/strp/ice. This is on the latest 1.6.17~34~0fc0946 on Debian jessie, but I am pretty sure it was happening on the last couple releases as well. > > It seems like something bad/wrong happens to the encrypted data going over the websocket coming from freeswitch when more then 1 websocket connection are going and so far ice/srtp/dtls also seem to be needed in the invite to duplicate it. > > I have tried many different languages and network/ssl stacks and keep running into this. It is always on data coming in from freeswitch on the websocket connection, and its very very random. Sometimes I will get it 20 times in a row, other times it takes thousands of connections/sessions before it happen. It also, obviously, completely goes away if I use plain ws instead wss. > > Here are the errors: > python: > SSLError: [SSL: DECRYPTION_FAILED_OR_BAD_RECORD_MAC] decryption failed or bad record mac (_ssl.c:1750) > c/c++ (stunnel4): > SSL_read: 1408F119: error:1408F119:SSL routines:SSL3_GET_RECORD:decryption failed or bad record mac > Java: > java.lang.IllegalArgumentException: Bad arguments > at javax.crypto.Mac.update(Mac.java:509) > at sun.security.ssl.MAC.compute(MAC.java:135) > at sun.security.ssl.InputRecord.checkMacTags(InputRecord.java:265) > at sun.security.ssl.InputRecord.decrypt(InputRecord.java:216) > at sun.security.ssl.EngineInputRecord.decrypt(EngineInputRecord.java:177) > at sun.security.ssl.SSLEngineImpl.readRecord(SSLEngineImpl.java:974) > at sun.security.ssl.SSLEngineImpl.readNetRecord(SSLEngineImpl.java:907) > at sun.security.ssl.SSLEngineImpl.unwrap(SSLEngineImpl.java:781) > at javax.net.ssl.SSLEngine.unwrap(SSLEngine.java:624) > > Attached are a simple python script to do the load, my dialplan and sip_profile. The python script can take a few runs before it see the error, and I know its not completing the sip or rtp, but even if it does this still happens. > > I have also looked at libsofia-sip-ua/tport/ws.c and I dont see anything obvious. I am getting setup to build v1.6 head and test this any guidance on ways I can trouble shoot this better or requests for more info are very welcome. > From mike at jerris.com Fri May 12 02:22:12 2017 From: mike at jerris.com (Michael Jerris) Date: Thu, 11 May 2017 18:22:12 -0400 Subject: [Freeswitch-users] Debian Stretch In-Reply-To: References: <000201d2ca8e$a53caeb0$efb60c10$@accra.ca> Message-ID: <3A8D18B9-294D-45F6-9D89-7674E7D871C5@jerris.com> Current freeswitch releases will not work on stretch. The openssl for sure is not compatible. > On May 11, 2017, at 5:54 PM, Karl Schmidt wrote: > > > There is this error: > > W: GPG error: http://files.freeswitch.org/repo/deb/debian jessie InRelease: The following signatures > couldn't be verified because the public key is not available: NO_PUBKEY D76EDC7725E010CF > > And adding the key from > http://files.freeswitch.org/repo/deb/debian/freeswitch_archive_g0.pub > > isn't helpful in Stretch - needs a new type of key.. exploits etc. > > ,.,.,. > > Also stretch is almost out so it it time to add it to the repository. > > 135 - bugs left.. > https://bugs.debian.org/release-critical/ > > And on that note - you probably want testing - or a testing symlink to sid if it is one and the same > thing. From mike at jerris.com Fri May 12 02:26:02 2017 From: mike at jerris.com (Michael Jerris) Date: Thu, 11 May 2017 18:26:02 -0400 Subject: [Freeswitch-users] Changing audio DTLS state from HANDSHAKE to FAIL issue In-Reply-To: <1493999540.S.17847.21554.f4-235-140.1494245336.524@webmail.rediffmail.com> References: <1493999540.S.17847.21554.f4-235-140.1494245336.524@webmail.rediffmail.com> Message-ID: <35ADA22B-2ABE-425F-B7D8-A8B0D1B489DD@jerris.com> we just had confirmed on Jira that 1.6.17 windows is working fine with latest chrome. If you are getting errors still with something that is other than sipml5, please confirm details of what exactly the issue is. > On May 8, 2017, at 8:08 AM, sachin wrote: > > Hello Michael, > > I tried using Sip.js instead of sipml5 and with Linux as the server. I have installed the FreeSWITCH Version 1.9.0+git~20170501T171230Z~e3ef041517~64bit (git e3ef041 2017-05-01 17:12:30Z 64bit) > > I am getting INCOMPATIBLE DESTINATION error. Also tried sipml5.. with bot the client I am getting the same error. > ? > Please let me know what I am missing. > ? > Thanks and Regards, > ?SD > > > From: Michael Jerris > > Sent: Fri, 05 May 2017 21:22:20 > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] Changing audio DTLS state from HANDSHAKE to FAIL issue > > can you try this using sip.js or something thats confirmed to work when using linux as a server please. This may just be an issue with sipml5. > >> On May 5, 2017, at 9:53 AM, Brian West wrote: >> >> Clearly that isn't the case, I suspect a define is missing to actually enable ECDSA at this point. >> >> On Fri, May 5, 2017 at 8:47 AM, Gregor Nanger wrote: >> Brian, isn't this solved in 1.6.16? >> >> FS-10037 [core] Update OpenSSL to version 1.0.2k for Windows >> >> 2017-05-05 15:26 GMT+02:00 Brian West : >> OpenSSL on the windows build needs to be updated. >> >> https://freeswitch.org/jira/browse/FS-9510 >> >> On Thu, May 4, 2017 at 2:31 PM, Gregor Nanger wrote: >> This error is familiar to me, I think so, if I remembered correctly. >> >> I am using windows. I had this error when client was Chrome. I just updated openssl dlls. I think that openssl library is updated on windows build now with version 1.6.17. And this version is also compiled on FS FTP to just download and install it. >> >> 2017-05-04 19:15 GMT+02:00 Anthony Minessale : >> FS 1.5 sounds like a bad plan. >> Try latest FS 1.6 or master. >> >> >> On Thu, May 4, 2017 at 12:04 PM, ?talo Rossi wrote: >> Old openssl version maybe? >> Em qui, 4 de mai de 2017 ?s 11:19, sachin escreveu: >> Hello All, >> >> ?I am trying to setup the Webrtc call between 2 SIPML5 client. I am using FreeSWITCH Version 1.6.14 on Windows. I am using internal LAN. >> >> The clients are getting registered over wss. I have created self signed certificates. In var.xml I have set the codecs setting as follows >> >> >> >> >> ?I am able to establish the call and there is 2 way voice when I call from Sipmpl5 to Linphone and it works. But when I call from Linphone to Firefox (Simpl5) then the call is not getting established and I am getting following error >> >> 2017-05-04 11:29:04.543434 [ERR] switch_rtp.c:2907 audio Handshake failure 1 >> 2017-05-04 11:29:04.543434 [INFO] switch_rtp.c:2908 Changing audio DTLS state from HANDSHAKE to FAIL >> 2017-05-04 11:29:04.543434 [NOTICE] switch_rtp.c:2889 Hangup sofia/internal/sip:1001 at df7jal 23ls0d.invalid [CS_EXCHANGE_MEDIA] [DESTINATION_OUT_OF_ORDER] >> >> My setup is as follows >> >> SIPML5 (FireFox browser 53.0)<-----> FS (v1.5)(Windows) <----> Linphone >> >> I am attaching the logs for the reference. >> fs-logleve9.txt : Debug trace with loglevel =9 >> ?fs-sip-trace.txt : Sip trace >> Please let me know what could the issue and pointers to resolve the same. >> >> ?Thanks and Regards >> SD -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170511/8368a064/attachment-0001.html From lwahlmeier at gmail.com Fri May 12 02:31:21 2017 From: lwahlmeier at gmail.com (Luke Wahlmeier) Date: Thu, 11 May 2017 16:31:21 -0600 Subject: [Freeswitch-users] WSS SSL errors "decryption failed or bad record mac" under load In-Reply-To: References: Message-ID: Yeah I can usually get it to happen within about 5 minutes or so of testing. Still getting all setup to build freeswitch in this environment, but I should have it working by tomorrow. I will try more w/o dtls/srtp as well and make sure it does not need to be on. Thanks Luke On Thu, May 11, 2017 at 4:20 PM, Michael Jerris wrote: > if you can reproduce this reliably, i?d try master as well. Unless this > is a bug in openssl, i can?t imagine how dtls would come into play in > something like this. > > > On May 11, 2017, at 5:48 PM, Luke Wahlmeier > wrote: > > > > I keep semi-regularly running into issues using the wss transport when > using dtls/strp/ice. This is on the latest 1.6.17~34~0fc0946 on Debian > jessie, but I am pretty sure it was happening on the last couple releases > as well. > > > > It seems like something bad/wrong happens to the encrypted data going > over the websocket coming from freeswitch when more then 1 websocket > connection are going and so far ice/srtp/dtls also seem to be needed in the > invite to duplicate it. > > > > I have tried many different languages and network/ssl stacks and keep > running into this. It is always on data coming in from freeswitch on the > websocket connection, and its very very random. Sometimes I will get it 20 > times in a row, other times it takes thousands of connections/sessions > before it happen. It also, obviously, completely goes away if I use plain > ws instead wss. > > > > Here are the errors: > > python: > > SSLError: [SSL: DECRYPTION_FAILED_OR_BAD_RECORD_MAC] decryption failed > or bad record mac (_ssl.c:1750) > > c/c++ (stunnel4): > > SSL_read: 1408F119: error:1408F119:SSL routines:SSL3_GET_RECORD:decryption > failed or bad record mac > > Java: > > java.lang.IllegalArgumentException: Bad arguments > > at javax.crypto.Mac.update(Mac.java:509) > > at sun.security.ssl.MAC.compute(MAC.java:135) > > at sun.security.ssl.InputRecord.checkMacTags(InputRecord.java:265) > > at sun.security.ssl.InputRecord.decrypt(InputRecord.java:216) > > at sun.security.ssl.EngineInputRecord.decrypt( > EngineInputRecord.java:177) > > at sun.security.ssl.SSLEngineImpl.readRecord(SSLEngineImpl.java:974) > > at sun.security.ssl.SSLEngineImpl.readNetRecord( > SSLEngineImpl.java:907) > > at sun.security.ssl.SSLEngineImpl.unwrap(SSLEngineImpl.java:781) > > at javax.net.ssl.SSLEngine.unwrap(SSLEngine.java:624) > > > > Attached are a simple python script to do the load, my dialplan and > sip_profile. The python script can take a few runs before it see the > error, and I know its not completing the sip or rtp, but even if it does > this still happens. > > > > I have also looked at libsofia-sip-ua/tport/ws.c and I dont see anything > obvious. I am getting setup to build v1.6 head and test this any guidance > on ways I can trouble shoot this better or requests for more info are very > welcome. > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170511/426de2ca/attachment.html From mike at jerris.com Fri May 12 02:34:08 2017 From: mike at jerris.com (Michael Jerris) Date: Thu, 11 May 2017 18:34:08 -0400 Subject: [Freeswitch-users] WSS SSL errors "decryption failed or bad record mac" under load In-Reply-To: References: Message-ID: what is ?this environment? ? > On May 11, 2017, at 6:31 PM, Luke Wahlmeier wrote: > > Yeah I can usually get it to happen within about 5 minutes or so of testing. Still getting all setup to build freeswitch in this environment, but I should have it working by tomorrow. I will try more w/o dtls/srtp as well and make sure it does not need to be on. > > Thanks > Luke > > On Thu, May 11, 2017 at 4:20 PM, Michael Jerris > wrote: > if you can reproduce this reliably, i?d try master as well. Unless this is a bug in openssl, i can?t imagine how dtls would come into play in something like this. > > > On May 11, 2017, at 5:48 PM, Luke Wahlmeier > wrote: > > > > I keep semi-regularly running into issues using the wss transport when using dtls/strp/ice. This is on the latest 1.6.17~34~0fc0946 on Debian jessie, but I am pretty sure it was happening on the last couple releases as well. > > > > It seems like something bad/wrong happens to the encrypted data going over the websocket coming from freeswitch when more then 1 websocket connection are going and so far ice/srtp/dtls also seem to be needed in the invite to duplicate it. > > > > I have tried many different languages and network/ssl stacks and keep running into this. It is always on data coming in from freeswitch on the websocket connection, and its very very random. Sometimes I will get it 20 times in a row, other times it takes thousands of connections/sessions before it happen. It also, obviously, completely goes away if I use plain ws instead wss. > > > > Here are the errors: > > python: > > SSLError: [SSL: DECRYPTION_FAILED_OR_BAD_RECORD_MAC] decryption failed or bad record mac (_ssl.c:1750) > > c/c++ (stunnel4): > > SSL_read: 1408F119: error:1408F119:SSL routines:SSL3_GET_RECORD:decryption failed or bad record mac > > Java: > > java.lang.IllegalArgumentException: Bad arguments > > at javax.crypto.Mac.update(Mac.java:509) > > at sun.security.ssl.MAC.compute(MAC.java:135) > > at sun.security.ssl.InputRecord.checkMacTags(InputRecord.java:265) > > at sun.security.ssl.InputRecord.decrypt(InputRecord.java:216) > > at sun.security.ssl.EngineInputRecord.decrypt(EngineInputRecord.java:177) > > at sun.security.ssl.SSLEngineImpl.readRecord(SSLEngineImpl.java:974) > > at sun.security.ssl.SSLEngineImpl.readNetRecord(SSLEngineImpl.java:907) > > at sun.security.ssl.SSLEngineImpl.unwrap(SSLEngineImpl.java:781) > > at javax.net.ssl.SSLEngine.unwrap(SSLEngine.java:624) > > > > Attached are a simple python script to do the load, my dialplan and sip_profile. The python script can take a few runs before it see the error, and I know its not completing the sip or rtp, but even if it does this still happens. > > > > I have also looked at libsofia-sip-ua/tport/ws.c and I dont see anything obvious. I am getting setup to build v1.6 head and test this any guidance on ways I can trouble shoot this better or requests for more info are very welcome. > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170511/9d323d76/attachment.html From lwahlmeier at gmail.com Fri May 12 03:12:32 2017 From: lwahlmeier at gmail.com (Luke Wahlmeier) Date: Thu, 11 May 2017 17:12:32 -0600 Subject: [Freeswitch-users] WSS SSL errors "decryption failed or bad record mac" under load In-Reply-To: References: Message-ID: Its just in our isolated lab, pretty normal dell xeon server running Jessie 8.6. I just want to get it building on the same box I am testing with so setting that all up. I was able to reproduce it w/o DTLS/Srtp. here is a much simpler and cleaned up version of the python script. On Thu, May 11, 2017 at 4:34 PM, Michael Jerris wrote: > what is ?this environment? ? > > On May 11, 2017, at 6:31 PM, Luke Wahlmeier wrote: > > Yeah I can usually get it to happen within about 5 minutes or so of > testing. Still getting all setup to build freeswitch in this environment, > but I should have it working by tomorrow. I will try more w/o dtls/srtp as > well and make sure it does not need to be on. > > Thanks > Luke > > On Thu, May 11, 2017 at 4:20 PM, Michael Jerris wrote: > >> if you can reproduce this reliably, i?d try master as well. Unless this >> is a bug in openssl, i can?t imagine how dtls would come into play in >> something like this. >> >> > On May 11, 2017, at 5:48 PM, Luke Wahlmeier >> wrote: >> > >> > I keep semi-regularly running into issues using the wss transport when >> using dtls/strp/ice. This is on the latest 1.6.17~34~0fc0946 on Debian >> jessie, but I am pretty sure it was happening on the last couple releases >> as well. >> > >> > It seems like something bad/wrong happens to the encrypted data going >> over the websocket coming from freeswitch when more then 1 websocket >> connection are going and so far ice/srtp/dtls also seem to be needed in the >> invite to duplicate it. >> > >> > I have tried many different languages and network/ssl stacks and keep >> running into this. It is always on data coming in from freeswitch on the >> websocket connection, and its very very random. Sometimes I will get it 20 >> times in a row, other times it takes thousands of connections/sessions >> before it happen. It also, obviously, completely goes away if I use plain >> ws instead wss. >> > >> > Here are the errors: >> > python: >> > SSLError: [SSL: DECRYPTION_FAILED_OR_BAD_RECORD_MAC] decryption failed >> or bad record mac (_ssl.c:1750) >> > c/c++ (stunnel4): >> > SSL_read: 1408F119: error:1408F119:SSL routines:SSL3_GET_RECORD:decryption >> failed or bad record mac >> > Java: >> > java.lang.IllegalArgumentException: Bad arguments >> > at javax.crypto.Mac.update(Mac.java:509) >> > at sun.security.ssl.MAC.compute(MAC.java:135) >> > at sun.security.ssl.InputRecord.checkMacTags(InputRecord.java:265) >> > at sun.security.ssl.InputRecord.decrypt(InputRecord.java:216) >> > at sun.security.ssl.EngineInputRecord.decrypt(EngineInputRecord >> .java:177) >> > at sun.security.ssl.SSLEngineImpl.readRecord(SSLEngineImpl. >> java:974) >> > at sun.security.ssl.SSLEngineImpl.readNetRecord(SSLEngineImpl. >> java:907) >> > at sun.security.ssl.SSLEngineImpl.unwrap(SSLEngineImpl.java:781) >> > at javax.net.ssl.SSLEngine.unwrap(SSLEngine.java:624) >> > >> > Attached are a simple python script to do the load, my dialplan and >> sip_profile. The python script can take a few runs before it see the >> error, and I know its not completing the sip or rtp, but even if it does >> this still happens. >> > >> > I have also looked at libsofia-sip-ua/tport/ws.c and I dont see >> anything obvious. I am getting setup to build v1.6 head and test this any >> guidance on ways I can trouble shoot this better or requests for more info >> are very welcome. >> > >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170511/89d60ee8/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: load.py Type: text/x-python Size: 2060 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170511/89d60ee8/attachment-0001.py From hunterj91 at hotmail.com Fri May 12 13:25:56 2017 From: hunterj91 at hotmail.com (Jonathan Hunter) Date: Fri, 12 May 2017 09:25:56 +0000 Subject: [Freeswitch-users] voicemail_ivr examples when checking voicemail In-Reply-To: References: <719512DD-9FEF-45BB-9E25-058B33BD278D@jerris.com>, , Message-ID: Hi Michael, Thanks for the response, so I can just exclude phrases in; etc/freeswitch/lang/en/vm/sounds.xml to stop it reading out particular options, as essentially I dont want the user ringing in to check their voicemail to not have options to record multiple greetings, just want them to be able to record one, thats only change I need ? Jon ---------- Forwarded message ---------- From: Michael Jerris > Date: Wed, May 10, 2017 at 7:49 PM Subject: Re: [Freeswitch-users] voicemail_ivr examples when checking voicemail To: FreeSWITCH Users Help > On May 10, 2017, at 2:34 PM, Jonathan Hunter > wrote: Hey guys, Sorry for the noise! I am looking to use voicemail_ivr so I can ensure my own phrases are played and I can avoid giving the user options to record just one voicemail message (as they have the option to upload a file from our portal too). You can do all this with regular voicemail app by modifying the phrases involved and options I think. No need to do your own completely. Take a look at the phrase configs used for voicemail and voicemail config file and let us know where it might be an issue. Can this be achieved using voicemail_ivr and if so are there any examples as confluence is good but doesnt have many configuration examples if you want to miss out phrases etc. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Jonathan Hunter Technical Director /Telephony Developer M:(+44) 7917 190 438 Email:jhunter at voxboxcoms.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170512/4cc9f4c5/attachment.html From david.villasmil.work at gmail.com Fri May 12 14:21:58 2017 From: david.villasmil.work at gmail.com (David Villasmil) Date: Fri, 12 May 2017 12:21:58 +0200 Subject: [Freeswitch-users] STRP Message-ID: Hello Guys, So i've got this scenario where i'm receving in TCP+SRTP(opus) and I need to send to the b-leg in TCP+STRP(PCMA)... i've enabled: But the sdp is being went without srtp, what am I missing? Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170512/d0965834/attachment.html From colin.morelli at gmail.com Fri May 12 14:41:46 2017 From: colin.morelli at gmail.com (Colin Morelli) Date: Fri, 12 May 2017 10:41:46 +0000 Subject: [Freeswitch-users] STRP In-Reply-To: References: Message-ID: All RTP related channel vars were renamed from sip_ to rtp_. Try rtp_secure_media. Best, Colin On Fri, May 12, 2017 at 6:24 AM David Villasmil < david.villasmil.work at gmail.com> wrote: > Hello Guys, > > So i've got this scenario where i'm receving in TCP+SRTP(opus) and I need > to send to the b-leg in TCP+STRP(PCMA)... i've enabled: > > data="{sip_secure_media=true,sdp_secure_savp_only=true,absolute_codec_string=^^:PCMA:PCMU}sofia/external/${destination_number}@myserver;transport=tcp" > /> > > > But the sdp is being went without srtp, what am I missing? > > > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 > ? > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170512/2919d110/attachment.html From asilva at wirelessmundi.com Fri May 12 14:55:23 2017 From: asilva at wirelessmundi.com (Antonio Silva) Date: Fri, 12 May 2017 12:55:23 +0200 Subject: [Freeswitch-users] mod_distribution feature Message-ID: <3e03ed2f-54eb-48a7-c78d-9df74764e98b@wirelessmundi.com> Hi all, Just found mod_distribution and is pretty cool feature to have. Right now it distributes calls to gateways in a weighted round-robin fashion, could it be cool to have /Weighted Random Distribution. /i found this paper explaining an implementation in python: https://www.electricmonk.nl/log/2009/12/23/weighted-random-distribution/ it could be ported to mod_distributor. / /What you guys think about it? -- Saludos / Regards / Cumprimentos, Ant?nio silva -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170512/13290337/attachment-0001.html From david.villasmil.work at gmail.com Fri May 12 15:03:06 2017 From: david.villasmil.work at gmail.com (David Villasmil) Date: Fri, 12 May 2017 13:03:06 +0200 Subject: [Freeswitch-users] STRP In-Reply-To: References: Message-ID: Hello, yeah I had completely forgotten that! At least now it's offering SAVP, but for some reason the negotiation fails... Can you give me a hand? I can't find where the issue might lie... Here's the log... 2017-05-12 10:52:25.432321 [DEBUG] switch_core_state_machine.c:584 (sofia/external/%2B16151920015 at 1.2.3.153[gw]:5111) Running State Change CS_CONSUME_MEDIA (Cur 2 Tot 76) send 2024 bytes to tcp/[1.2.3.153[gw]]:5111 at 10:52:25.441911: ------------------------------------------------------------------------ INVITE sip:+16151920015 at 1.2.3.153[gw]:5111;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 1.2.3.43[fs]:5111;branch=z9hG4bKyK3rSg8XK3X1B Max-Forwards: 68 From: "soluto" ;tag=Ua2NXSa4mF7rD To: Call-ID: ecef602a-b1a3-1235-749f-0cc47a205c6e CSeq: 106957108 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.6.17~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 1073 p-time: 396889 p-ident: afac52cc2fd51e1110a06d4f9d44c80d P-Application: SOLUTO X-Src: 0;8.8.0.24[client-ip];1.2.3.25[registrar-kamailio] X-FS-Support: update_display,send_info Remote-Party-ID: "soluto" ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1494558991 1494558992 IN IP4 1.2.3.43[fs] s=FreeSWITCH c=IN IP4 1.2.3.43[fs] t=0 0 m=audio 27354 RTP/SAVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=crypto:1 AEAD_AES_256_GCM_8 inline:2/QJPmYA6js+Vo2N5Z/iowwd4IKAwXOgVdmy3trDM56ys2vm/1pViViAtiE a=crypto:2 AEAD_AES_128_GCM_8 inline:O8Ib9toXNXjxAVAH5uI/xcZRvpgz4bLbtFPsQA a=crypto:3 AES_CM_256_HMAC_SHA1_80 inline:QMeDgbL1fXf1Q4pAUpG6EtQa95cceubGFgM7WIfiR6YfUe+aOHSitl+ZuPAdoQ a=crypto:4 AES_CM_192_HMAC_SHA1_80 inline:8JFbH6JihbqIriChDZba/dGKiViieJAQW6SKZ9QvX7HKXedRpzo a=crypto:5 AES_CM_128_HMAC_SHA1_80 inline:0SNbzrAEs4rSTY+K0oeEqL26/K5UvRv9/8Yw1LFP a=crypto:6 AES_CM_256_HMAC_SHA1_32 inline:z0DO9vlMLqHU2ATq9FJDGD6V8DVIYubt2KhCbrZ84zKiPmnSzxt2P8MtU3OHbA a=crypto:7 AES_CM_192_HMAC_SHA1_32 inline:IHXpuYGvEuJE/0coHGAIhMMuQnzJb/J7kTJBT54uVoyVlByAeqY a=crypto:8 AES_CM_128_HMAC_SHA1_32 inline:8um9NRNdmXJcOU8ymod3ohf0tnrh1rBjIEigQOLW a=crypto:9 AES_CM_128_NULL_AUTH inline:1cDItzdQZ0Z8whBwb0w7z2ZbY52Kzy4EroNQwJWd a=ptime:20 ------------------------------------------------------------------------ tport.c:2296 tport_set_secondary_timer() tport(0x7fa14000a110): reset timer nta.c:8304 outgoing_send() nta: sent INVITE (106957108) to tcp/1.2.3.153 [gw]:5111 tport.c:4160 tport_pend() tport_pend(0x7fa14000a110): pending 0x7fa140019990 for tcp/1.2.3.153[gw]:5111 (already 0) nua_session.c:4139 signal_call_state_change() nua(0x7fa100006de0): call state changed: init -> calling, sent offer soa.c:1270 soa_get_local_sdp() soa_get_local_sdp(static::0x7fa14000f210, [0x7fa16690eab0], [0x7fa16690eab8], [(nil)]) called nua_stack.c:269 nua_stack_event() nua(0x7fa100006de0): event i_state INVITE sent 2017-05-12 10:52:25.432321 [DEBUG] switch_core_state_machine.c:662 (sofia/external/%2B16151920015 at 1.2.3.153[gw]:5111) State CONSUME_MEDIA 2017-05-12 10:52:25.432321 [DEBUG] switch_core_state_machine.c:662 (sofia/external/%2B16151920015 at 1.2.3.153[gw]:5111) State CONSUME_MEDIA going to sleep nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 2017-05-12 10:52:25.432321 [DEBUG] sofia.c:7048 Channel sofia/external/% 2B16151920015 at 1.2.3.153[gw]:5111 entering state [calling][0] nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering tport.c:2773 tport_wakeup() tport_wakeup(0x7fa14000a110): events IN tport.c:2864 tport_recv_event() tport_recv_event(0x7fa14000a110) tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fa14000a110) msg 0x7fa140016780 from (tcp/1.2.3.153[gw]:5111) has 369 bytes, veclen = 1 recv 369 bytes from tcp/[1.2.3.153[gw]]:5111 at 10:52:25.447037: ------------------------------------------------------------------------ SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/TCP 1.2.3.43[fs]:5111;branch=z9hG4bKyK3rSg8XK3X1B From: "soluto" ;tag=Ua2NXSa4mF7rD To: Call-ID: ecef602a-b1a3-1235-749f-0cc47a205c6e CSeq: 106957108 INVITE Server: kamailio (4.2.8 (x86_64/linux)) Content-Length: 0 ------------------------------------------------------------------------ tport.c:3023 tport_deliver() tport_deliver(0x7fa14000a110): msg 0x7fa140016780 (369 bytes) from tcp/1.2.3.153[gw]:5111/sip next=(nil) nta.c:3299 agent_recv_response() nta: received 100 trying -- your call is important to us for INVITE (106957108) nta.c:3366 agent_recv_response() nta: 100 trying -- your call is important to us is going to a transaction nta.c:9564 outgoing_estimate_delay() nta_outgoing: RTT is 5.235 ms tport.c:4222 tport_release() tport_release(0x7fa14000a110): 0x7fa140019990 by 0x7fa1400132d0 with 0x7fa140016780 (preliminary) tport.c:2296 tport_set_secondary_timer() tport(0x7fa14000a110): reset timer tport.c:2773 tport_wakeup() tport_wakeup(0x7fa14000a110): events IN tport.c:2864 tport_recv_event() tport_recv_event(0x7fa14000a110) tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fa14000a110) msg 0x7fa140016780 from (tcp/1.2.3.153[gw]:5111) has 1125 bytes, veclen = 1 recv 1125 bytes from tcp/[1.2.3.153[gw]]:5111 at 10:52:25.597905: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/TCP 1.2.3.43[fs]:5111;branch=z9hG4bKyK3rSg8XK3X1B From: "soluto" ;tag=Ua2NXSa4mF7rD To: ;tag=SDbqa6299-801c7bd26943e712b98591536c00 Call-ID: ecef602a-b1a3-1235-749f-0cc47a205c6e CSeq: 106957108 INVITE Record-Route: Record-Route: P-Asserted-Identity: P-AV-Message-Id: 1_2 Server: Avaya CM/R016x.03.0.124.0 AVAYA-SM-7.0.0.0.700007 Contact: Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, NOTIFY, REFER, INFO, UPDATE Supported: histinfo, join, replaces, timer Content-Type: application/sdp Content-Length: 282 v=0 o=- 2716084251 2 IN IP4 1.2.3.165[gw-rtp-ip] s=- c=IN IP4 1.2.3.165[gw-rtp-ip] b=AS:64 t=0 0 m=audio 14974 RTP/SAVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv a=ptime:20 a=crypto:5 AES_CM_128_HMAC_SHA1_80 inline:NE/0Ttf1pbNEUyUUxxILCBQGC/taRBRyBGFbsggE ------------------------------------------------------------------------ tport.c:3023 tport_deliver() tport_deliver(0x7fa14000a110): msg 0x7fa140016780 (1125 bytes) from tcp/1.2.3.153[gw]:5111/sip next=(nil) nta.c:3299 agent_recv_response() nta: received 180 Ringing for INVITE (106957108) nta.c:3366 agent_recv_response() nta: 180 Ringing is going to a transaction tport.c:4222 tport_release() tport_release(0x7fa14000a110): 0x7fa140019990 by 0x7fa1400132d0 with 0x7fa140016780 (preliminary) soa.c:1171 soa_set_remote_sdp() soa_set_remote_sdp(static::0x7fa14000f210, (nil), 0x7fa14001638b, 282) called soa.c:1595 soa_process_answer() soa_process_answer(static::0x7fa14000f210) called soa_static.c:1148 offer_answer_step() soa_static_offer_answer_action(0x7fa14000f210, soa_process_answer): called soa_static.c:1029 soa_sdp_mode_set() soa_sdp_mode_set(0x7fa140010c50, 0x7fa14001d5c0, ""): called soa_static.c:1304 offer_answer_step() soa_static(0x7fa14000f210, soa_process_answer): upgrade codecs with remote description soa_static.c:1446 offer_answer_step() soa_static(0x7fa14000f210, soa_process_answer): storing local description soa.c:1730 soa_activate() soa_activate(static::0x7fa14000f210, (nil)) called nua_session.c:988 nua_session_client_response() nua(0x7fa100006de0): INVITE: processed SDP answer in 180 Ringing nua_stack.c:271 nua_stack_event() nua(0x7fa100006de0): event r_invite 180 Ringing nua_session.c:4139 signal_call_state_change() nua(0x7fa100006de0): call state changed: calling -> proceeding, received answer soa.c:1098 soa_get_remote_sdp() soa_get_remote_sdp(static::0x7fa14000f210, [0x7fa16690e310], [0x7fa16690e318], [(nil)]) called soa.c:616 soa_get_params() soa_get_params(static::0x7fa14000f210, ...) called nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua_stack.c:271 nua_stack_event() nua(0x7fa100006de0): event i_state 180 Ringing tport.c:2296 tport_set_secondary_timer() tport(0x7fa14000a110): reset timer nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 2017-05-12 10:52:25.592332 [INFO] sofia.c:1279 sofia/external/% 2B16151920015 at 1.2.3.153[gw]:5111 Update Callee ID to "Outbound Call" <6151920015> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 2017-05-12 10:52:25.592332 [DEBUG] sofia.c:7048 Channel sofia/external/% 2B16151920015 at 1.2.3.153[gw]:5111 entering state [proceeding][180] 2017-05-12 10:52:25.592332 [DEBUG] sofia.c:7058 Remote SDP: v=0 o=- 2716084251 2 IN IP4 1.2.3.165[gw-rtp-ip] s=- c=IN IP4 1.2.3.165[gw-rtp-ip] b=AS:64 t=0 0 m=audio 14974 RTP/SAVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=ptime:20 a=crypto:5 AES_CM_128_HMAC_SHA1_80 inline:NE/0Ttf1pbNEUyUUxxILCBQGC/taRBRyBGFbsggE 2017-05-12 10:52:25.592332 [DEBUG] switch_core_media.c:1391 looking for crypto suite [AEAD_AES_256_GCM_8] in [5 AES_CM_128_HMAC_SHA1_80 inline:NE/0Ttf1pbNEUyUUxxILCBQGC/taRBRyBGFbsggE] 2017-05-12 10:52:25.592332 [DEBUG] switch_core_media.c:1391 looking for crypto suite [AEAD_AES_128_GCM_8] in [5 AES_CM_128_HMAC_SHA1_80 inline:NE/0Ttf1pbNEUyUUxxILCBQGC/taRBRyBGFbsggE] 2017-05-12 10:52:25.592332 [DEBUG] switch_core_media.c:1391 looking for crypto suite [AES_CM_256_HMAC_SHA1_80] in [5 AES_CM_128_HMAC_SHA1_80 inline:NE/0Ttf1pbNEUyUUxxILCBQGC/taRBRyBGFbsggE] 2017-05-12 10:52:25.592332 [DEBUG] switch_core_media.c:1391 looking for crypto suite [AES_CM_192_HMAC_SHA1_80] in [5 AES_CM_128_HMAC_SHA1_80 inline:NE/0Ttf1pbNEUyUUxxILCBQGC/taRBRyBGFbsggE] 2017-05-12 10:52:25.592332 [DEBUG] switch_core_media.c:1391 looking for crypto suite [AES_CM_128_HMAC_SHA1_80] in [5 AES_CM_128_HMAC_SHA1_80 inline:NE/0Ttf1pbNEUyUUxxILCBQGC/taRBRyBGFbsggE] 2017-05-12 10:52:25.592332 [DEBUG] switch_core_media.c:1396 Found suite AES_CM_128_HMAC_SHA1_80 2017-05-12 10:52:25.592332 [DEBUG] switch_core_media.c:1462 Set Remote Key [5 AES_CM_128_HMAC_SHA1_80 inline:NE/0Ttf1pbNEUyUUxxILCBQGC/taRBRyBGFbsggE] 2017-05-12 10:52:25.592332 [DEBUG] switch_core_media.c:4436 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2017-05-12 10:52:25.592332 [DEBUG] switch_core_media.c:4436 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2017-05-12 10:52:25.592332 [DEBUG] switch_core_media.c:4491 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match 2017-05-12 10:52:25.592332 [DEBUG] switch_core_media.c:4352 Set telephone-event payload to 101 at 8000 2017-05-12 10:52:25.592332 [DEBUG] switch_core_media.c:3056 Set Codec sofia/external/%2B16151920015 at 1.2.3.153[gw]:5111 PCMU/8000 20 ms 160 samples 64000 bits 1 channels 2017-05-12 10:52:25.592332 [DEBUG] switch_core_codec.c:111 sofia/external/% 2B16151920015 at 1.2.3.153[gw]:5111 Original read codec set to PCMU:0 2017-05-12 10:52:25.592332 [DEBUG] switch_core_media.c:4695 Set telephone-event payload to 101 at 8000 2017-05-12 10:52:25.592332 [DEBUG] switch_core_media.c:4754 sofia/external/% 2B16151920015 at 1.2.3.153[gw]:5111 Set 2833 dtmf send payload to 101 recv payload to 101 2017-05-12 10:52:25.592332 [DEBUG] switch_core_media.c:6865 AUDIO RTP [sofia/external/%2B16151920015 at 1.2.3.153[gw]:5111] 1.2.3.43[fs] port 27354 -> 1.2.3.165[gw-rtp-ip] port 14974 codec: 0 ms: 20 2017-05-12 10:52:25.592332 [DEBUG] switch_rtp.c:4096 Starting timer [soft] 160 bytes per 20ms 2017-05-12 10:52:25.592332 [DEBUG] switch_core_media.c:7166 sofia/external/% 2B16151920015 at 1.2.3.153[gw]:5111 Set 2833 dtmf send payload to 101 2017-05-12 10:52:25.592332 [DEBUG] switch_core_media.c:7173 sofia/external/% 2B16151920015 at 1.2.3.153[gw]:5111 Set 2833 dtmf receive payload to 101 2017-05-12 10:52:25.592332 [DEBUG] switch_core_media.c:7196 sofia/external/% 2B16151920015 at 1.2.3.153[gw]:5111 Set rtp dtmf delay to 40 2017-05-12 10:52:25.592332 [INFO] switch_rtp.c:3908 Activating audio Secure RTP SEND 2017-05-12 10:52:25.592332 [INFO] switch_rtp.c:3886 Activating audio Secure RTP RECV 2017-05-12 10:52:25.592332 [DEBUG] switch_core_sqldb.c:2617 Secure Type: srtp:sdes:AES_CM_128_HMAC_SHA1_80 2017-05-12 10:52:25.592332 [DEBUG] switch_core_sqldb.c:2617 Secure Type: srtp:sdes:AES_CM_128_HMAC_SHA1_80 2017-05-12 10:52:25.592332 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/%2B16151920015 at 1.2.3.153[gw]:5111! 2017-05-12 10:52:25.592332 [DEBUG] switch_channel.c:3473 (sofia/external/% 2B16151920015 at 1.2.3.153[gw]:5111) Callstate Change DOWN -> EARLY nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 2017-05-12 10:52:25.592332 [INFO] switch_ivr_originate.c:3639 Sending early media 2017-05-12 10:52:25.592332 [DEBUG] switch_core_media.c:4436 Audio Codec Compare [opus:107:48000:20:0:2]/[G722:9:8000:20:64000:1] 2017-05-12 10:52:25.592332 [DEBUG] switch_core_media.c:4436 Audio Codec Compare [opus:107:48000:20:0:2]/[PCMU:0:8000:20:64000:1] 2017-05-12 10:52:25.592332 [DEBUG] switch_core_media.c:4436 Audio Codec Compare [opus:107:48000:20:0:2]/[PCMA:8:8000:20:64000:1] 2017-05-12 10:52:25.592332 [DEBUG] switch_core_media.c:4352 Set telephone-event payload to 96 at 8000 2017-05-12 10:52:25.592332 [DEBUG] switch_core_media.c:4754 sofia/external/ soluto at 1.2.3.25[registrar-kamailio] Set 2833 dtmf send payload to 96 recv payload to 96 2017-05-12 10:52:25.592332 [ERR] mod_sofia.c:2342 CODEC NEGOTIATION ERROR. SDP: v=0 o=- 3703575145 3703575145 IN IP4 192.168.1.171 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 33228 RTP/SAVP 107 96 c=IN IP4 67.205.173.126 b=TIAS:64000 a=rtpmap:107 opus/48000/2 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=rtcp:33229 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:4h3KXpeuR1qSWxR4+/XCLDtCO3UGqrO5gXK6HIGF a=setup:actpass a=fingerprint:sha-1 7B:5A:DF:A5:CC:88:FD:CA:88:C2:A8:D1:73:1F:3A:D8:29:9F:B3:D0 a=ice-ufrag:T0L06kG3 a=ice-pwd:uLNAX5ppVlJmXAdY7E4SZKsSKd a=candidate:eTePfBnNsoctce5m 1 UDP 2130706431 67.205.173.126 33228 typ host a=candidate:eTePfBnNsoctce5m 2 UDP 2130706430 67.205.173.126 33229 typ host 2017-05-12 10:52:25.592332 [NOTICE] switch_channel.c:3514 Hangup sofia/external/soluto at 1.2.3.25[registrar-kamailio] [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] 2017-05-12 10:52:25.592332 [DEBUG] switch_ivr_originate.c:3646 sofia/external/soluto at 1.2.3.25[registrar-kamailio] Media Establishment Failed. 2017-05-12 10:52:25.592332 [NOTICE] switch_ivr_originate.c:3648 Hangup sofia/external/%2B16151920015 at 1.2.3.153[gw]:5111 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] 2017-05-12 10:52:25.592332 [DEBUG] switch_ivr_originate.c:3833 Originate Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION] 2017-05-12 10:52:25.592332 [DEBUG] switch_core_state_machine.c:584 (sofia/external/%2B16151920015 at 1.2.3.153[gw]:5111) Running State Change CS_HANGUP (Cur 2 Tot 76) 2017-05-12 10:52:25.592332 [DEBUG] switch_core_state_machine.c:850 (sofia/external/%2B16151920015 at 1.2.3.153[gw]:5111) Callstate Change EARLY -> HANGUP 2017-05-12 10:52:25.592332 [DEBUG] switch_core_state_machine.c:852 (sofia/external/%2B16151920015 at 1.2.3.153[gw]:5111) State HANGUP 2017-05-12 10:52:25.592332 [INFO] mod_dptools.c:3418 Originate Failed. Cause: INCOMPATIBLE_DESTINATION 2017-05-12 10:52:25.592332 [DEBUG] mod_sofia.c:438 Channel sofia/external/% 2B16151920015 at 1.2.3.153[gw]:5111 hanging up, cause: INCOMPATIBLE_DESTINATION 2017-05-12 10:52:25.592332 [DEBUG] mod_sofia.c:502 Sending CANCEL to sofia/external/%2B16151920015 at 1.2.3.153[gw]:5111 nua.c:651 nua_cancel() nua: nua_cancel: entering nua_stack.c:529 nua_signal() nua(0x7fa100006de0): sent signal r_cancel 2017-05-12 10:52:25.592332 [DEBUG] switch_core_state_machine.c:60 sofia/external/%2B16151920015 at 1.2.3.153[gw]:5111 Standard HANGUP, cause: INCOMPATIBLE_DESTINATION 2017-05-12 10:52:25.592332 [DEBUG] switch_core_state_machine.c:852 (sofia/external/%2B16151920015 at 1.2.3.153[gw]:5111) State HANGUP going to sleep nua_stack.c:569 nua_stack_signal() nua(0x7fa100006de0): recv signal r_cancel 2017-05-12 10:52:25.592332 [DEBUG] switch_core_session.c:2814 sofia/external/soluto at 1.2.3.25[registrar-kamailio] skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering 2017-05-12 10:52:25.592332 [DEBUG] switch_core_state_machine.c:650 (sofia/external/soluto at 1.2.3.25[registrar-kamailio]) State EXECUTE going to sleep soa.c:403 soa_set_params() soa_set_params(static::0x7fa14000f210, ...) called 2017-05-12 10:52:25.592332 [DEBUG] switch_core_state_machine.c:584 (sofia/external/soluto at 1.2.3.25[registrar-kamailio]) Running State Change CS_HANGUP (Cur 2 Tot 76) 2017-05-12 10:52:25.592332 [DEBUG] switch_core_state_machine.c:619 (sofia/external/%2B16151920015 at 1.2.3.153[gw]:5111) State Change CS_HANGUP -> CS_REPORTING tport.c:4588 tport_by_name() tport(0x7fa140004b30): found 0x7fa14000a110 by name tcp/1.2.3.153[gw]:5111 2017-05-12 10:52:25.592332 [DEBUG] switch_core_state_machine.c:584 (sofia/external/%2B16151920015 at 1.2.3.153[gw]:5111) Running State Change CS_REPORTING (Cur 2 Tot 76) tport.c:3257 tport_tsend() tport_tsend(0x7fa14000a110) tpn = tcp/1.2.3.153 [gw]:5111 tport.c:3594 tport_vsend() tport_vsend(0x7fa14000a110): 417 bytes of 417 to tcp/1.2.3.153[gw]:5111 tport.c:3492 tport_send_msg() tport_vsend returned 417 send 417 bytes to tcp/[1.2.3.153[gw]]:5111 at 10:52:25.604158: ------------------------------------------------------------------------ CANCEL sip:+16151920015 at 1.2.3.153[gw]:5111;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 1.2.3.43[fs]:5111;branch=z9hG4bKyK3rSg8XK3X1B Max-Forwards: 68 From: "soluto" ;tag=Ua2NXSa4mF7rD To: 2017-05-12 10:52:25.592332 [DEBUG] switch_core_state_machine.c:938 (sofia/external/%2B16151920015 at 1.2.3.153[gw]:5111) State REPORTING Call-ID: ecef602a-b1a3-1235-749f-0cc47a205c6e CSeq: 106957108 CANCEL Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" Content-Length: 0 ? Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 On Fri, May 12, 2017 at 12:41 PM, Colin Morelli wrote: > All RTP related channel vars were renamed from sip_ to rtp_. Try > rtp_secure_media. > > Best, > Colin > > On Fri, May 12, 2017 at 6:24 AM David Villasmil < > david.villasmil.work at gmail.com> wrote: > >> Hello Guys, >> >> So i've got this scenario where i'm receving in TCP+SRTP(opus) and I need >> to send to the b-leg in TCP+STRP(PCMA)... i've enabled: >> >> >> >> >> But the sdp is being went without srtp, what am I missing? >> >> >> Regards, >> >> David Villasmil >> email: david.villasmil.work at gmail.com >> phone: +34669448337 <+34%20669%2044%2083%2037> >> ? >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170512/c2acc7cf/attachment-0001.html From david.villasmil.work at gmail.com Fri May 12 15:09:00 2017 From: david.villasmil.work at gmail.com (David Villasmil) Date: Fri, 12 May 2017 13:09:00 +0200 Subject: [Freeswitch-users] STRP In-Reply-To: References: Message-ID: Nevermind, Opus wasn't compiled... :( ? Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 On Fri, May 12, 2017 at 1:03 PM, David Villasmil < david.villasmil.work at gmail.com> wrote: > Hello, > > yeah I had completely forgotten that! > > At least now it's offering SAVP, but for some reason the negotiation > fails... Can you give me a hand? I can't find where the issue might lie... > > Here's the log... > > > 2017-05-12 10:52:25.432321 [DEBUG] switch_core_state_machine.c:584 > (sofia/external/%2B16151920015 at 1.2.3.153[gw]:5111) Running State Change > CS_CONSUME_MEDIA (Cur 2 Tot 76) > send 2024 bytes to tcp/[1.2.3.153[gw]]:5111 at 10:52:25.441911: > ----------------------------------------------------------- > ------------- > INVITE sip:+16151920015 at 1.2.3.153[gw]:5111;transport=tcp SIP/2.0 > Via: SIP/2.0/TCP 1.2.3.43[fs]:5111;branch=z9hG4bKyK3rSg8XK3X1B > Max-Forwards: 68 > From: "soluto" ;tag=Ua2NXSa4mF7rD > To: > Call-ID: ecef602a-b1a3-1235-749f-0cc47a205c6e > CSeq: 106957108 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.6.17~64bit > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, > REGISTER, REFER, NOTIFY > Supported: timer, path, replaces > Allow-Events: talk, hold, conference, refer > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 1073 > p-time: 396889 > p-ident: afac52cc2fd51e1110a06d4f9d44c80d > P-Application: SOLUTO > X-Src: 0;8.8.0.24[client-ip];1.2.3.25[registrar-kamailio] > X-FS-Support: update_display,send_info > Remote-Party-ID: "soluto" ; > party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 1494558991 1494558992 IN IP4 1.2.3.43[fs] > s=FreeSWITCH > c=IN IP4 1.2.3.43[fs] > t=0 0 > m=audio 27354 RTP/SAVP 8 0 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=crypto:1 AEAD_AES_256_GCM_8 inline:2/QJPmYA6js+Vo2N5Z/ > iowwd4IKAwXOgVdmy3trDM56ys2vm/1pViViAtiE > a=crypto:2 AEAD_AES_128_GCM_8 inline:O8Ib9toXNXjxAVAH5uI/ > xcZRvpgz4bLbtFPsQA > a=crypto:3 AES_CM_256_HMAC_SHA1_80 inline: > QMeDgbL1fXf1Q4pAUpG6EtQa95cceubGFgM7WIfiR6YfUe+aOHSitl+ZuPAdoQ > a=crypto:4 AES_CM_192_HMAC_SHA1_80 inline:8JFbH6JihbqIriChDZba/ > dGKiViieJAQW6SKZ9QvX7HKXedRpzo > a=crypto:5 AES_CM_128_HMAC_SHA1_80 inline:0SNbzrAEs4rSTY+ > K0oeEqL26/K5UvRv9/8Yw1LFP > a=crypto:6 AES_CM_256_HMAC_SHA1_32 inline: > z0DO9vlMLqHU2ATq9FJDGD6V8DVIYubt2KhCbrZ84zKiPmnSzxt2P8MtU3OHbA > a=crypto:7 AES_CM_192_HMAC_SHA1_32 inline:IHXpuYGvEuJE/ > 0coHGAIhMMuQnzJb/J7kTJBT54uVoyVlByAeqY > a=crypto:8 AES_CM_128_HMAC_SHA1_32 inline: > 8um9NRNdmXJcOU8ymod3ohf0tnrh1rBjIEigQOLW > a=crypto:9 AES_CM_128_NULL_AUTH inline:1cDItzdQZ0Z8whBwb0w7z2ZbY52Kzy > 4EroNQwJWd > a=ptime:20 > ----------------------------------------------------------- > ------------- > tport.c:2296 tport_set_secondary_timer() tport(0x7fa14000a110): reset timer > nta.c:8304 outgoing_send() nta: sent INVITE (106957108) to tcp/1.2.3.153 > [gw]:5111 > tport.c:4160 tport_pend() tport_pend(0x7fa14000a110): pending > 0x7fa140019990 for tcp/1.2.3.153[gw]:5111 (already 0) > nua_session.c:4139 signal_call_state_change() nua(0x7fa100006de0): call > state changed: init -> calling, sent offer > soa.c:1270 soa_get_local_sdp() soa_get_local_sdp(static::0x7fa14000f210, > [0x7fa16690eab0], [0x7fa16690eab8], [(nil)]) called > nua_stack.c:269 nua_stack_event() nua(0x7fa100006de0): event i_state > INVITE sent > 2017-05-12 10:52:25.432321 [DEBUG] switch_core_state_machine.c:662 > (sofia/external/%2B16151920015 at 1.2.3.153[gw]:5111) State CONSUME_MEDIA > 2017-05-12 10:52:25.432321 [DEBUG] switch_core_state_machine.c:662 > (sofia/external/%2B16151920015 at 1.2.3.153[gw]:5111) State CONSUME_MEDIA > going to sleep > nua_stack.c:359 nua_application_event() nua: nua_application_event: > entering > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > 2017-05-12 10:52:25.432321 [DEBUG] sofia.c:7048 Channel sofia/external/% > 2B16151920015 at 1.2.3.153[gw]:5111 entering state [calling][0] > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > tport.c:2773 tport_wakeup() tport_wakeup(0x7fa14000a110): events IN > tport.c:2864 tport_recv_event() tport_recv_event(0x7fa14000a110) > tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fa14000a110) msg > 0x7fa140016780 from (tcp/1.2.3.153[gw]:5111) has 369 bytes, veclen = 1 > recv 369 bytes from tcp/[1.2.3.153[gw]]:5111 at 10:52:25.447037: > ----------------------------------------------------------- > ------------- > SIP/2.0 100 trying -- your call is important to us > Via: SIP/2.0/TCP 1.2.3.43[fs]:5111;branch=z9hG4bKyK3rSg8XK3X1B > From: "soluto" ;tag=Ua2NXSa4mF7rD > To: > Call-ID: ecef602a-b1a3-1235-749f-0cc47a205c6e > CSeq: 106957108 INVITE > Server: kamailio (4.2.8 (x86_64/linux)) > Content-Length: 0 > > ----------------------------------------------------------- > ------------- > tport.c:3023 tport_deliver() tport_deliver(0x7fa14000a110): msg > 0x7fa140016780 (369 bytes) from tcp/1.2.3.153[gw]:5111/sip next=(nil) > nta.c:3299 agent_recv_response() nta: received 100 trying -- your call is > important to us for INVITE (106957108) > nta.c:3366 agent_recv_response() nta: 100 trying -- your call is important > to us is going to a transaction > nta.c:9564 outgoing_estimate_delay() nta_outgoing: RTT is 5.235 ms > tport.c:4222 tport_release() tport_release(0x7fa14000a110): 0x7fa140019990 > by 0x7fa1400132d0 with 0x7fa140016780 (preliminary) > tport.c:2296 tport_set_secondary_timer() tport(0x7fa14000a110): reset timer > tport.c:2773 tport_wakeup() tport_wakeup(0x7fa14000a110): events IN > tport.c:2864 tport_recv_event() tport_recv_event(0x7fa14000a110) > tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fa14000a110) msg > 0x7fa140016780 from (tcp/1.2.3.153[gw]:5111) has 1125 bytes, veclen = 1 > recv 1125 bytes from tcp/[1.2.3.153[gw]]:5111 at 10:52:25.597905: > ----------------------------------------------------------- > ------------- > SIP/2.0 180 Ringing > Via: SIP/2.0/TCP 1.2.3.43[fs]:5111;branch=z9hG4bKyK3rSg8XK3X1B > From: "soluto" ;tag=Ua2NXSa4mF7rD > To: > ;tag=SDbqa6299-801c7bd26943e712b98591536c00 > Call-ID: ecef602a-b1a3-1235-749f-0cc47a205c6e > CSeq: 106957108 INVITE > Record-Route: > Record-Route: > P-Asserted-Identity: > P-AV-Message-Id: 1_2 > Server: Avaya CM/R016x.03.0.124.0 AVAYA-SM-7.0.0.0.700007 > Contact: 97d8-000c292eaf17;transport=tls> > Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, NOTIFY, REFER, INFO, UPDATE > Supported: histinfo, join, replaces, timer > Content-Type: application/sdp > Content-Length: 282 > > v=0 > o=- 2716084251 2 IN IP4 1.2.3.165[gw-rtp-ip] > s=- > c=IN IP4 1.2.3.165[gw-rtp-ip] > b=AS:64 > t=0 0 > m=audio 14974 RTP/SAVP 0 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=sendrecv > a=ptime:20 > a=crypto:5 AES_CM_128_HMAC_SHA1_80 inline:NE/0Ttf1pbNEUyUUxxILCBQGC/ > taRBRyBGFbsggE > ----------------------------------------------------------- > ------------- > tport.c:3023 tport_deliver() tport_deliver(0x7fa14000a110): msg > 0x7fa140016780 (1125 bytes) from tcp/1.2.3.153[gw]:5111/sip next=(nil) > nta.c:3299 agent_recv_response() nta: received 180 Ringing for INVITE > (106957108) > nta.c:3366 agent_recv_response() nta: 180 Ringing is going to a transaction > tport.c:4222 tport_release() tport_release(0x7fa14000a110): 0x7fa140019990 > by 0x7fa1400132d0 with 0x7fa140016780 (preliminary) > soa.c:1171 soa_set_remote_sdp() soa_set_remote_sdp(static::0x7fa14000f210, > (nil), 0x7fa14001638b, 282) called > soa.c:1595 soa_process_answer() soa_process_answer(static::0x7fa14000f210) > called > soa_static.c:1148 offer_answer_step() soa_static_offer_answer_action(0x7fa14000f210, > soa_process_answer): called > soa_static.c:1029 soa_sdp_mode_set() soa_sdp_mode_set(0x7fa140010c50, > 0x7fa14001d5c0, ""): called > soa_static.c:1304 offer_answer_step() soa_static(0x7fa14000f210, > soa_process_answer): upgrade codecs with remote description > soa_static.c:1446 offer_answer_step() soa_static(0x7fa14000f210, > soa_process_answer): storing local description > soa.c:1730 soa_activate() soa_activate(static::0x7fa14000f210, (nil)) > called > nua_session.c:988 nua_session_client_response() nua(0x7fa100006de0): > INVITE: processed SDP answer in 180 Ringing > nua_stack.c:271 nua_stack_event() nua(0x7fa100006de0): event r_invite 180 > Ringing > nua_session.c:4139 signal_call_state_change() nua(0x7fa100006de0): call > state changed: calling -> proceeding, received answer > soa.c:1098 soa_get_remote_sdp() soa_get_remote_sdp(static::0x7fa14000f210, > [0x7fa16690e310], [0x7fa16690e318], [(nil)]) called > soa.c:616 soa_get_params() soa_get_params(static::0x7fa14000f210, ...) > called > nua_stack.c:359 nua_application_event() nua: nua_application_event: > entering > nua_stack.c:271 nua_stack_event() nua(0x7fa100006de0): event i_state 180 > Ringing > tport.c:2296 tport_set_secondary_timer() tport(0x7fa14000a110): reset timer > nua_stack.c:359 nua_application_event() nua: nua_application_event: > entering > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > 2017-05-12 10:52:25.592332 [INFO] sofia.c:1279 sofia/external/% > 2B16151920015 at 1.2.3.153[gw]:5111 Update Callee ID to "Outbound Call" > <6151920015> > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > 2017-05-12 10:52:25.592332 [DEBUG] sofia.c:7048 Channel sofia/external/% > 2B16151920015 at 1.2.3.153[gw]:5111 entering state [proceeding][180] > 2017-05-12 10:52:25.592332 [DEBUG] sofia.c:7058 Remote SDP: > v=0 > o=- 2716084251 2 IN IP4 1.2.3.165[gw-rtp-ip] > s=- > c=IN IP4 1.2.3.165[gw-rtp-ip] > b=AS:64 > t=0 0 > m=audio 14974 RTP/SAVP 0 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=ptime:20 > a=crypto:5 AES_CM_128_HMAC_SHA1_80 inline:NE/0Ttf1pbNEUyUUxxILCBQGC/ > taRBRyBGFbsggE > > 2017-05-12 10:52:25.592332 [DEBUG] switch_core_media.c:1391 looking for > crypto suite [AEAD_AES_256_GCM_8] in [5 AES_CM_128_HMAC_SHA1_80 inline:NE/ > 0Ttf1pbNEUyUUxxILCBQGC/taRBRyBGFbsggE] > 2017-05-12 10:52:25.592332 [DEBUG] switch_core_media.c:1391 looking for > crypto suite [AEAD_AES_128_GCM_8] in [5 AES_CM_128_HMAC_SHA1_80 inline:NE/ > 0Ttf1pbNEUyUUxxILCBQGC/taRBRyBGFbsggE] > 2017-05-12 10:52:25.592332 [DEBUG] switch_core_media.c:1391 looking for > crypto suite [AES_CM_256_HMAC_SHA1_80] in [5 AES_CM_128_HMAC_SHA1_80 > inline:NE/0Ttf1pbNEUyUUxxILCBQGC/taRBRyBGFbsggE] > 2017-05-12 10:52:25.592332 [DEBUG] switch_core_media.c:1391 looking for > crypto suite [AES_CM_192_HMAC_SHA1_80] in [5 AES_CM_128_HMAC_SHA1_80 > inline:NE/0Ttf1pbNEUyUUxxILCBQGC/taRBRyBGFbsggE] > 2017-05-12 10:52:25.592332 [DEBUG] switch_core_media.c:1391 looking for > crypto suite [AES_CM_128_HMAC_SHA1_80] in [5 AES_CM_128_HMAC_SHA1_80 > inline:NE/0Ttf1pbNEUyUUxxILCBQGC/taRBRyBGFbsggE] > 2017-05-12 10:52:25.592332 [DEBUG] switch_core_media.c:1396 Found suite > AES_CM_128_HMAC_SHA1_80 > 2017-05-12 10:52:25.592332 [DEBUG] switch_core_media.c:1462 Set Remote Key > [5 AES_CM_128_HMAC_SHA1_80 inline:NE/0Ttf1pbNEUyUUxxILCBQGC/ > taRBRyBGFbsggE] > 2017-05-12 10:52:25.592332 [DEBUG] switch_core_media.c:4436 Audio Codec > Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] > 2017-05-12 10:52:25.592332 [DEBUG] switch_core_media.c:4436 Audio Codec > Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] > 2017-05-12 10:52:25.592332 [DEBUG] switch_core_media.c:4491 Audio Codec > Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match > 2017-05-12 10:52:25.592332 [DEBUG] switch_core_media.c:4352 Set > telephone-event payload to 101 at 8000 > 2017-05-12 10:52:25.592332 [DEBUG] switch_core_media.c:3056 Set Codec > sofia/external/%2B16151920015 at 1.2.3.153[gw]:5111 PCMU/8000 20 ms 160 > samples 64000 bits 1 channels > 2017-05-12 10:52:25.592332 [DEBUG] switch_core_codec.c:111 sofia/external/% > 2B16151920015 at 1.2.3.153[gw]:5111 Original read codec set to PCMU:0 > 2017-05-12 10:52:25.592332 [DEBUG] switch_core_media.c:4695 Set > telephone-event payload to 101 at 8000 > 2017-05-12 10:52:25.592332 [DEBUG] switch_core_media.c:4754 > sofia/external/%2B16151920015 at 1.2.3.153[gw]:5111 Set 2833 dtmf send > payload to 101 recv payload to 101 > 2017-05-12 10:52:25.592332 [DEBUG] switch_core_media.c:6865 AUDIO RTP > [sofia/external/%2B16151920015 at 1.2.3.153[gw]:5111] 1.2.3.43[fs] port > 27354 -> 1.2.3.165[gw-rtp-ip] port 14974 codec: 0 ms: 20 > 2017-05-12 10:52:25.592332 [DEBUG] switch_rtp.c:4096 Starting timer [soft] > 160 bytes per 20ms > 2017-05-12 10:52:25.592332 [DEBUG] switch_core_media.c:7166 > sofia/external/%2B16151920015 at 1.2.3.153[gw]:5111 Set 2833 dtmf send > payload to 101 > 2017-05-12 10:52:25.592332 [DEBUG] switch_core_media.c:7173 > sofia/external/%2B16151920015 at 1.2.3.153[gw]:5111 Set 2833 dtmf receive > payload to 101 > 2017-05-12 10:52:25.592332 [DEBUG] switch_core_media.c:7196 > sofia/external/%2B16151920015 at 1.2.3.153[gw]:5111 Set rtp dtmf delay to 40 > 2017-05-12 10:52:25.592332 [INFO] switch_rtp.c:3908 Activating audio > Secure RTP SEND > 2017-05-12 10:52:25.592332 [INFO] switch_rtp.c:3886 Activating audio > Secure RTP RECV > 2017-05-12 10:52:25.592332 [DEBUG] switch_core_sqldb.c:2617 Secure Type: > srtp:sdes:AES_CM_128_HMAC_SHA1_80 > 2017-05-12 10:52:25.592332 [DEBUG] switch_core_sqldb.c:2617 Secure Type: > srtp:sdes:AES_CM_128_HMAC_SHA1_80 > 2017-05-12 10:52:25.592332 [NOTICE] sofia_media.c:92 Pre-Answer > sofia/external/%2B16151920015 at 1.2.3.153[gw]:5111! > 2017-05-12 10:52:25.592332 [DEBUG] switch_channel.c:3473 (sofia/external/% > 2B16151920015 at 1.2.3.153[gw]:5111) Callstate Change DOWN -> EARLY > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > 2017-05-12 10:52:25.592332 [INFO] switch_ivr_originate.c:3639 Sending > early media > 2017-05-12 10:52:25.592332 [DEBUG] switch_core_media.c:4436 Audio Codec > Compare [opus:107:48000:20:0:2]/[G722:9:8000:20:64000:1] > 2017-05-12 10:52:25.592332 [DEBUG] switch_core_media.c:4436 Audio Codec > Compare [opus:107:48000:20:0:2]/[PCMU:0:8000:20:64000:1] > 2017-05-12 10:52:25.592332 [DEBUG] switch_core_media.c:4436 Audio Codec > Compare [opus:107:48000:20:0:2]/[PCMA:8:8000:20:64000:1] > 2017-05-12 10:52:25.592332 [DEBUG] switch_core_media.c:4352 Set > telephone-event payload to 96 at 8000 > 2017-05-12 10:52:25.592332 [DEBUG] switch_core_media.c:4754 sofia/external/ > soluto at 1.2.3.25[registrar-kamailio] Set 2833 dtmf send payload to 96 recv > payload to 96 > 2017-05-12 10:52:25.592332 [ERR] mod_sofia.c:2342 CODEC NEGOTIATION > ERROR. SDP: > v=0 > o=- 3703575145 3703575145 IN IP4 192.168.1.171 > s=pjmedia > b=AS:84 > t=0 0 > a=X-nat:0 > m=audio 33228 RTP/SAVP 107 96 > c=IN IP4 67.205.173.126 > b=TIAS:64000 > a=rtpmap:107 opus/48000/2 > a=rtpmap:96 telephone-event/8000 > a=fmtp:96 0-16 > a=rtcp:33229 > a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:4h3KXpeuR1qSWxR4+/ > XCLDtCO3UGqrO5gXK6HIGF > a=setup:actpass > a=fingerprint:sha-1 7B:5A:DF:A5:CC:88:FD:CA:88:C2: > A8:D1:73:1F:3A:D8:29:9F:B3:D0 > a=ice-ufrag:T0L06kG3 > a=ice-pwd:uLNAX5ppVlJmXAdY7E4SZKsSKd > a=candidate:eTePfBnNsoctce5m 1 UDP 2130706431 67.205.173.126 33228 typ host > a=candidate:eTePfBnNsoctce5m 2 UDP 2130706430 67.205.173.126 33229 typ host > > 2017-05-12 10:52:25.592332 [NOTICE] switch_channel.c:3514 Hangup > sofia/external/soluto at 1.2.3.25[registrar-kamailio] [CS_EXECUTE] > [INCOMPATIBLE_DESTINATION] > 2017-05-12 10:52:25.592332 [DEBUG] switch_ivr_originate.c:3646 > sofia/external/soluto at 1.2.3.25[registrar-kamailio] Media Establishment > Failed. > 2017-05-12 10:52:25.592332 [NOTICE] switch_ivr_originate.c:3648 Hangup > sofia/external/%2B16151920015 at 1.2.3.153[gw]:5111 [CS_CONSUME_MEDIA] > [INCOMPATIBLE_DESTINATION] > 2017-05-12 10:52:25.592332 [DEBUG] switch_ivr_originate.c:3833 Originate > Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION] > 2017-05-12 10:52:25.592332 [DEBUG] switch_core_state_machine.c:584 > (sofia/external/%2B16151920015 at 1.2.3.153[gw]:5111) Running State Change > CS_HANGUP (Cur 2 Tot 76) > 2017-05-12 10:52:25.592332 [DEBUG] switch_core_state_machine.c:850 > (sofia/external/%2B16151920015 at 1.2.3.153[gw]:5111) Callstate Change EARLY > -> HANGUP > 2017-05-12 10:52:25.592332 [DEBUG] switch_core_state_machine.c:852 > (sofia/external/%2B16151920015 at 1.2.3.153[gw]:5111) State HANGUP > 2017-05-12 10:52:25.592332 [INFO] mod_dptools.c:3418 Originate Failed. > Cause: INCOMPATIBLE_DESTINATION > 2017-05-12 10:52:25.592332 [DEBUG] mod_sofia.c:438 Channel sofia/external/% > 2B16151920015 at 1.2.3.153[gw]:5111 hanging up, cause: > INCOMPATIBLE_DESTINATION > 2017-05-12 10:52:25.592332 [DEBUG] mod_sofia.c:502 Sending CANCEL to > sofia/external/%2B16151920015 at 1.2.3.153[gw]:5111 > nua.c:651 nua_cancel() nua: nua_cancel: entering > nua_stack.c:529 nua_signal() nua(0x7fa100006de0): sent signal r_cancel > 2017-05-12 10:52:25.592332 [DEBUG] switch_core_state_machine.c:60 > sofia/external/%2B16151920015 at 1.2.3.153[gw]:5111 Standard HANGUP, cause: > INCOMPATIBLE_DESTINATION > 2017-05-12 10:52:25.592332 [DEBUG] switch_core_state_machine.c:852 > (sofia/external/%2B16151920015 at 1.2.3.153[gw]:5111) State HANGUP going to > sleep > nua_stack.c:569 nua_stack_signal() nua(0x7fa100006de0): recv signal > r_cancel > 2017-05-12 10:52:25.592332 [DEBUG] switch_core_session.c:2814 > sofia/external/soluto at 1.2.3.25[registrar-kamailio] skip receive message > [APPLICATION_EXEC_COMPLETE] (channel is hungup already) > nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering > 2017-05-12 10:52:25.592332 [DEBUG] switch_core_state_machine.c:650 > (sofia/external/soluto at 1.2.3.25[registrar-kamailio]) State EXECUTE going > to sleep > soa.c:403 soa_set_params() soa_set_params(static::0x7fa14000f210, ...) > called > 2017-05-12 10:52:25.592332 [DEBUG] switch_core_state_machine.c:584 > (sofia/external/soluto at 1.2.3.25[registrar-kamailio]) Running State Change > CS_HANGUP (Cur 2 Tot 76) > 2017-05-12 10:52:25.592332 [DEBUG] switch_core_state_machine.c:619 > (sofia/external/%2B16151920015 at 1.2.3.153[gw]:5111) State Change CS_HANGUP > -> CS_REPORTING > tport.c:4588 tport_by_name() tport(0x7fa140004b30): found 0x7fa14000a110 > by name tcp/1.2.3.153[gw]:5111 > 2017-05-12 10:52:25.592332 [DEBUG] switch_core_state_machine.c:584 > (sofia/external/%2B16151920015 at 1.2.3.153[gw]:5111) Running State Change > CS_REPORTING (Cur 2 Tot 76) > tport.c:3257 tport_tsend() tport_tsend(0x7fa14000a110) tpn = tcp/1.2.3.153 > [gw]:5111 > tport.c:3594 tport_vsend() tport_vsend(0x7fa14000a110): 417 bytes of 417 > to tcp/1.2.3.153[gw]:5111 > tport.c:3492 tport_send_msg() tport_vsend returned 417 > send 417 bytes to tcp/[1.2.3.153[gw]]:5111 at 10:52:25.604158: > ----------------------------------------------------------- > ------------- > CANCEL sip:+16151920015 at 1.2.3.153[gw]:5111;transport=tcp SIP/2.0 > Via: SIP/2.0/TCP 1.2.3.43[fs]:5111;branch=z9hG4bKyK3rSg8XK3X1B > Max-Forwards: 68 > From: "soluto" ;tag=Ua2NXSa4mF7rD > To: > 2017-05-12 10:52:25.592332 [DEBUG] switch_core_state_machine.c:938 > (sofia/external/%2B16151920015 at 1.2.3.153[gw]:5111) State REPORTING > Call-ID: ecef602a-b1a3-1235-749f-0cc47a205c6e > CSeq: 106957108 CANCEL > Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" > Content-Length: 0 > > ? > > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 <+34%20669%2044%2083%2037> > > On Fri, May 12, 2017 at 12:41 PM, Colin Morelli > wrote: > >> All RTP related channel vars were renamed from sip_ to rtp_. Try >> rtp_secure_media. >> >> Best, >> Colin >> >> On Fri, May 12, 2017 at 6:24 AM David Villasmil < >> david.villasmil.work at gmail.com> wrote: >> >>> Hello Guys, >>> >>> So i've got this scenario where i'm receving in TCP+SRTP(opus) and I >>> need to send to the b-leg in TCP+STRP(PCMA)... i've enabled: >>> >>> >>> >>> >>> But the sdp is being went without srtp, what am I missing? >>> >>> >>> Regards, >>> >>> David Villasmil >>> email: david.villasmil.work at gmail.com >>> phone: +34669448337 <+34%20669%2044%2083%2037> >>> ? >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170512/ca17251b/attachment-0001.html From lexxua at gmail.com Fri May 12 15:19:26 2017 From: lexxua at gmail.com (Volodymyr Fedorov) Date: Fri, 12 May 2017 13:19:26 +0200 Subject: [Freeswitch-users] STRP In-Reply-To: References: Message-ID: Hi, maybe you have to change sip_ to rtp_ variable ? rtp_secure_media=true On Fri, May 12, 2017 at 12:21 PM, David Villasmil < david.villasmil.work at gmail.com> wrote: > Hello Guys, > > So i've got this scenario where i'm receving in TCP+SRTP(opus) and I need > to send to the b-leg in TCP+STRP(PCMA)... i've enabled: > > > > > But the sdp is being went without srtp, what am I missing? > > > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 <+34%20669%2044%2083%2037> > ? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Best regards, Volodymyr -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170512/df9d6195/attachment.html From david.villasmil.work at gmail.com Fri May 12 15:22:09 2017 From: david.villasmil.work at gmail.com (David Villasmil) Date: Fri, 12 May 2017 11:22:09 +0000 Subject: [Freeswitch-users] STRP In-Reply-To: References: Message-ID: Yep, it's working great, thanks! On Fri, May 12, 2017 at 1:20 PM Volodymyr Fedorov wrote: > Hi, > maybe you have to change sip_ to rtp_ variable ? > rtp_secure_media=true > > > > On Fri, May 12, 2017 at 12:21 PM, David Villasmil < > david.villasmil.work at gmail.com> wrote: > >> Hello Guys, >> >> So i've got this scenario where i'm receving in TCP+SRTP(opus) and I need >> to send to the b-leg in TCP+STRP(PCMA)... i've enabled: >> >> > data="{sip_secure_media=true,sdp_secure_savp_only=true,absolute_codec_string=^^:PCMA:PCMU}sofia/external/${destination_number}@myserver;transport=tcp" >> /> >> >> >> But the sdp is being went without srtp, what am I missing? >> >> >> Regards, >> >> David Villasmil >> email: david.villasmil.work at gmail.com >> phone: +34669448337 <+34%20669%2044%2083%2037> >> ? >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Best regards, > Volodymyr > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170512/7e853b30/attachment.html From david.villasmil.work at gmail.com Fri May 12 15:23:30 2017 From: david.villasmil.work at gmail.com (David Villasmil) Date: Fri, 12 May 2017 11:23:30 +0000 Subject: [Freeswitch-users] mod_distribution feature In-Reply-To: <3e03ed2f-54eb-48a7-c78d-9df74764e98b@wirelessmundi.com> References: <3e03ed2f-54eb-48a7-c78d-9df74764e98b@wirelessmundi.com> Message-ID: Hello, Contributions are always welcomed! Or else you can offer a bounty :) David On Fri, May 12, 2017 at 12:56 PM Antonio Silva wrote: > Hi all, > > Just found mod_distribution and is pretty cool feature to have. Right now > it distributes calls to gateways in a weighted round-robin fashion, could > it be cool to have > *Weighted Random Distribution. *i found this paper explaining an > implementation in python: > https://www.electricmonk.nl/log/2009/12/23/weighted-random-distribution/ > it could be ported to mod_distributor. > > What you guys think about it? > > -- > > Saludos / Regards / Cumprimentos, > Ant?nio 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170512/a922b825/attachment.html From kkothari157 at gmail.com Fri May 12 16:48:14 2017 From: kkothari157 at gmail.com (Ketan Kothari) Date: Fri, 12 May 2017 18:18:14 +0530 Subject: [Freeswitch-users] Voicemail not detecting DTMF Message-ID: Hello, I'm trying to call voicemail box number 7777 to hear the recorded messages if any, and its asking for password but when enter digit its not showing in freeswitch logs after some seconds its again asking for password. https://pastebin.com/ypZFvXD2 after the above log if i press digit '1' i should get something like this, [DEBUG] switch_rtp.c:6927 RTP RECV DTMF 1:960 But i am getting nothing. However at the time of leaving the voicemail, the DTMF is detected fine. what could be the issue? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170512/1691a866/attachment.html From lwahlmeier at gmail.com Fri May 12 19:42:03 2017 From: lwahlmeier at gmail.com (Luke Wahlmeier) Date: Fri, 12 May 2017 09:42:03 -0600 Subject: [Freeswitch-users] WSS SSL errors "decryption failed or bad record mac" under load In-Reply-To: References: Message-ID: Just got done testing this on v1.6 head and master, both seem to still have this issue. This box is using libssl version 1.0.1t-1+deb8u6. I am gonna start digging more into the ws/wss/sofia code to see if I can figure it out. Any suggestions on debugging this would be appreciated. Thanks Luke On Thu, May 11, 2017 at 5:12 PM, Luke Wahlmeier wrote: > Its just in our isolated lab, pretty normal dell xeon server running > Jessie 8.6. I just want to get it building on the same box I am testing > with so setting that all up. > > I was able to reproduce it w/o DTLS/Srtp. here is a much simpler and > cleaned up version of the python script. > > > > On Thu, May 11, 2017 at 4:34 PM, Michael Jerris wrote: > >> what is ?this environment? ? >> >> On May 11, 2017, at 6:31 PM, Luke Wahlmeier wrote: >> >> Yeah I can usually get it to happen within about 5 minutes or so of >> testing. Still getting all setup to build freeswitch in this environment, >> but I should have it working by tomorrow. I will try more w/o dtls/srtp as >> well and make sure it does not need to be on. >> >> Thanks >> Luke >> >> On Thu, May 11, 2017 at 4:20 PM, Michael Jerris wrote: >> >>> if you can reproduce this reliably, i?d try master as well. Unless this >>> is a bug in openssl, i can?t imagine how dtls would come into play in >>> something like this. >>> >>> > On May 11, 2017, at 5:48 PM, Luke Wahlmeier >>> wrote: >>> > >>> > I keep semi-regularly running into issues using the wss transport when >>> using dtls/strp/ice. This is on the latest 1.6.17~34~0fc0946 on Debian >>> jessie, but I am pretty sure it was happening on the last couple releases >>> as well. >>> > >>> > It seems like something bad/wrong happens to the encrypted data going >>> over the websocket coming from freeswitch when more then 1 websocket >>> connection are going and so far ice/srtp/dtls also seem to be needed in the >>> invite to duplicate it. >>> > >>> > I have tried many different languages and network/ssl stacks and keep >>> running into this. It is always on data coming in from freeswitch on the >>> websocket connection, and its very very random. Sometimes I will get it 20 >>> times in a row, other times it takes thousands of connections/sessions >>> before it happen. It also, obviously, completely goes away if I use plain >>> ws instead wss. >>> > >>> > Here are the errors: >>> > python: >>> > SSLError: [SSL: DECRYPTION_FAILED_OR_BAD_RECORD_MAC] decryption >>> failed or bad record mac (_ssl.c:1750) >>> > c/c++ (stunnel4): >>> > SSL_read: 1408F119: error:1408F119:SSL routines:SSL3_GET_RECORD:decryption >>> failed or bad record mac >>> > Java: >>> > java.lang.IllegalArgumentException: Bad arguments >>> > at javax.crypto.Mac.update(Mac.java:509) >>> > at sun.security.ssl.MAC.compute(MAC.java:135) >>> > at sun.security.ssl.InputRecord.checkMacTags(InputRecord.java:265) >>> > at sun.security.ssl.InputRecord.decrypt(InputRecord.java:216) >>> > at sun.security.ssl.EngineInputRecord.decrypt(EngineInputRecord >>> .java:177) >>> > at sun.security.ssl.SSLEngineImpl.readRecord(SSLEngineImpl.java >>> :974) >>> > at sun.security.ssl.SSLEngineImpl.readNetRecord(SSLEngineImpl.j >>> ava:907) >>> > at sun.security.ssl.SSLEngineImpl.unwrap(SSLEngineImpl.java:781) >>> > at javax.net.ssl.SSLEngine.unwrap(SSLEngine.java:624) >>> > >>> > Attached are a simple python script to do the load, my dialplan and >>> sip_profile. The python script can take a few runs before it see the >>> error, and I know its not completing the sip or rtp, but even if it does >>> this still happens. >>> > >>> > I have also looked at libsofia-sip-ua/tport/ws.c and I dont see >>> anything obvious. I am getting setup to build v1.6 head and test this any >>> guidance on ways I can trouble shoot this better or requests for more info >>> are very welcome. >>> > >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170512/0093ade4/attachment-0001.html From mike at jerris.com Fri May 12 20:20:30 2017 From: mike at jerris.com (Michael Jerris) Date: Fri, 12 May 2017 12:20:30 -0400 Subject: [Freeswitch-users] WSS SSL errors "decryption failed or bad record mac" under load In-Reply-To: References: Message-ID: <358E7E05-A60A-463A-911F-A921F6CDEAA2@jerris.com> test on master.. work a similar test for verto maybe, this might have to do with sip specifically trying to keep state. Might make sense to build something out of libks as it has basically the same web socket code, and has both client and server web socket support in it, to do a ?real? test?, instead of this fake sip without any state over web sockets. > On May 12, 2017, at 11:42 AM, Luke Wahlmeier wrote: > > Just got done testing this on v1.6 head and master, both seem to still have this issue. This box is using libssl version 1.0.1t-1+deb8u6. I am gonna start digging more into the ws/wss/sofia code to see if I can figure it out. Any suggestions on debugging this would be appreciated. > > Thanks > Luke > > On Thu, May 11, 2017 at 5:12 PM, Luke Wahlmeier > wrote: > Its just in our isolated lab, pretty normal dell xeon server running Jessie 8.6. I just want to get it building on the same box I am testing with so setting that all up. > > I was able to reproduce it w/o DTLS/Srtp. here is a much simpler and cleaned up version of the python script. > > > > On Thu, May 11, 2017 at 4:34 PM, Michael Jerris > wrote: > what is ?this environment? ? > >> On May 11, 2017, at 6:31 PM, Luke Wahlmeier > wrote: >> >> Yeah I can usually get it to happen within about 5 minutes or so of testing. Still getting all setup to build freeswitch in this environment, but I should have it working by tomorrow. I will try more w/o dtls/srtp as well and make sure it does not need to be on. >> >> Thanks >> Luke >> >> On Thu, May 11, 2017 at 4:20 PM, Michael Jerris > wrote: >> if you can reproduce this reliably, i?d try master as well. Unless this is a bug in openssl, i can?t imagine how dtls would come into play in something like this. >> >> > On May 11, 2017, at 5:48 PM, Luke Wahlmeier > wrote: >> > >> > I keep semi-regularly running into issues using the wss transport when using dtls/strp/ice. This is on the latest 1.6.17~34~0fc0946 on Debian jessie, but I am pretty sure it was happening on the last couple releases as well. >> > >> > It seems like something bad/wrong happens to the encrypted data going over the websocket coming from freeswitch when more then 1 websocket connection are going and so far ice/srtp/dtls also seem to be needed in the invite to duplicate it. >> > >> > I have tried many different languages and network/ssl stacks and keep running into this. It is always on data coming in from freeswitch on the websocket connection, and its very very random. Sometimes I will get it 20 times in a row, other times it takes thousands of connections/sessions before it happen. It also, obviously, completely goes away if I use plain ws instead wss. >> > >> > Here are the errors: >> > python: >> > SSLError: [SSL: DECRYPTION_FAILED_OR_BAD_RECORD_MAC] decryption failed or bad record mac (_ssl.c:1750) >> > c/c++ (stunnel4): >> > SSL_read: 1408F119: error:1408F119:SSL routines:SSL3_GET_RECORD:decryption failed or bad record mac >> > Java: >> > java.lang.IllegalArgumentException: Bad arguments >> > at javax.crypto.Mac.update(Mac.java:509) >> > at sun.security.ssl.MAC.compute(MAC.java:135) >> > at sun.security.ssl.InputRecord.checkMacTags(InputRecord.java:265) >> > at sun.security.ssl.InputRecord.decrypt(InputRecord.java:216) >> > at sun.security.ssl.EngineInputRecord.decrypt(EngineInputRecord.java:177) >> > at sun.security.ssl.SSLEngineImpl.readRecord(SSLEngineImpl.java:974) >> > at sun.security.ssl.SSLEngineImpl.readNetRecord(SSLEngineImpl.java:907) >> > at sun.security.ssl.SSLEngineImpl.unwrap(SSLEngineImpl.java:781) >> > at javax.net.ssl.SSLEngine.unwrap(SSLEngine.java:624) >> > >> > Attached are a simple python script to do the load, my dialplan and sip_profile. The python script can take a few runs before it see the error, and I know its not completing the sip or rtp, but even if it does this still happens. >> > >> > I have also looked at libsofia-sip-ua/tport/ws.c and I dont see anything obvious. I am getting setup to build v1.6 head and test this any guidance on ways I can trouble shoot this better or requests for more info are very welcome. >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170512/6d9d3fc5/attachment.html From lwahlmeier at gmail.com Fri May 12 21:51:23 2017 From: lwahlmeier at gmail.com (Luke Wahlmeier) Date: Fri, 12 May 2017 11:51:23 -0600 Subject: [Freeswitch-users] WSS SSL errors "decryption failed or bad record mac" under load In-Reply-To: <358E7E05-A60A-463A-911F-A921F6CDEAA2@jerris.com> References: <358E7E05-A60A-463A-911F-A921F6CDEAA2@jerris.com> Message-ID: Thanks Michael, I am more then happy to setup something with libks if needed. I have figured out some more however. It appears that this only happens when a wss connections session has not fully established and is cleaning up because of timing out. The problem is that it causes another wss connection it to get this ssl error, even if that other wss connection has a fully established and running audio session. It is important to note it does not seem to interrupt audio just the wss sip channel, which I am fairly sure can be reestablished for that audio session w/o an issue. The sessions that is being cleaned up sends the logs messages as its doing it: 2017-05-12 17:32:46.607768 [NOTICE] sofia.c:8438 Hangup sofia/websocket/nobody at 1LF3F6I924P9WH6U [CS_EXECUTE] [NORMAL_UNSPECIFIED] 2017-05-12 17:32:46.627768 [INFO] conference_loop.c:1621 Channel leaving conference, cause: NORMAL_UNSPECIFIED 2017-05-12 17:32:46.627768 [NOTICE] switch_core_session.c:1730 Session 46 (sofia/websocket/nobody at 1LF3F6I924P9WH6U) Ended 2017-05-12 17:32:46.627768 [NOTICE] switch_core_session.c:1734 Close Channel sofia/websocket/nobody at 1LF3F6I924P9WH6U [CS_DESTROY] I have attached the updated python script, it can duplicate this every time now with only 2 connections. I verified with a webRTC client that if I initiate this first connection in the script, let it close, then connect the webRTC client and get full audio, once the first session from the script times out it causes the webRTC wss connection to get an error and close. The webRTC connection is in chrome with sip.js. Sorry the python script is so nasty, was working through any possible duplicated sip session stuff in it to make sure that was not why it was hitting the second connection. On Fri, May 12, 2017 at 10:20 AM, Michael Jerris wrote: > test on master.. work a similar test for verto maybe, this might have to > do with sip specifically trying to keep state. Might make sense to build > something out of libks as it has basically the same web socket code, and > has both client and server web socket support in it, to do a ?real? test?, > instead of this fake sip without any state over web sockets. > > > On May 12, 2017, at 11:42 AM, Luke Wahlmeier wrote: > > Just got done testing this on v1.6 head and master, both seem to still > have this issue. This box is using libssl version 1.0.1t-1+deb8u6. I am > gonna start digging more into the ws/wss/sofia code to see if I can figure > it out. Any suggestions on debugging this would be appreciated. > > Thanks > Luke > > On Thu, May 11, 2017 at 5:12 PM, Luke Wahlmeier > wrote: > >> Its just in our isolated lab, pretty normal dell xeon server running >> Jessie 8.6. I just want to get it building on the same box I am testing >> with so setting that all up. >> >> I was able to reproduce it w/o DTLS/Srtp. here is a much simpler and >> cleaned up version of the python script. >> >> >> >> On Thu, May 11, 2017 at 4:34 PM, Michael Jerris wrote: >> >>> what is ?this environment? ? >>> >>> On May 11, 2017, at 6:31 PM, Luke Wahlmeier >>> wrote: >>> >>> Yeah I can usually get it to happen within about 5 minutes or so of >>> testing. Still getting all setup to build freeswitch in this environment, >>> but I should have it working by tomorrow. I will try more w/o dtls/srtp as >>> well and make sure it does not need to be on. >>> >>> Thanks >>> Luke >>> >>> On Thu, May 11, 2017 at 4:20 PM, Michael Jerris wrote: >>> >>>> if you can reproduce this reliably, i?d try master as well. Unless >>>> this is a bug in openssl, i can?t imagine how dtls would come into play in >>>> something like this. >>>> >>>> > On May 11, 2017, at 5:48 PM, Luke Wahlmeier >>>> wrote: >>>> > >>>> > I keep semi-regularly running into issues using the wss transport >>>> when using dtls/strp/ice. This is on the latest 1.6.17~34~0fc0946 on >>>> Debian jessie, but I am pretty sure it was happening on the last couple >>>> releases as well. >>>> > >>>> > It seems like something bad/wrong happens to the encrypted data going >>>> over the websocket coming from freeswitch when more then 1 websocket >>>> connection are going and so far ice/srtp/dtls also seem to be needed in the >>>> invite to duplicate it. >>>> > >>>> > I have tried many different languages and network/ssl stacks and keep >>>> running into this. It is always on data coming in from freeswitch on the >>>> websocket connection, and its very very random. Sometimes I will get it 20 >>>> times in a row, other times it takes thousands of connections/sessions >>>> before it happen. It also, obviously, completely goes away if I use plain >>>> ws instead wss. >>>> > >>>> > Here are the errors: >>>> > python: >>>> > SSLError: [SSL: DECRYPTION_FAILED_OR_BAD_RECORD_MAC] decryption >>>> failed or bad record mac (_ssl.c:1750) >>>> > c/c++ (stunnel4): >>>> > SSL_read: 1408F119: error:1408F119:SSL routines:SSL3_GET_RECORD:decryption >>>> failed or bad record mac >>>> > Java: >>>> > java.lang.IllegalArgumentException: Bad arguments >>>> > at javax.crypto.Mac.update(Mac.java:509) >>>> > at sun.security.ssl.MAC.compute(MAC.java:135) >>>> > at sun.security.ssl.InputRecord.checkMacTags(InputRecord.java:2 >>>> 65) >>>> > at sun.security.ssl.InputRecord.decrypt(InputRecord.java:216) >>>> > at sun.security.ssl.EngineInputRecord.decrypt(EngineInputRecord >>>> .java:177) >>>> > at sun.security.ssl.SSLEngineImpl.readRecord(SSLEngineImpl.java >>>> :974) >>>> > at sun.security.ssl.SSLEngineImpl.readNetRecord(SSLEngineImpl.j >>>> ava:907) >>>> > at sun.security.ssl.SSLEngineImpl.unwrap(SSLEngineImpl.java:781) >>>> > at javax.net.ssl.SSLEngine.unwrap(SSLEngine.java:624) >>>> > >>>> > Attached are a simple python script to do the load, my dialplan and >>>> sip_profile. The python script can take a few runs before it see the >>>> error, and I know its not completing the sip or rtp, but even if it does >>>> this still happens. >>>> > >>>> > I have also looked at libsofia-sip-ua/tport/ws.c and I dont see >>>> anything obvious. I am getting setup to build v1.6 head and test this any >>>> guidance on ways I can trouble shoot this better or requests for more info >>>> are very welcome. >>>> > >>>> >>> >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170512/fe6786fd/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: load.py Type: text/x-python Size: 2976 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170512/fe6786fd/attachment-0001.py From lwahlmeier at gmail.com Fri May 12 22:03:07 2017 From: lwahlmeier at gmail.com (Luke Wahlmeier) Date: Fri, 12 May 2017 12:03:07 -0600 Subject: [Freeswitch-users] WSS SSL errors "decryption failed or bad record mac" under load In-Reply-To: References: <358E7E05-A60A-463A-911F-A921F6CDEAA2@jerris.com> Message-ID: Here is the debug logs from this happening: f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [NOTICE] switch_channel.c:1104 New Channel sofia/websocket/nobody at 52LX8LP8BBWG6990 [f9a97c98-373b-11e7-9136-499bb33ea37e] f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] switch_core_state_machine.c:584 (sofia/websocket/nobody at 52LX8LP8BBWG6990) Running State Change CS_NEW (Cur 1 Tot 1) f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] sofia.c:10028 sofia/websocket/nobody at 52LX8LP8BBWG6990 receiving invite from 192.168.56.151:53442 version: 1.9.0 git db24869 2017-05-11 18:22:45Z 64bit f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] sofia.c:11325 Setting NAT mode based on websockets f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] sofia.c:7247 Channel sofia/websocket/nobody at 52LX8LP8BBWG6990 entering state [received][100] f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] sofia.c:7257 Remote SDP: f9a97c98-373b-11e7-9136-499bb33ea37e v=0 f9a97c98-373b-11e7-9136-499bb33ea37e o=- 196478633 2 IN IP4 192.168.56.151 f9a97c98-373b-11e7-9136-499bb33ea37e s=- f9a97c98-373b-11e7-9136-499bb33ea37e c=IN IP4 192.168.56.151 f9a97c98-373b-11e7-9136-499bb33ea37e t=0 0 f9a97c98-373b-11e7-9136-499bb33ea37e m=audio 42504 RTP/AVP 0 100 f9a97c98-373b-11e7-9136-499bb33ea37e a=rtpmap:0 PCMU/8000 f9a97c98-373b-11e7-9136-499bb33ea37e a=maxptime:20 f9a97c98-373b-11e7-9136-499bb33ea37e f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] switch_core_media.c:5110 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] switch_core_media.c:5165 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] switch_core_media.c:5110 Audio Codec Compare [:100:0:20:0:1]/[PCMU:0:8000:20:64000:1] f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] switch_core_media.c:3425 Set Codec sofia/websocket/nobody at 52LX8LP8BBWG6990 PCMU/8000 20 ms 160 samples 64000 bits 1 channels f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] switch_core_codec.c:111 sofia/websocket/nobody at 52LX8LP8BBWG6990 Original read codec set to PCMU:0 f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] switch_core_media.c:5398 No 2833 in SDP. Liberal DTMF mode adding 101 as telephone-event. f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] switch_core_media.c:5427 sofia/websocket/nobody at 52LX8LP8BBWG6990 Set 2833 dtmf send payload to 101 recv payload to 101 f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] sofia.c:7670 (sofia/websocket/nobody at 52LX8LP8BBWG6990) State Change CS_NEW -> CS_INIT f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] switch_core_state_machine.c:603 (sofia/websocket/nobody at 52LX8LP8BBWG6990) State NEW f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] switch_core_state_machine.c:584 (sofia/websocket/nobody at 52LX8LP8BBWG6990) Running State Change CS_INIT (Cur 1 Tot 1) f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] switch_core_state_machine.c:627 (sofia/websocket/nobody at 52LX8LP8BBWG6990) State INIT f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] mod_sofia.c:93 sofia/websocket/nobody at 52LX8LP8BBWG6990 SOFIA INIT f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] switch_core_state_machine.c:40 sofia/websocket/nobody at 52LX8LP8BBWG6990 Standard INIT f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] switch_core_state_machine.c:48 (sofia/websocket/nobody at 52LX8LP8BBWG6990) State Change CS_INIT -> CS_ROUTING f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] switch_core_state_machine.c:627 (sofia/websocket/nobody at 52LX8LP8BBWG6990) State INIT going to sleep f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] switch_core_state_machine.c:584 (sofia/websocket/nobody at 52LX8LP8BBWG6990) Running State Change CS_ROUTING (Cur 1 Tot 1) f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] switch_channel.c:2249 (sofia/websocket/nobody at 52LX8LP8BBWG6990) Callstate Change DOWN -> RINGING f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] switch_core_state_machine.c:643 (sofia/websocket/nobody at 52LX8LP8BBWG6990) State ROUTING f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] mod_sofia.c:154 sofia/websocket/nobody at 52LX8LP8BBWG6990 SOFIA ROUTING f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] switch_core_state_machine.c:236 sofia/websocket/nobody at 52LX8LP8BBWG6990 Standard ROUTING f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [INFO] mod_dialplan_xml.c:637 Processing unknown <>->52LX8LP8BBWG6990 in context wss-dialplan f9a97c98-373b-11e7-9136-499bb33ea37e Dialplan: sofia/websocket/nobody at 52LX8LP8BBWG6990 parsing [wss-dialplan->wss-dialplan] continue=false f9a97c98-373b-11e7-9136-499bb33ea37e Dialplan: sofia/websocket/nobody at 52LX8LP8BBWG6990 Regex (PASS) [wss-dialplan] destination_number(52LX8LP8BBWG6990) =~ /.+/ break=on-false f9a97c98-373b-11e7-9136-499bb33ea37e Dialplan: sofia/websocket/nobody at 52LX8LP8BBWG6990 Action conference(test123) f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] switch_core_state_machine.c:286 (sofia/websocket/nobody at 52LX8LP8BBWG6990) State Change CS_ROUTING -> CS_EXECUTE f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] switch_core_state_machine.c:643 (sofia/websocket/nobody at 52LX8LP8BBWG6990) State ROUTING going to sleep f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] switch_core_state_machine.c:584 (sofia/websocket/nobody at 52LX8LP8BBWG6990) Running State Change CS_EXECUTE (Cur 1 Tot 1) f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] switch_core_state_machine.c:650 (sofia/websocket/nobody at 52LX8LP8BBWG6990) State EXECUTE f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] mod_sofia.c:209 sofia/websocket/nobody at 52LX8LP8BBWG6990 SOFIA EXECUTE f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] switch_core_state_machine.c:328 sofia/websocket/nobody at 52LX8LP8BBWG6990 Standard EXECUTE f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] switch_core_session.c:2707 Application conference Requires media! pre_answering channel sofia/websocket/nobody at 52LX8LP8BBWG6990 f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [INFO] switch_core_session.c:2709 Sending early media f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] switch_core_media.c:8144 AUDIO RTP [sofia/websocket/nobody at 52LX8LP8BBWG6990] 172.16.19.215 port 27662 -> 192.168.56.151 port 42504 codec: 0 ms: 20 f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] switch_rtp.c:4096 Starting timer [soft] 160 bytes per 20ms f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] switch_core_media.c:8447 sofia/websocket/nobody at 52LX8LP8BBWG6990 Set 2833 dtmf send payload to 101 f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] switch_core_media.c:8454 sofia/websocket/nobody at 52LX8LP8BBWG6990 Set 2833 dtmf receive payload to 101 f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] switch_core_media.c:8477 sofia/websocket/nobody at 52LX8LP8BBWG6990 Set rtp dtmf delay to 40 f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] mod_sofia.c:2512 Ring SDP: f9a97c98-373b-11e7-9136-499bb33ea37e v=0 f9a97c98-373b-11e7-9136-499bb33ea37e o=FreeSWITCH 1494583976 1494583977 IN IP4 172.16.19.215 f9a97c98-373b-11e7-9136-499bb33ea37e s=FreeSWITCH f9a97c98-373b-11e7-9136-499bb33ea37e c=IN IP4 172.16.19.215 f9a97c98-373b-11e7-9136-499bb33ea37e t=0 0 f9a97c98-373b-11e7-9136-499bb33ea37e m=audio 27662 RTP/AVP 0 101 f9a97c98-373b-11e7-9136-499bb33ea37e a=rtpmap:0 PCMU/8000 f9a97c98-373b-11e7-9136-499bb33ea37e a=rtpmap:101 telephone-event/8000 f9a97c98-373b-11e7-9136-499bb33ea37e a=fmtp:101 0-16 f9a97c98-373b-11e7-9136-499bb33ea37e a=ptime:20 f9a97c98-373b-11e7-9136-499bb33ea37e a=sendrecv f9a97c98-373b-11e7-9136-499bb33ea37e f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [NOTICE] mod_sofia.c:2515 Pre-Answer sofia/websocket/nobody at 52LX8LP8BBWG6990! f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] switch_channel.c:3481 (sofia/websocket/nobody at 52LX8LP8BBWG6990) Callstate Change RINGING -> EARLY f9a97c98-373b-11e7-9136-499bb33ea37e EXECUTE sofia/websocket/nobody at 52LX8LP8BBWG6990 conference(test123) f9a97c98-373b-11e7-9136-499bb33ea37e v=0 f9a97c98-373b-11e7-9136-499bb33ea37e o=FreeSWITCH 1494583976 1494583978 IN IP4 172.16.19.215 f9a97c98-373b-11e7-9136-499bb33ea37e s=FreeSWITCH f9a97c98-373b-11e7-9136-499bb33ea37e c=IN IP4 172.16.19.215 f9a97c98-373b-11e7-9136-499bb33ea37e t=0 0 f9a97c98-373b-11e7-9136-499bb33ea37e m=audio 27662 RTP/AVP 0 101 f9a97c98-373b-11e7-9136-499bb33ea37e a=rtpmap:0 PCMU/8000 f9a97c98-373b-11e7-9136-499bb33ea37e a=rtpmap:101 telephone-event/8000 f9a97c98-373b-11e7-9136-499bb33ea37e a=fmtp:101 0-16 f9a97c98-373b-11e7-9136-499bb33ea37e a=ptime:20 f9a97c98-373b-11e7-9136-499bb33ea37e a=sendrecv f9a97c98-373b-11e7-9136-499bb33ea37e f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] sofia.c:7247 Channel sofia/websocket/nobody at 52LX8LP8BBWG6990 entering state [early][183] f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [NOTICE] mod_conference.c:1829 Channel [sofia/websocket/nobody at 52LX8LP8BBWG6990] has been answered f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] switch_channel.c:3780 (sofia/websocket/nobody at 52LX8LP8BBWG6990) Callstate Change EARLY -> ACTIVE f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] sofia.c:7247 Channel sofia/websocket/nobody at 52LX8LP8BBWG6990 entering state [completed][200] 2017-05-12 17:53:58.932851 [DEBUG] mod_conference.c:3289 using channel sound prefix: /tmp/fs1.8/share/freeswitch/sounds/en/us/callie f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] conference_member.c:1715 Raw Codec Activation Success L16 at 8000hz 1 channel 20ms f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] conference_member.c:1762 Raw Codec Activation Success L16 at 8000hz 1 channel 20ms f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] switch_core_codec.c:223 sofia/websocket/nobody at 52LX8LP8BBWG6990 Push codec L16:100 2017-05-12 17:53:58.932851 [DEBUG] mod_conference.c:227 Setup timer success interval: 20 samples: 160 2017-05-12 17:53:58.932851 [ERR] switch_core_video.c:2868 This function is not available, libpng not installed f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] conference_loop.c:1306 Setup timer soft success interval: 20 samples: 160 from codec PCMU 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [NOTICE] switch_channel.c:1104 New Channel sofia/websocket/nobody at 3PXT9NIJPMV6KC8W [02b30be2-373c-11e7-913a-499bb33ea37e] 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] switch_core_state_machine.c:584 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) Running State Change CS_NEW (Cur 2 Tot 2) 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] sofia.c:10028 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W receiving invite from 192.168.56.151:53568 version: 1.9.0 git db24869 2017-05-11 18:22:45Z 64bit 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] sofia.c:11325 Setting NAT mode based on websockets 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] sofia.c:7247 Channel sofia/websocket/nobody at 3PXT9NIJPMV6KC8W entering state [received][100] 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] sofia.c:7257 Remote SDP: 02b30be2-373c-11e7-913a-499bb33ea37e v=0 02b30be2-373c-11e7-913a-499bb33ea37e o=- 716582477 2 IN IP4 192.168.56.151 02b30be2-373c-11e7-913a-499bb33ea37e s=- 02b30be2-373c-11e7-913a-499bb33ea37e c=IN IP4 192.168.56.151 02b30be2-373c-11e7-913a-499bb33ea37e t=0 0 02b30be2-373c-11e7-913a-499bb33ea37e m=audio 11866 RTP/AVP 0 100 02b30be2-373c-11e7-913a-499bb33ea37e a=rtpmap:0 PCMU/8000 02b30be2-373c-11e7-913a-499bb33ea37e a=maxptime:20 02b30be2-373c-11e7-913a-499bb33ea37e 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] switch_core_media.c:5110 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] switch_core_media.c:5165 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] switch_core_media.c:5110 Audio Codec Compare [:100:0:20:0:1]/[PCMU:0:8000:20:64000:1] 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] switch_core_media.c:3425 Set Codec sofia/websocket/nobody at 3PXT9NIJPMV6KC8W PCMU/8000 20 ms 160 samples 64000 bits 1 channels 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] switch_core_codec.c:111 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W Original read codec set to PCMU:0 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] switch_core_media.c:5398 No 2833 in SDP. Liberal DTMF mode adding 101 as telephone-event. 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] switch_core_media.c:5427 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W Set 2833 dtmf send payload to 101 recv payload to 101 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] sofia.c:7670 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) State Change CS_NEW -> CS_INIT 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] switch_core_state_machine.c:603 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) State NEW 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] switch_core_state_machine.c:584 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) Running State Change CS_INIT (Cur 2 Tot 2) 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] switch_core_state_machine.c:627 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) State INIT 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] mod_sofia.c:93 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W SOFIA INIT 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] switch_core_state_machine.c:40 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W Standard INIT 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] switch_core_state_machine.c:48 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) State Change CS_INIT -> CS_ROUTING 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] switch_core_state_machine.c:627 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) State INIT going to sleep 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] switch_core_state_machine.c:584 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) Running State Change CS_ROUTING (Cur 2 Tot 2) 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] switch_channel.c:2249 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) Callstate Change DOWN -> RINGING 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] switch_core_state_machine.c:643 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) State ROUTING 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] mod_sofia.c:154 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W SOFIA ROUTING 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] switch_core_state_machine.c:236 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W Standard ROUTING 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [INFO] mod_dialplan_xml.c:637 Processing unknown <>->3PXT9NIJPMV6KC8W in context wss-dialplan 02b30be2-373c-11e7-913a-499bb33ea37e Dialplan: sofia/websocket/nobody at 3PXT9NIJPMV6KC8W parsing [wss-dialplan->wss-dialplan] continue=false 02b30be2-373c-11e7-913a-499bb33ea37e Dialplan: sofia/websocket/nobody at 3PXT9NIJPMV6KC8W Regex (PASS) [wss-dialplan] destination_number(3PXT9NIJPMV6KC8W) =~ /.+/ break=on-false 02b30be2-373c-11e7-913a-499bb33ea37e Dialplan: sofia/websocket/nobody at 3PXT9NIJPMV6KC8W Action conference(test123) 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] switch_core_state_machine.c:286 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) State Change CS_ROUTING -> CS_EXECUTE 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] switch_core_state_machine.c:643 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) State ROUTING going to sleep 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] switch_core_state_machine.c:584 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) Running State Change CS_EXECUTE (Cur 2 Tot 2) 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] switch_core_state_machine.c:650 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) State EXECUTE 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] mod_sofia.c:209 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W SOFIA EXECUTE 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] switch_core_state_machine.c:328 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W Standard EXECUTE 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] switch_core_session.c:2707 Application conference Requires media! pre_answering channel sofia/websocket/nobody at 3PXT9NIJPMV6KC8W 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [INFO] switch_core_session.c:2709 Sending early media 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] switch_core_media.c:8144 AUDIO RTP [sofia/websocket/nobody at 3PXT9NIJPMV6KC8W] 172.16.19.215 port 28858 -> 192.168.56.151 port 11866 codec: 0 ms: 20 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] switch_rtp.c:4096 Starting timer [soft] 160 bytes per 20ms 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] switch_core_media.c:8447 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W Set 2833 dtmf send payload to 101 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] switch_core_media.c:8454 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W Set 2833 dtmf receive payload to 101 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] switch_core_media.c:8477 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W Set rtp dtmf delay to 40 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] mod_sofia.c:2512 Ring SDP: 02b30be2-373c-11e7-913a-499bb33ea37e v=0 02b30be2-373c-11e7-913a-499bb33ea37e o=FreeSWITCH 1494582796 1494582797 IN IP4 172.16.19.215 02b30be2-373c-11e7-913a-499bb33ea37e s=FreeSWITCH 02b30be2-373c-11e7-913a-499bb33ea37e c=IN IP4 172.16.19.215 02b30be2-373c-11e7-913a-499bb33ea37e t=0 0 02b30be2-373c-11e7-913a-499bb33ea37e m=audio 28858 RTP/AVP 0 101 02b30be2-373c-11e7-913a-499bb33ea37e a=rtpmap:0 PCMU/8000 02b30be2-373c-11e7-913a-499bb33ea37e a=rtpmap:101 telephone-event/8000 02b30be2-373c-11e7-913a-499bb33ea37e a=fmtp:101 0-16 02b30be2-373c-11e7-913a-499bb33ea37e a=ptime:20 02b30be2-373c-11e7-913a-499bb33ea37e a=sendrecv 02b30be2-373c-11e7-913a-499bb33ea37e 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [NOTICE] mod_sofia.c:2515 Pre-Answer sofia/websocket/nobody at 3PXT9NIJPMV6KC8W! 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] switch_channel.c:3481 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) Callstate Change RINGING -> EARLY 02b30be2-373c-11e7-913a-499bb33ea37e EXECUTE sofia/websocket/nobody at 3PXT9NIJPMV6KC8W conference(test123) 02b30be2-373c-11e7-913a-499bb33ea37e v=0 02b30be2-373c-11e7-913a-499bb33ea37e o=FreeSWITCH 1494582796 1494582798 IN IP4 172.16.19.215 02b30be2-373c-11e7-913a-499bb33ea37e s=FreeSWITCH 02b30be2-373c-11e7-913a-499bb33ea37e c=IN IP4 172.16.19.215 02b30be2-373c-11e7-913a-499bb33ea37e t=0 0 02b30be2-373c-11e7-913a-499bb33ea37e m=audio 28858 RTP/AVP 0 101 02b30be2-373c-11e7-913a-499bb33ea37e a=rtpmap:0 PCMU/8000 02b30be2-373c-11e7-913a-499bb33ea37e a=rtpmap:101 telephone-event/8000 02b30be2-373c-11e7-913a-499bb33ea37e a=fmtp:101 0-16 02b30be2-373c-11e7-913a-499bb33ea37e a=ptime:20 02b30be2-373c-11e7-913a-499bb33ea37e a=sendrecv 02b30be2-373c-11e7-913a-499bb33ea37e 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] sofia.c:7247 Channel sofia/websocket/nobody at 3PXT9NIJPMV6KC8W entering state [early][183] 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [NOTICE] mod_conference.c:1829 Channel [sofia/websocket/nobody at 3PXT9NIJPMV6KC8W] has been answered 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] switch_channel.c:3780 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) Callstate Change EARLY -> ACTIVE 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] sofia.c:7247 Channel sofia/websocket/nobody at 3PXT9NIJPMV6KC8W entering state [completed][200] 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] conference_member.c:1715 Raw Codec Activation Success L16 at 8000hz 1 channel 20ms 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] conference_member.c:1762 Raw Codec Activation Success L16 at 8000hz 1 channel 20ms 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] switch_core_codec.c:223 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W Push codec L16:100 2017-05-12 17:54:14.092839 [ERR] switch_core_video.c:2868 This function is not available, libpng not installed 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] conference_loop.c:1306 Setup timer soft success interval: 20 samples: 160 from codec PCMU This is where session 02b30be2-373c-11e7-913a-499bb33ea37e wss socket gets an ssl errror: f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.952843 [DEBUG] sofia.c:7247 Channel sofia/websocket/nobody at 52LX8LP8BBWG6990 entering state [terminating][0] f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.952843 [NOTICE] sofia.c:8438 Hangup sofia/websocket/nobody at 52LX8LP8BBWG6990 [CS_EXECUTE] [NORMAL_UNSPECIFIED] f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [INFO] conference_loop.c:1621 Channel leaving conference, cause: NORMAL_UNSPECIFIED f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] mod_conference.c:2404 sofia/websocket/nobody at 52LX8LP8BBWG6990 skip receive message [TRANSFER] (channel is hungup already) f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] switch_core_media.c:11838 sofia/websocket/nobody at 52LX8LP8BBWG6990 skip receive message [BITRATE_REQ] (channel is hungup already) f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] switch_core_codec.c:248 sofia/websocket/nobody at 52LX8LP8BBWG6990 Restore previous codec PCMU:0. f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] switch_core_session.c:2884 sofia/websocket/nobody at 52LX8LP8BBWG6990 skip receive message [PHONE_EVENT] (channel is hungup already) f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] switch_core_state_machine.c:650 (sofia/websocket/nobody at 52LX8LP8BBWG6990) State EXECUTE going to sleep f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] switch_core_state_machine.c:584 (sofia/websocket/nobody at 52LX8LP8BBWG6990) Running State Change CS_HANGUP (Cur 2 Tot 2) f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] switch_core_state_machine.c:850 (sofia/websocket/nobody at 52LX8LP8BBWG6990) Callstate Change ACTIVE -> HANGUP f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] switch_core_state_machine.c:852 (sofia/websocket/nobody at 52LX8LP8BBWG6990) State HANGUP f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] mod_sofia.c:449 Channel sofia/websocket/nobody at 52LX8LP8BBWG6990 hanging up, cause: NORMAL_UNSPECIFIED f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] switch_core_state_machine.c:60 sofia/websocket/nobody at 52LX8LP8BBWG6990 Standard HANGUP, cause: NORMAL_UNSPECIFIED f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] switch_core_state_machine.c:852 (sofia/websocket/nobody at 52LX8LP8BBWG6990) State HANGUP going to sleep f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] switch_core_state_machine.c:619 (sofia/websocket/nobody at 52LX8LP8BBWG6990) State Change CS_HANGUP -> CS_REPORTING f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] switch_core_state_machine.c:584 (sofia/websocket/nobody at 52LX8LP8BBWG6990) Running State Change CS_REPORTING (Cur 2 Tot 2) f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] switch_core_state_machine.c:938 (sofia/websocket/nobody at 52LX8LP8BBWG6990) State REPORTING f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] switch_core_state_machine.c:174 sofia/websocket/nobody at 52LX8LP8BBWG6990 Standard REPORTING, cause: NORMAL_UNSPECIFIED f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] switch_core_state_machine.c:938 (sofia/websocket/nobody at 52LX8LP8BBWG6990) State REPORTING going to sleep f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] switch_core_state_machine.c:610 (sofia/websocket/nobody at 52LX8LP8BBWG6990) State Change CS_REPORTING -> CS_DESTROY f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] switch_core_session.c:1712 Session 1 (sofia/websocket/nobody at 52LX8LP8BBWG6990) Locked, Waiting on external entities f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [NOTICE] switch_core_session.c:1730 Session 1 (sofia/websocket/nobody at 52LX8LP8BBWG6990) Ended f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [NOTICE] switch_core_session.c:1734 Close Channel sofia/websocket/nobody at 52LX8LP8BBWG6990 [CS_DESTROY] f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] switch_core_state_machine.c:741 (sofia/websocket/nobody at 52LX8LP8BBWG6990) Running State Change CS_DESTROY (Cur 1 Tot 2) f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] switch_core_state_machine.c:751 (sofia/websocket/nobody at 52LX8LP8BBWG6990) State DESTROY f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] mod_sofia.c:354 sofia/websocket/nobody at 52LX8LP8BBWG6990 SOFIA DESTROY f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] switch_core_state_machine.c:181 sofia/websocket/nobody at 52LX8LP8BBWG6990 Standard DESTROY f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] switch_core_state_machine.c:751 (sofia/websocket/nobody at 52LX8LP8BBWG6990) State DESTROY going to sleep 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.112842 [DEBUG] sofia.c:7247 Channel sofia/websocket/nobody at 3PXT9NIJPMV6KC8W entering state [terminating][0] 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.112842 [NOTICE] sofia.c:8438 Hangup sofia/websocket/nobody at 3PXT9NIJPMV6KC8W [CS_EXECUTE] [NORMAL_UNSPECIFIED] 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [INFO] conference_loop.c:1621 Channel leaving conference, cause: NORMAL_UNSPECIFIED 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] mod_conference.c:2404 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W skip receive message [TRANSFER] (channel is hungup already) 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] switch_core_media.c:11838 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W skip receive message [BITRATE_REQ] (channel is hungup already) 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] switch_core_codec.c:248 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W Restore previous codec PCMU:0. 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] switch_core_session.c:2884 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W skip receive message [PHONE_EVENT] (channel is hungup already) 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] switch_core_state_machine.c:650 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) State EXECUTE going to sleep 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] switch_core_state_machine.c:584 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) Running State Change CS_HANGUP (Cur 1 Tot 2) 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] switch_core_state_machine.c:850 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) Callstate Change ACTIVE -> HANGUP 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] switch_core_state_machine.c:852 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) State HANGUP 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] mod_sofia.c:449 Channel sofia/websocket/nobody at 3PXT9NIJPMV6KC8W hanging up, cause: NORMAL_UNSPECIFIED 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] switch_core_state_machine.c:60 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W Standard HANGUP, cause: NORMAL_UNSPECIFIED 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] switch_core_state_machine.c:852 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) State HANGUP going to sleep 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] switch_core_state_machine.c:619 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) State Change CS_HANGUP -> CS_REPORTING 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] switch_core_state_machine.c:584 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) Running State Change CS_REPORTING (Cur 1 Tot 2) 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] switch_core_state_machine.c:938 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) State REPORTING 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] switch_core_state_machine.c:174 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W Standard REPORTING, cause: NORMAL_UNSPECIFIED 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] switch_core_state_machine.c:938 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) State REPORTING going to sleep 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] switch_core_state_machine.c:610 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) State Change CS_REPORTING -> CS_DESTROY 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] switch_core_session.c:1712 Session 2 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) Locked, Waiting on external entities 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [NOTICE] switch_core_session.c:1730 Session 2 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) Ended 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [NOTICE] switch_core_session.c:1734 Close Channel sofia/websocket/nobody at 3PXT9NIJPMV6KC8W [CS_DESTROY] On Fri, May 12, 2017 at 11:51 AM, Luke Wahlmeier wrote: > Thanks Michael, > > I am more then happy to setup something with libks if needed. > > I have figured out some more however. It appears that this only happens > when a wss connections session has not fully established and is cleaning up > because of timing out. The problem is that it causes another wss > connection it to get this ssl error, even if that other wss connection has > a fully established and running audio session. It is important to note it > does not seem to interrupt audio just the wss sip channel, which I am > fairly sure can be reestablished for that audio session w/o an issue. > > The sessions that is being cleaned up sends the logs messages as its doing > it: > 2017-05-12 17:32:46.607768 [NOTICE] sofia.c:8438 Hangup > sofia/websocket/nobody at 1LF3F6I924P9WH6U [CS_EXECUTE] [NORMAL_UNSPECIFIED] > 2017-05-12 17:32:46.627768 [INFO] conference_loop.c:1621 Channel leaving > conference, cause: NORMAL_UNSPECIFIED > 2017-05-12 17:32:46.627768 [NOTICE] switch_core_session.c:1730 Session 46 > (sofia/websocket/nobody at 1LF3F6I924P9WH6U) Ended > 2017-05-12 17:32:46.627768 [NOTICE] switch_core_session.c:1734 Close > Channel sofia/websocket/nobody at 1LF3F6I924P9WH6U [CS_DESTROY] > > I have attached the updated python script, it can duplicate this every > time now with only 2 connections. I verified with a webRTC client that if > I initiate this first connection in the script, let it close, then connect > the webRTC client and get full audio, once the first session from the > script times out it causes the webRTC wss connection to get an error and > close. > > The webRTC connection is in chrome with sip.js. > > Sorry the python script is so nasty, was working through any possible > duplicated sip session stuff in it to make sure that was not why it was > hitting the second connection. > > > On Fri, May 12, 2017 at 10:20 AM, Michael Jerris wrote: > >> test on master.. work a similar test for verto maybe, this might have to >> do with sip specifically trying to keep state. Might make sense to build >> something out of libks as it has basically the same web socket code, and >> has both client and server web socket support in it, to do a ?real? test?, >> instead of this fake sip without any state over web sockets. >> >> >> On May 12, 2017, at 11:42 AM, Luke Wahlmeier >> wrote: >> >> Just got done testing this on v1.6 head and master, both seem to still >> have this issue. This box is using libssl version 1.0.1t-1+deb8u6. I am >> gonna start digging more into the ws/wss/sofia code to see if I can figure >> it out. Any suggestions on debugging this would be appreciated. >> >> Thanks >> Luke >> >> On Thu, May 11, 2017 at 5:12 PM, Luke Wahlmeier >> wrote: >> >>> Its just in our isolated lab, pretty normal dell xeon server running >>> Jessie 8.6. I just want to get it building on the same box I am testing >>> with so setting that all up. >>> >>> I was able to reproduce it w/o DTLS/Srtp. here is a much simpler and >>> cleaned up version of the python script. >>> >>> >>> >>> On Thu, May 11, 2017 at 4:34 PM, Michael Jerris wrote: >>> >>>> what is ?this environment? ? >>>> >>>> On May 11, 2017, at 6:31 PM, Luke Wahlmeier >>>> wrote: >>>> >>>> Yeah I can usually get it to happen within about 5 minutes or so of >>>> testing. Still getting all setup to build freeswitch in this environment, >>>> but I should have it working by tomorrow. I will try more w/o dtls/srtp as >>>> well and make sure it does not need to be on. >>>> >>>> Thanks >>>> Luke >>>> >>>> On Thu, May 11, 2017 at 4:20 PM, Michael Jerris >>>> wrote: >>>> >>>>> if you can reproduce this reliably, i?d try master as well. Unless >>>>> this is a bug in openssl, i can?t imagine how dtls would come into play in >>>>> something like this. >>>>> >>>>> > On May 11, 2017, at 5:48 PM, Luke Wahlmeier >>>>> wrote: >>>>> > >>>>> > I keep semi-regularly running into issues using the wss transport >>>>> when using dtls/strp/ice. This is on the latest 1.6.17~34~0fc0946 on >>>>> Debian jessie, but I am pretty sure it was happening on the last couple >>>>> releases as well. >>>>> > >>>>> > It seems like something bad/wrong happens to the encrypted data >>>>> going over the websocket coming from freeswitch when more then 1 websocket >>>>> connection are going and so far ice/srtp/dtls also seem to be needed in the >>>>> invite to duplicate it. >>>>> > >>>>> > I have tried many different languages and network/ssl stacks and >>>>> keep running into this. It is always on data coming in from freeswitch on >>>>> the websocket connection, and its very very random. Sometimes I will get >>>>> it 20 times in a row, other times it takes thousands of >>>>> connections/sessions before it happen. It also, obviously, completely goes >>>>> away if I use plain ws instead wss. >>>>> > >>>>> > Here are the errors: >>>>> > python: >>>>> > SSLError: [SSL: DECRYPTION_FAILED_OR_BAD_RECORD_MAC] decryption >>>>> failed or bad record mac (_ssl.c:1750) >>>>> > c/c++ (stunnel4): >>>>> > SSL_read: 1408F119: error:1408F119:SSL routines:SSL3_GET_RECORD:decryption >>>>> failed or bad record mac >>>>> > Java: >>>>> > java.lang.IllegalArgumentException: Bad arguments >>>>> > at javax.crypto.Mac.update(Mac.java:509) >>>>> > at sun.security.ssl.MAC.compute(MAC.java:135) >>>>> > at sun.security.ssl.InputRecord.checkMacTags(InputRecord.java:2 >>>>> 65) >>>>> > at sun.security.ssl.InputRecord.decrypt(InputRecord.java:216) >>>>> > at sun.security.ssl.EngineInputRecord.decrypt(EngineInputRecord >>>>> .java:177) >>>>> > at sun.security.ssl.SSLEngineImpl.readRecord(SSLEngineImpl.java >>>>> :974) >>>>> > at sun.security.ssl.SSLEngineImpl.readNetRecord(SSLEngineImpl.j >>>>> ava:907) >>>>> > at sun.security.ssl.SSLEngineImpl.unwrap(SSLEngineImpl.java:781) >>>>> > at javax.net.ssl.SSLEngine.unwrap(SSLEngine.java:624) >>>>> > >>>>> > Attached are a simple python script to do the load, my dialplan and >>>>> sip_profile. The python script can take a few runs before it see the >>>>> error, and I know its not completing the sip or rtp, but even if it does >>>>> this still happens. >>>>> > >>>>> > I have also looked at libsofia-sip-ua/tport/ws.c and I dont see >>>>> anything obvious. I am getting setup to build v1.6 head and test this any >>>>> guidance on ways I can trouble shoot this better or requests for more info >>>>> are very welcome. >>>>> > >>>>> >>>> >>>> >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-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 >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170512/5d2ff5dc/attachment-0001.html From dig1234 at gmail.com Fri May 12 22:37:21 2017 From: dig1234 at gmail.com (Daniel Greenwald) Date: Fri, 12 May 2017 14:37:21 -0400 Subject: [Freeswitch-users] [OpenSIPS-Users] TLS SIP packet tracing and visualization In-Reply-To: References: Message-ID: Giovanni- I personally experience this issue on three different systems. I've tried all hep versions, and confirmed not a homer issue via wireshark. ie the HEP is not being sent for Inbound TLS messages. I did report it to Jira in Oct 2016 but got little attention. I still find HEP useful for outbound messages but it would be really nice to have two way SIP conversation in cleartext... https://freeswitch.org/jira/browse/FS-9657 On Thu, May 11, 2017 at 11:45 AM, Tamas Jalsovszky wrote: > Does HEP send verto signaling info to Homer too? > > On 10 May 2017 at 21:38, Daniel Greenwald wrote: > >> Thanks for this script! >> Theoretically it is possible to see TLS SIP traffic with freeswitch >> sending HEP to Homer. But there seems to be a bug in FS that only sends one >> side of SIP conversation (ie the FS side, not inbound messages).. >> >> On Tue, May 9, 2017 at 11:10 AM, Giovanni Maruzzelli >> wrote: >> >>> On 9 May 2017 at 15:18, Bogdan-Andrei Iancu wrote: >>> >>>> Thank you Giovanni, that is a useful tool - we will document it in the >>>> OpenSIPS TLS tutorial, so other can benefit ;) >>>> >>>> >>> >>> Glad about it! >>> Be sure to get it from https://freeswitch.org/conflue >>> nce/display/FREESWITCH/Packet+Capture#PacketCapture-TLSwithsharka , is >>> the latest version with a couple fixes. >>> >>> -giovanni >>> >>> >>> >>> >>>> Many thanks, >>>> >>>> Bogdan-Andrei Iancu >>>> OpenSIPS Founder and Developer >>>> http://www.opensips-solutions.com >>>> >>>> OpenSIPS Summit May 2017 Amsterdam >>>> http://www.opensips.org/events/Summit-2017Amsterdam.html >>>> >>>> On 05/02/2017 05:52 PM, Giovanni Maruzzelli wrote: >>>> >>>> For a cut and paste ready version, that has the correct carriage >>>> returns (mangled by mail), check it in FreeSWITCH documentation: >>>> >>>> https://freeswitch.org/confluence/display/FREESWITCH/Packet+ >>>> Capture#PacketCapture-TLSwithsharka >>>> >>>> -giovanni >>>> >>>> On 2 May 2017 at 16:26, Giovanni Maruzzelli wrote: >>>> >>>>> Hello fellows, >>>>> >>>>> after some experimentation with various tools, I come out with a >>>>> little shell tool that maybe can be useful to you too. >>>>> >>>>> It can only work with non-forward secrecy ciphers, obviously, and only >>>>> if is started before the client do the initial TLS handshake (eg, just >>>>> restart the client). Forward secrecy cannot be decrypted after fact, so >>>>> don't waste effort. >>>>> >>>>> An example of ciphers that can be decrypted are the "AES256-SHA" >>>>> openssl cipher group. You can use ssldump to check what cipher is used by >>>>> serverhello. >>>>> >>>>> Enjoy, make it better, and share it :) >>>>> >>>>> >>>>> #!/bin/bash >>>>> # brought to you by Giovanni Maruzzelli >>>>> # >>>>> SERVERIP="192.168.1.150" >>>>> SERVERPORT="5061" >>>>> PRIVKEY="/etc/certs/privkey.pem" >>>>> STDERR2DEVNULL=" 2>/dev/null " >>>>> REGEX="notyet" >>>>> >>>>> if [ -z "$1" ]; then >>>>> REGEX="\\\.*" >>>>> else >>>>> REGEX="$1" >>>>> fi >>>>> FILTER="ssl.app_data and sip matches" >>>>> FILTER2="$FILTER \"$REGEX\"" >>>>> FILTER3="'$FILTER2'" >>>>> ARGUMENT="-i 1 -Y $FILTER3 -E header=y -T fields -e frame.number -e >>>>> frame.time -e frame.time_delta_displayed -e ip.src -e ip.dst -e >>>>> sip.Status-Line -e sip.Request-Line -e sip.msg_hdr -l -d >>>>> tcp.port\=\=5061,sip -o \"ssl.keys_list: $SERVERIP,$SERVERPORT,sip,$PRIVKEY\" >>>>> $STDERR2DEVNULL | sed -u 's/\t/\n/g' | sed -u '/^$/d' | sed -u >>>>> 's/^[0-9]*$/\n==&==============================/g'" >>>>> >>>>> echo "" >>>>> echo "NB: if it do not works, edit script so that STDERR2DEVNULL=\" \" >>>>> and try again" >>>>> echo "" >>>>> echo "NB: remember to quote and escape match patterns, using triple >>>>> slash" >>>>> echo " eg, for matching 1010 at pbx.example.com, use \" >>>>> 1010 at pbx.example.com\"" >>>>> echo " eg, for matching anything, use \"\\\\\\.*\"" >>>>> echo " eg, for matching *98, use \"\\\\\\*98\"" >>>>> echo "USAGE: $0 \"\\\\\\*98 at pbx.example.com\"" >>>>> echo "" >>>>> >>>>> >>>>> case "$1" in >>>>> -help|--help|?) >>>>> exit 0 >>>>> ;; >>>>> *) >>>>> echo "THIS TIME WE'RE DOING:" >>>>> echo "tshark $ARGUMENT" >>>>> echo "" >>>>> bash -c "tshark $ARGUMENT" >>>>> ;; >>>>> esac >>>>> >>>>> >>>>> >>>>> -- >>>>> >>>>> Sincerely, >>>>> >>>>> Giovanni Maruzzelli >>>>> OpenTelecom.IT >>>>> cell: +39 347 266 56 18 >>>>> >>>> >>>> >>>> >>>> -- >>>> >>>> Sincerely, >>>> >>>> Giovanni Maruzzelli >>>> OpenTelecom.IT >>>> cell: +39 347 266 56 18 >>>> >>>> >>>> _______________________________________________ >>>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>>> >>> >>> >>> -- >>> >>> Sincerely, >>> >>> Giovanni Maruzzelli >>> OpenTelecom.IT >>> cell: +39 347 266 56 18 >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170512/60becdaa/attachment.html From gmaruzz at gmail.com Fri May 12 22:49:45 2017 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Fri, 12 May 2017 20:49:45 +0200 Subject: [Freeswitch-users] [OpenSIPS-Users] TLS SIP packet tracing and visualization In-Reply-To: References: Message-ID: Hello Daniel, I would suggest this: test it with latest master git code, add to your jira all relevant traces etc with latest (eg, today's) git master code. This will bump attention to it. -giovanni On 12 May 2017 at 20:37, Daniel Greenwald wrote: > Giovanni- I personally experience this issue on three different systems. > I've tried all hep versions, and confirmed not a homer issue via wireshark. > ie the HEP is not being sent for Inbound TLS messages. I did report it to > Jira in Oct 2016 but got little attention. I still find HEP useful for > outbound messages but it would be really nice to have two way SIP > conversation in cleartext... > > https://freeswitch.org/jira/browse/FS-9657 > > > > On Thu, May 11, 2017 at 11:45 AM, Tamas Jalsovszky > wrote: > >> Does HEP send verto signaling info to Homer too? >> >> On 10 May 2017 at 21:38, Daniel Greenwald wrote: >> >>> Thanks for this script! >>> Theoretically it is possible to see TLS SIP traffic with freeswitch >>> sending HEP to Homer. But there seems to be a bug in FS that only sends one >>> side of SIP conversation (ie the FS side, not inbound messages).. >>> >>> On Tue, May 9, 2017 at 11:10 AM, Giovanni Maruzzelli >>> wrote: >>> >>>> On 9 May 2017 at 15:18, Bogdan-Andrei Iancu >>>> wrote: >>>> >>>>> Thank you Giovanni, that is a useful tool - we will document it in the >>>>> OpenSIPS TLS tutorial, so other can benefit ;) >>>>> >>>>> >>>> >>>> Glad about it! >>>> Be sure to get it from https://freeswitch.org/conflue >>>> nce/display/FREESWITCH/Packet+Capture#PacketCapture-TLSwithsharka , is >>>> the latest version with a couple fixes. >>>> >>>> -giovanni >>>> >>>> >>>> >>>> >>>>> Many thanks, >>>>> >>>>> Bogdan-Andrei Iancu >>>>> OpenSIPS Founder and Developer >>>>> http://www.opensips-solutions.com >>>>> >>>>> OpenSIPS Summit May 2017 Amsterdam >>>>> http://www.opensips.org/events/Summit-2017Amsterdam.html >>>>> >>>>> On 05/02/2017 05:52 PM, Giovanni Maruzzelli wrote: >>>>> >>>>> For a cut and paste ready version, that has the correct carriage >>>>> returns (mangled by mail), check it in FreeSWITCH documentation: >>>>> >>>>> https://freeswitch.org/confluence/display/FREESWITCH/Packet+ >>>>> Capture#PacketCapture-TLSwithsharka >>>>> >>>>> -giovanni >>>>> >>>>> On 2 May 2017 at 16:26, Giovanni Maruzzelli wrote: >>>>> >>>>>> Hello fellows, >>>>>> >>>>>> after some experimentation with various tools, I come out with a >>>>>> little shell tool that maybe can be useful to you too. >>>>>> >>>>>> It can only work with non-forward secrecy ciphers, obviously, and >>>>>> only if is started before the client do the initial TLS handshake (eg, just >>>>>> restart the client). Forward secrecy cannot be decrypted after fact, so >>>>>> don't waste effort. >>>>>> >>>>>> An example of ciphers that can be decrypted are the "AES256-SHA" >>>>>> openssl cipher group. You can use ssldump to check what cipher is used by >>>>>> serverhello. >>>>>> >>>>>> Enjoy, make it better, and share it :) >>>>>> >>>>>> >>>>>> #!/bin/bash >>>>>> # brought to you by Giovanni Maruzzelli >>>>>> # >>>>>> SERVERIP="192.168.1.150" >>>>>> SERVERPORT="5061" >>>>>> PRIVKEY="/etc/certs/privkey.pem" >>>>>> STDERR2DEVNULL=" 2>/dev/null " >>>>>> REGEX="notyet" >>>>>> >>>>>> if [ -z "$1" ]; then >>>>>> REGEX="\\\.*" >>>>>> else >>>>>> REGEX="$1" >>>>>> fi >>>>>> FILTER="ssl.app_data and sip matches" >>>>>> FILTER2="$FILTER \"$REGEX\"" >>>>>> FILTER3="'$FILTER2'" >>>>>> ARGUMENT="-i 1 -Y $FILTER3 -E header=y -T fields -e frame.number -e >>>>>> frame.time -e frame.time_delta_displayed -e ip.src -e ip.dst -e >>>>>> sip.Status-Line -e sip.Request-Line -e sip.msg_hdr -l -d >>>>>> tcp.port\=\=5061,sip -o \"ssl.keys_list: $SERVERIP,$SERVERPORT,sip,$PRIVKEY\" >>>>>> $STDERR2DEVNULL | sed -u 's/\t/\n/g' | sed -u '/^$/d' | sed -u >>>>>> 's/^[0-9]*$/\n==&==============================/g'" >>>>>> >>>>>> echo "" >>>>>> echo "NB: if it do not works, edit script so that STDERR2DEVNULL=\" >>>>>> \" and try again" >>>>>> echo "" >>>>>> echo "NB: remember to quote and escape match patterns, using triple >>>>>> slash" >>>>>> echo " eg, for matching 1010 at pbx.example.com, use \" >>>>>> 1010 at pbx.example.com\"" >>>>>> echo " eg, for matching anything, use \"\\\\\\.*\"" >>>>>> echo " eg, for matching *98, use \"\\\\\\*98\"" >>>>>> echo "USAGE: $0 \"\\\\\\*98 at pbx.example.com\"" >>>>>> echo "" >>>>>> >>>>>> >>>>>> case "$1" in >>>>>> -help|--help|?) >>>>>> exit 0 >>>>>> ;; >>>>>> *) >>>>>> echo "THIS TIME WE'RE DOING:" >>>>>> echo "tshark $ARGUMENT" >>>>>> echo "" >>>>>> bash -c "tshark $ARGUMENT" >>>>>> ;; >>>>>> esac >>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> >>>>>> Sincerely, >>>>>> >>>>>> Giovanni Maruzzelli >>>>>> OpenTelecom.IT >>>>>> cell: +39 347 266 56 18 >>>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> >>>>> Sincerely, >>>>> >>>>> Giovanni Maruzzelli >>>>> OpenTelecom.IT >>>>> cell: +39 347 266 56 18 >>>>> >>>>> >>>>> _______________________________________________ >>>>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>> >>>>> >>>>> >>>> >>>> >>>> -- >>>> >>>> Sincerely, >>>> >>>> Giovanni Maruzzelli >>>> OpenTelecom.IT >>>> cell: +39 347 266 56 18 >>>> >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-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 > -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170512/024fda7c/attachment-0001.html From david.villasmil.work at gmail.com Sat May 13 19:00:28 2017 From: david.villasmil.work at gmail.com (David Villasmil) Date: Sat, 13 May 2017 17:00:28 +0200 Subject: [Freeswitch-users] SIPS scheme Message-ID: Hello guys. I've got this setup, where the client is sending as contact "sip:" coming from kamailio proxy in tls ip until kamailio. Kamailio forwards to FS in tcp, but freeswitch is answering with "sip:" and this breaks the client who answers "SIPS Required"... Is there a way of specifying the contact should be sips ? Has anyone encountered this and solved it? Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170513/65875d76/attachment.html From ashwinrath at gmail.com Sat May 13 21:14:52 2017 From: ashwinrath at gmail.com (Ashwin Rath) Date: Sat, 13 May 2017 22:44:52 +0530 Subject: [Freeswitch-users] Conditional call forward Message-ID: Hi I have an extension which has call forward setup BUT i would like the call forward to work only when dialed from a certain number and not from another numbers. Can this be achieved ? -- Ashwin Kumar Rath -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170513/481cbab9/attachment.html From brians at iptel.co Sun May 14 01:27:52 2017 From: brians at iptel.co (Brian :) Date: Sat, 13 May 2017 22:27:52 +0100 Subject: [Freeswitch-users] DTMF Message-ID: Hello List, We have a PITA carrier that we do 2833 with 99.8% of the time. Sometimes if the carrier they are sending the call to doesn't support 2833 when they will send an OK back to us the SDP won't have 2833 in the RTP Map. Is it possible to detect this in dialog and start dtmf generate so we will send tones inband for these calls? For inbound calls we detect it with something like on the initial invite. Just not sure how to do this for an OK in dialog or if its possible. Thanks! Brian From gmaruzz at gmail.com Sun May 14 02:58:44 2017 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Sun, 14 May 2017 00:58:44 +0200 Subject: [Freeswitch-users] SIPS scheme In-Reply-To: References: Message-ID: If kamailio is under your control, you can change contact back and forth from it (eg, before forwarding to freeswitch) sent from mobile cell: +39 347 266 56 18 Giovanni Maruzzelli OpenTelecom.IT On May 13, 2017 5:01 PM, "David Villasmil" wrote: > Hello guys. > > I've got this setup, where the client is sending as contact "sip:" coming > from kamailio proxy in tls ip until kamailio. Kamailio forwards to FS in > tcp, but freeswitch is answering with "sip:" and this breaks the client who > answers "SIPS Required"... > > Is there a way of specifying the contact should be sips ? > > Has anyone encountered this and solved it? > > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 <+34%20669%2044%2083%2037> > ? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170514/67110d2b/attachment.html From david.villasmil.work at gmail.com Sun May 14 03:09:07 2017 From: david.villasmil.work at gmail.com (David Villasmil) Date: Sun, 14 May 2017 01:09:07 +0200 Subject: [Freeswitch-users] SIPS scheme In-Reply-To: References: Message-ID: hello, thanks for replying. Yes, i thought about this, but asterisk has a patch to do this already without any changes on kamailoi, and I was looking more in the same direction. Regards, David ? Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 On Sun, May 14, 2017 at 12:58 AM, Giovanni Maruzzelli wrote: > If kamailio is under your control, you can change contact back and forth > from it (eg, before forwarding to freeswitch) > > sent from mobile > cell: +39 347 266 56 18 > Giovanni Maruzzelli > OpenTelecom.IT > > On May 13, 2017 5:01 PM, "David Villasmil" > wrote: > >> Hello guys. >> >> I've got this setup, where the client is sending as contact "sip:" coming >> from kamailio proxy in tls ip until kamailio. Kamailio forwards to FS in >> tcp, but freeswitch is answering with "sip:" and this breaks the client who >> answers "SIPS Required"... >> >> Is there a way of specifying the contact should be sips ? >> >> Has anyone encountered this and solved it? >> >> Regards, >> >> David Villasmil >> email: david.villasmil.work at gmail.com >> phone: +34669448337 <+34%20669%2044%2083%2037> >> ? >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170514/fb010435/attachment.html From s.safarov at gmail.com Sun May 14 07:20:01 2017 From: s.safarov at gmail.com (Sergey Safarov) Date: Sun, 14 May 2017 03:20:01 +0000 Subject: [Freeswitch-users] SIPS scheme In-Reply-To: References: Message-ID: FS not supports SIPS scheme. Exist jira ticket about this. ??, 14 ??? 2017, 2:10 David Villasmil : > hello, thanks for replying. > > Yes, i thought about this, but asterisk has a patch to do this already > without any changes on kamailoi, and I was looking more in the same > direction. > > Regards, > > David > ? > > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 > > On Sun, May 14, 2017 at 12:58 AM, Giovanni Maruzzelli > wrote: > >> If kamailio is under your control, you can change contact back and forth >> from it (eg, before forwarding to freeswitch) >> >> sent from mobile >> cell: +39 347 266 56 18 >> Giovanni Maruzzelli >> OpenTelecom.IT >> >> On May 13, 2017 5:01 PM, "David Villasmil" < >> david.villasmil.work at gmail.com> wrote: >> >>> Hello guys. >>> >>> I've got this setup, where the client is sending as contact "sip:" >>> coming from kamailio proxy in tls ip until kamailio. Kamailio forwards to >>> FS in tcp, but freeswitch is answering with "sip:" and this breaks the >>> client who answers "SIPS Required"... >>> >>> Is there a way of specifying the contact should be sips ? >>> >>> Has anyone encountered this and solved it? >>> >>> Regards, >>> >>> David Villasmil >>> email: david.villasmil.work at gmail.com >>> phone: +34669448337 <+34%20669%2044%2083%2037> >>> ? >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170514/e551cb08/attachment-0001.html From david.villasmil.work at gmail.com Mon May 15 03:35:36 2017 From: david.villasmil.work at gmail.com (David Villasmil) Date: Mon, 15 May 2017 01:35:36 +0200 Subject: [Freeswitch-users] SIPS scheme In-Reply-To: References: Message-ID: I fixed it with kamilio like this: if(is_method("INVITE") && status=~"200") { if( subst('/sip:/sips:/ig') ) { xlog("L_ERR","[ONREPLY]: Changed contact [$ct]!\n"); } } On the onreply_route Thanks David ? Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 On Sun, May 14, 2017 at 5:20 AM, Sergey Safarov wrote: > FS not supports SIPS scheme. Exist jira ticket about this. > > ??, 14 ??? 2017, 2:10 David Villasmil : > >> hello, thanks for replying. >> >> Yes, i thought about this, but asterisk has a patch to do this already >> without any changes on kamailoi, and I was looking more in the same >> direction. >> >> Regards, >> >> David >> ? >> >> Regards, >> >> David Villasmil >> email: david.villasmil.work at gmail.com >> phone: +34669448337 <+34%20669%2044%2083%2037> >> >> On Sun, May 14, 2017 at 12:58 AM, Giovanni Maruzzelli >> wrote: >> >>> If kamailio is under your control, you can change contact back and forth >>> from it (eg, before forwarding to freeswitch) >>> >>> sent from mobile >>> cell: +39 347 266 56 18 >>> Giovanni Maruzzelli >>> OpenTelecom.IT >>> >>> On May 13, 2017 5:01 PM, "David Villasmil" >> com> wrote: >>> >>>> Hello guys. >>>> >>>> I've got this setup, where the client is sending as contact "sip:" >>>> coming from kamailio proxy in tls ip until kamailio. Kamailio forwards to >>>> FS in tcp, but freeswitch is answering with "sip:" and this breaks the >>>> client who answers "SIPS Required"... >>>> >>>> Is there a way of specifying the contact should be sips ? >>>> >>>> Has anyone encountered this and solved it? >>>> >>>> Regards, >>>> >>>> David Villasmil >>>> email: david.villasmil.work at gmail.com >>>> phone: +34669448337 <+34%20669%2044%2083%2037> >>>> ? >>>> >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170515/9c7400df/attachment.html From lpopov at blasterphone.com Mon May 15 06:23:13 2017 From: lpopov at blasterphone.com (Lyubo Popov) Date: Sun, 14 May 2017 23:23:13 -0300 Subject: [Freeswitch-users] 300 Multiple Choices - how to force FS to replace From field In-Reply-To: References: Message-ID: Hello Brian, Sorry, did not see your response on May, 5th. Yes I have this in sofia profile. Is there anything else that I need to add or change in order to make FS to use the returned FROM value by the SIP REDIRECT server? Cheers, L.Popov On Fri, May 5, 2017 at 10:31 AM, Brian West wrote: > are you setting manual-redirect in your Sofia profile? > > > https://freeswitch.org/confluence/display/FREESWITCH/Handling+SIP+Redirect > > On Thu, May 4, 2017 at 10:20 AM, Lyubo Popov wrote: > >> Hello Brian, >> >> The authentication is DIGEST done via RADIUS. The Username is the same as >> the Caller ID...or maybe I understood your question wrong..? The number >> that shows in the from field is actually the SIP username created in the >> system ( for routing, billing, radius AAA, etc. ) and it is as well the >> callers number ( Caller ID). I use Raduis AAA to authenticate and account >> the calls and the user you see 551000 is actually a username of voip >> account created in the billing. That is why you see the incoming call from >> that username (551000). This is what FS is using in the FROM field. Since >> many of the accounts are created in format different from E164, often it is >> necessary to rewrite the account number ( the caller number ) to E164 or >> the termination will not accept the call. This is what I am trying to do >> now, set a rewrite rule in the billing system to convert 551000 to >> 1140031556 and this is what it is returned to FS as you can see in the >> packets the billing sends back.. >> >> Cheers, >> L.Popov >> >> >> On Wed, May 3, 2017 at 7:20 PM, Brian West wrote: >> >>> Why are you using the from field for authentication? >>> >>> On Wed, May 3, 2017 at 12:42 PM, Lyubo Popov >>> wrote: >>> >>>> Hello everyone, >>>> >>>> I would like to express my thanks in advance to anyone who may be able >>>> to help me with some insides. >>>> >>>> I am using a routing software with SIP Redirect to send routes to FS >>>> with 300 Multiple Choices and mod_xml_radius to authenticate the SIP users. >>>> In the Sip redirect server I am manipulating as well the FROM number and >>>> sending back to FS, but FS will not respect this and continue using the SIP >>>> account that sent the call in the first place in the FROM field. Here are >>>> some SIP packets from both sides to clarify the whole process. >>>> >>>> 1. Sending call to FS with Zoiper, destination 556230951662 >>>> >>>> INVITE sip:556230951662 at 216.x.x.x:5080;transport=UDP SIP/2.0 >>>> Via: SIP/2.0/UDP 177.x.x.x:1048;branch=z9hG4bK- >>>> d8754z-038f1c7d251308c2-1---d8754z-;rport >>>> Max-Forwards: 70 >>>> Contact: >>>> To: >>>> From: "551000";tag=25599d20 >>>> Call-ID: NmUzYTAwNmQ1NTZjMDM2ZjVhYTgzMDdiY2RiMmI0ZTc. >>>> CSeq: 1 INVITE >>>> Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, >>>> INFO, SUBSCRIBE >>>> Content-Type: application/sdp >>>> Supported: replaces, norefersub, extended-refer, timer, >>>> X-cisco-serviceuri >>>> User-Agent: Zoiper for Windows 2.43 r24984 >>>> Allow-Events: presence, kpml >>>> Content-Length: 232 >>>> >>>> v=0 >>>> o=Zoiper_user 0 0 IN IP4 177.x.x.x >>>> s=Zoiper_session >>>> c=IN IP4 177.x.x.x >>>> t=0 0 >>>> m=audio 8000 RTP/AVP 8 0 101 >>>> a=rtpmap:8 PCMA/8000 >>>> a=rtpmap:0 PCMU/8000 >>>> a=rtpmap:101 telephone-event/8000 >>>> a=fmtp:101 0-15 >>>> a=sendrecv >>>> >>>> >>>> 2. FS sending INVITE to SIP Redirect server >>>> >>>> INVITE sip:556230951662 at 69.x.x.x:5060 SIP/2.0 >>>> Via: SIP/2.0/UDP 216.245.218.230;rport;branch=z9hG4bKateZg87rDBpZa >>>> Max-Forwards: 69 >>>> From: "551000" ;tag=FeNXS71300N0c >>>> To: >>>> Call-ID: e70ee145-aac6-1235-79ba-002590a0ec9b >>>> CSeq: 106579790 INVITE >>>> Contact: >>>> User-Agent: FreeSWITCH >>>> 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: 397 >>>> X-FS-Support: update_display,send_info >>>> Remote-Party-ID: "551000" ;party=c >>>> alling;screen=yes;privacy=off >>>> >>>> v=0 >>>> o=FreeSWITCH 1493809233 1493809234 IN IP4 216.x.x.x >>>> 2017-05-03 12:15:09.816119 [ERR] mod_xml_radius.c:911 Didn't match: >>>> 69.x.x.x:5060 == ^69\.x\.x\.x >>>> s=FreeSWITCH >>>> c=IN IP4 216.x.x.x >>>> t=0 0 >>>> m=audio 22476 RTP/AVP 8 0 18 101 13 >>>> a=rtpmap:8 PCMA/8000 >>>> a=rtpmap:0 PCMU/8000 >>>> a=rtpmap:18 G729/8000 >>>> a=rtpmap:101 telephone-event/8000 >>>> a=fmtp:101 0-16 >>>> a=ptime:20 >>>> m=audio 22476 RTP/AVP 4 101 13 >>>> a=rtpmap:4 G723/8000 >>>> a=rtpmap:101 telephone-event/8000 >>>> a=fmtp:101 0-16 >>>> a=ptime:30 >>>> >>>> >>>> 3. SIP Redirect returns 300 Multiple choices with the termination IP in >>>> Contact and with FROM field as instructed ( update 551000 with 1140031556) >>>> >>>> SIP/2.0 300 Multiple Choices >>>> Via: SIP/2.0/UDP 216.x.x.x;rport;branch=z9hG4bKateZg87rDBpZa >>>> From: "1140031556" ;tag=FeNXS71300N0c >>>> To: >>>> Contact: ;q=1.00 >>>> Call-ID: e70ee145-aac6-1235-79ba-002590a0ec9b >>>> CSeq: 106579790 INVITE >>>> Max-Forwards: 69 >>>> Content-Length: 0 >>>> Server: SIP Redirect Server >>>> >>>> >>>> 4. FS will send the call to the Termination IP WITHOUT changing the >>>> FROM field >>>> >>>> INVITE sip:556230951662 at 162.x.x.x:5060 SIP/2.0 >>>> Via: SIP/2.0/UDP 216.x.x.x;rport;branch=z9hG4bKB37Qj3rvamcjp >>>> Max-Forwards: 68 >>>> From: "551000" ;tag=gQepU2j7X9BKr >>>> To: >>>> Call-ID: e7159715-aac6-1235-79ba-002590a0ec9b >>>> CSeq: 106579790 INVITE >>>> Contact: >>>> 2017-05-03 12:15:09.856127 [ERR] mod_xml_radius.c:914 Result of true >>>> match: 162.x.x.x:5060 == ^69\.x\.x\.x >>>> User-Agent: FreeSWITCH >>>> 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: 397 >>>> X-FS-Support: update_display,send_info >>>> Remote-Party-ID: "551000" ;party=c >>>> alling;screen=yes;privacy=off >>>> >>>> v=0 >>>> o=FreeSWITCH 1493811013 1493811014 IN IP4 216.x.x.x >>>> s=FreeSWITCH >>>> c=IN IP4 216.x.x.x >>>> t=0 0 >>>> m=audio 20696 RTP/AVP 8 0 18 101 13 >>>> a=rtpmap:8 PCMA/8000 >>>> a=rtpmap:0 PCMU/8000 >>>> a=rtpmap:18 G729/8000 >>>> a=rtpmap:101 telephone-event/8000 >>>> a=fmtp:101 0-16 >>>> a=ptime:20 >>>> m=audio 20696 RTP/AVP 4 101 13 >>>> a=rtpmap:4 G723/8000 >>>> a=rtpmap:101 telephone-event/8000 >>>> a=fmtp:101 0-16 >>>> a=ptime:30 >>>> >>>> This is the dialplan I use to send calls to SIP redirect server >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>> /> >>>> >>>> >>>> >>>> >>>> >>>> >>> /> >>>> >>>> >>>> >>>> >>>> >>> > >>>> >>>> >>>> >>> >>>> Is there any variable that will force FS to change the FROM field as >>>> returned by the SIP Redirect server and send it to the termination >>>> provider? Any help on this is really greatly appreciated! >>>> >>>> >>>> Best regards, >>>> >>>> L. Popov >>>> >>>> >>>> >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-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 >>>> >>> >>> >>> >>> -- >>> >>> *Brian West* >>> brian at freeswitch.org >>> >>> *Twitter: @FreeSWITCH , @briankwest* >>> >>> http://www.freeswitchbook.com >>> http://www.freeswitchcookbook.com >>> >>> Book a phone call (CST) >>> >>> Allison prompts for FreeSWITCH: >>> >>> *https://www.gofundme.com/allison-prompts-for-freeswitch* >>> >>> >>> Got Bugs? Report them here ! | Reddit: >>> /r/freeswitch >>> >>> *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 >>> <(918)%20420-9002> | *M:*+1918424WEST (9378) >>> *Skype:*briankwest >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 >> > > > > -- > > *Brian West* > brian at freeswitch.org > > *Twitter: @FreeSWITCH , @briankwest* > > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > Book a phone call (CST) > > Allison prompts for FreeSWITCH: > > *https://www.gofundme.com/allison-prompts-for-freeswitch* > > > Got Bugs? Report them here ! | Reddit: > /r/freeswitch > > *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002> > | *M:*+1918424WEST (9378) > *Skype:*briankwest > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Atenciosamente, ============================ Lyubo Popov CEO - BlasterPhone LLC Tel: 4003-1556 ( Outside Brazil 55 11 4003-1556) iNum: +883 510001-354111 Website: http://www.blastervoip.com.br/ ============================ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170514/94c91df0/attachment-0001.html From gmaruzz at gmail.com Mon May 15 10:13:38 2017 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Mon, 15 May 2017 08:13:38 +0200 Subject: [Freeswitch-users] SIPS scheme In-Reply-To: References: Message-ID: David, Thanks for reporting the solution. On May 15, 2017 1:37 AM, "David Villasmil" wrote: > I fixed it with kamilio like this: > > if(is_method("INVITE") && status=~"200") { > if( subst('/sip:/sips:/ig') ) { > xlog("L_ERR","[ONREPLY]: Changed contact [$ct]!\n"); > } > } > > On the onreply_route > > Thanks > > David > ? > > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 <+34%20669%2044%2083%2037> > > On Sun, May 14, 2017 at 5:20 AM, Sergey Safarov > wrote: > >> FS not supports SIPS scheme. Exist jira ticket about this. >> >> ??, 14 ??? 2017, 2:10 David Villasmil : >> >>> hello, thanks for replying. >>> >>> Yes, i thought about this, but asterisk has a patch to do this already >>> without any changes on kamailoi, and I was looking more in the same >>> direction. >>> >>> Regards, >>> >>> David >>> ? >>> >>> Regards, >>> >>> David Villasmil >>> email: david.villasmil.work at gmail.com >>> phone: +34669448337 <+34%20669%2044%2083%2037> >>> >>> On Sun, May 14, 2017 at 12:58 AM, Giovanni Maruzzelli >> > wrote: >>> >>>> If kamailio is under your control, you can change contact back and >>>> forth from it (eg, before forwarding to freeswitch) >>>> >>>> sent from mobile >>>> cell: +39 347 266 56 18 >>>> Giovanni Maruzzelli >>>> OpenTelecom.IT >>>> >>>> On May 13, 2017 5:01 PM, "David Villasmil" < >>>> david.villasmil.work at gmail.com> wrote: >>>> >>>>> Hello guys. >>>>> >>>>> I've got this setup, where the client is sending as contact "sip:" >>>>> coming from kamailio proxy in tls ip until kamailio. Kamailio forwards to >>>>> FS in tcp, but freeswitch is answering with "sip:" and this breaks the >>>>> client who answers "SIPS Required"... >>>>> >>>>> Is there a way of specifying the contact should be sips ? >>>>> >>>>> Has anyone encountered this and solved it? >>>>> >>>>> Regards, >>>>> >>>>> David Villasmil >>>>> email: david.villasmil.work at gmail.com >>>>> phone: +34669448337 <+34%20669%2044%2083%2037> >>>>> ? >>>>> >>>>> ____________________________________________________________ >>>>> _____________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170515/dea8045c/attachment.html From j.peral at airenetworks.es Fri May 12 18:46:31 2017 From: j.peral at airenetworks.es (Joaquin Peral) Date: Fri, 12 May 2017 16:46:31 +0200 Subject: [Freeswitch-users] Fwd: Change the signaling port after the ACK In-Reply-To: <1dc9b4a3-14a5-30e0-afde-4633000f4b4e@airenetworks.es> References: <1dc9b4a3-14a5-30e0-afde-4633000f4b4e@airenetworks.es> Message-ID: An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170512/4e85b39a/attachment.html From kk-mailinglist at ednt.de Sat May 13 11:15:09 2017 From: kk-mailinglist at ednt.de (kk-mailinglist at ednt.de) Date: Sat, 13 May 2017 09:15:09 +0200 Subject: [Freeswitch-users] Presence Server Message-ID: <5916B27D.3020207@ednt.de> i try to use the LED?s on the Phone to sgive user a indication of the state of my Phone is redirected or ithis Phone is subscribed in a Queue. For this i use a free extension in sample 1000 and put on the Phone the LED to BLF 1000. This Works until the phone subscribe again after 3600 sec. The my manipulation will be lost in the presence server. So i need a way to take care about this state. Is there a common solution for this issue ? regards From mike at jerris.com Mon May 15 19:11:46 2017 From: mike at jerris.com (Michael Jerris) Date: Mon, 15 May 2017 11:11:46 -0400 Subject: [Freeswitch-users] Change the signaling port after the ACK In-Reply-To: References: <1dc9b4a3-14a5-30e0-afde-4633000f4b4e@airenetworks.es> Message-ID: I?d have to see the sip trace to answer this better? From the ladder diagram, i don?t see the ack looking any difference than the invite. The BYE is sent I assume to exactly where the contact tells us to send it to. > On May 12, 2017, at 10:46 AM, Joaquin Peral wrote: > > Hi all, I have a special case, a user with a Cisco system. He uses the standard 5060 port but uses NAT. Everything is correct until send the ACK signaling and the port changes. Any ideas? > ? > > external cfg: > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > -- > [Joaquin Peral Cascales] > Departamento de Telefon?a > 911090048 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170515/fba133ff/attachment-0001.html From francesco.piccinin at insiel.it Mon May 15 20:41:11 2017 From: francesco.piccinin at insiel.it (Piccinin Francesco) Date: Mon, 15 May 2017 16:41:11 +0000 Subject: [Freeswitch-users] Freeswitch Time of Day Routing XML_curl Message-ID: Hi all, we are trying to setup Freeswitch ToD routing using XML curl for XML dialplan dynamic generation. Please note that our dialplan is mainly stored on static file (plus lua support) with few exceptions like IVRs. In order to avoid freeswitch sending http post on every call (as would happen adding binding=directory on xml_curl configuration module), we would like to find a way to generate dynamic xml dialplan only on particular case like ToD. Can you please give us some clues about this? Thanks Regards Francesco From brian at freeswitch.org Mon May 15 21:28:22 2017 From: brian at freeswitch.org (Brian West) Date: Mon, 15 May 2017 12:28:22 -0500 Subject: [Freeswitch-users] Freeswitch Time of Day Routing XML_curl In-Reply-To: References: Message-ID: ToD routing when using xml_curl makes no sense, You can already answer the request with exactly what needs to run for the current time the request is being made. /b On Mon, May 15, 2017 at 11:41 AM, Piccinin Francesco < francesco.piccinin at insiel.it> wrote: > Hi all, > we are trying to setup Freeswitch ToD routing using XML curl for XML > dialplan dynamic generation. > > Please note that our dialplan is mainly stored on static file (plus lua > support) with few exceptions like IVRs. > In order to avoid freeswitch sending http post on every call (as would > happen adding binding=directory on xml_curl configuration module), we would > like to find a way to generate dynamic xml dialplan only on particular case > like ToD. > > Can you please give us some clues about this? > > Thanks > Regards > > Francesco > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Book a phone call (CST) Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170515/dede5825/attachment.html From caioebassis at hotmail.com Mon May 15 22:30:52 2017 From: caioebassis at hotmail.com (Caio Assis) Date: Mon, 15 May 2017 18:30:52 +0000 Subject: [Freeswitch-users] MOD_SOFIA In-Reply-To: References: , Message-ID: It worked as expected. Thank You!! ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org on behalf of E. Schmidbauer Sent: Friday, May 5, 2017 12:34 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] MOD_SOFIA try this `sofia profile [profile name] rescan` On Fri, May 5, 2017 at 11:27 AM, Caio Assis > wrote: Good Afternoon. Is there a command I type that loads a new SIP account? When I type reload mod_sofia and it's already in use, it gives me an error message "+OK Reloading XML -ERR unloading module [Module in use.] -ERR loading module [Module already loaded]" I'm looking for an asterisk 'sip reload' freeswitch equivalent. If I have calls in progress and create a SIP account to use, I either have to wait for all calls to finish or restart Freeswitch, which makes it almost inviable. Can anyone help me with this? 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170515/9f82bced/attachment.html From gregor at infomedia.si Mon May 15 22:37:32 2017 From: gregor at infomedia.si (Gregor Nanger) Date: Mon, 15 May 2017 18:37:32 +0000 Subject: [Freeswitch-users] Freeswitch Time of Day Routing XML_curl In-Reply-To: References: Message-ID: Even if FS makes http post, you can offload FS with all logic in server side, that you can return only small dial plan. On Mon, May 15, 2017, 19:29 Brian West wrote: > ToD routing when using xml_curl makes no sense, You can already answer the > request with exactly what needs to run for the current time the request is > being made. > > /b > > > On Mon, May 15, 2017 at 11:41 AM, Piccinin Francesco < > francesco.piccinin at insiel.it> wrote: > >> Hi all, >> we are trying to setup Freeswitch ToD routing using XML curl for XML >> dialplan dynamic generation. >> >> Please note that our dialplan is mainly stored on static file (plus lua >> support) with few exceptions like IVRs. >> In order to avoid freeswitch sending http post on every call (as would >> happen adding binding=directory on xml_curl configuration module), we would >> like to find a way to generate dynamic xml dialplan only on particular case >> like ToD. >> >> Can you please give us some clues about this? >> >> Thanks >> Regards >> >> Francesco >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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.org > > *Twitter: @FreeSWITCH , @briankwest* > > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > Book a phone call (CST) > > Allison prompts for FreeSWITCH: > > *https://www.gofundme.com/allison-prompts-for-freeswitch* > > > Got Bugs? Report them here ! | Reddit: > /r/freeswitch > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *Skype:*briankwest > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170515/1a5e7c99/attachment-0001.html From bipin at xbipin.com Tue May 16 00:06:02 2017 From: bipin at xbipin.com (Bipin Patel) Date: Tue, 16 May 2017 00:06:02 +0400 Subject: [Freeswitch-users] Fax result when using mail to fax Message-ID: <15c0db7b410.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com> Hi, I have a python script that checks a mailbox for faxes to be sent, when it finds a new mail it downloads the PDF and converts to Tiff and sends via the originate command, now the problem is I can't seem to find a way to get the fax report, nor in cdr it says what happened and the originate just says ok followed by uuid of the bridge. I'm sending the fax over a fxo gateway device. Due to it going over fxo, I have no clue if the remote fax machine answered call and the only place the fax report comes is in the console. Is there a variable or anything else that can be resent to the python script telling what the result was. I tried the execute on fax successful and fail variable but they don't seem to work. My python script generates the originate command going to loopback which goes to dial plan where the actual bridge to fxo gateway happens. How are others dealing with fax report I would love to hear. Regards, Bipin From francesco.piccinin at insiel.it Tue May 16 00:19:48 2017 From: francesco.piccinin at insiel.it (Piccinin Francesco) Date: Mon, 15 May 2017 20:19:48 +0000 Subject: [Freeswitch-users] R: Freeswitch Time of Day Routing XML_curl In-Reply-To: References: Message-ID: Hi Brian, thanks for you answer. We already considered that case but we have to add ?dialplan? section binding for gateway-url tag on current xml_curl config file (we are just using gateways and directory bindings). This implies that freeswitch send an http request for every incoming call leg, isn?t it? Should we run into delay issue as long as registrations and calls rise up? Is there a way to use xml_curl just for part of then dialplan like it happens for ivrs? It seems that ivrs are using "configuration" section and not dialplan... Thanks Regards Francesco Da: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Per conto di Brian West Inviato: luned? 15 maggio 2017 19:28 A: FreeSWITCH Users Help Oggetto: Re: [Freeswitch-users] Freeswitch Time of Day Routing XML_curl ToD routing when using xml_curl makes no sense, You can already answer the request with exactly what needs to run for the current time the request is being made. /b On Mon, May 15, 2017 at 11:41 AM, Piccinin Francesco wrote: Hi all, we are trying to setup Freeswitch ToD routing using XML curl for XML dialplan dynamic generation. Please note that our dialplan is mainly stored on static file (plus lua support) with few exceptions like IVRs. In order to avoid freeswitch sending http post on every call (as would happen adding binding=directory on xml_curl configuration module), we would like to find a way to generate dynamic xml dialplan only on particular case like ToD. Can you please give us some clues about this? Thanks Regards Francesco _________________________________________________________________________ Professional FreeSWITCH Consulting Services: mailto:consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list mailto:FreeSWITCH-users at 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 mailto:brian at freeswitch.org Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com? http://www.freeswitchcookbook.com https://freeswitch.com/appointment Allison prompts for FreeSWITCH: https://www.gofundme.com/allison-prompts-for-freeswitch Got Bugs? Report them https://freeswitch.org/jira! | Reddit:?https://www.reddit.com/r/freeswitch T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) Skype:briankwest From lists at telefaks.de Tue May 16 00:24:44 2017 From: lists at telefaks.de (Peter Steinbach) Date: Mon, 15 May 2017 22:24:44 +0200 Subject: [Freeswitch-users] Fax result when using mail to fax In-Reply-To: <15c0db7b410.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com> References: <15c0db7b410.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com> Message-ID: <591A0E8C.6070700@telefaks.de> Hello Bipin, there are some events fired via ESL, if you subscribe to them. We evaluate txfaxresult event and hangup event. The only thing which is critcal here, is that the contents of the 2 events have to be joined, in order to have all informations. And in my experience txfaxresult and hangup event do not always come in the same order, dependend on which side of the channels hangs up first. Best regards Peter On 05/15/17 22:06, Bipin Patel wrote: > Hi, > > I have a python script that checks a mailbox for faxes to be sent, when it > finds a new mail it downloads the PDF and converts to Tiff and sends via > the originate command, now the problem is I can't seem to find a way to get > the fax report, nor in cdr it says what happened and the originate just > says ok followed by uuid of the bridge. I'm sending the fax over a fxo > gateway device. Due to it going over fxo, I have no clue if the remote fax > machine answered call and the only place the fax report comes is in the > console. Is there a variable or anything else that can be resent to the > python script telling what the result was. I tried the execute on fax > successful and fail variable but they don't seem to work. > > My python script generates the originate command going to loopback which > goes to dial plan where the actual bridge to fxo gateway happens. > > How are others dealing with fax report I would love to hear. > > 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 > -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de From francesco.piccinin at insiel.it Tue May 16 00:34:18 2017 From: francesco.piccinin at insiel.it (Piccinin Francesco) Date: Mon, 15 May 2017 20:34:18 +0000 Subject: [Freeswitch-users] R: Freeswitch Time of Day Routing XML_curl In-Reply-To: References: Message-ID: Hi Gregor, our platform is a cluster of FS server and all the config resides on a centralized DB hosted on HA servers. If I got it right, offloading logic on server side for all FS implies replicating data (and keep them updated and sync) from DB to all fs server. Is it really a good solution for scalability? Thanks Francesco Da: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Per conto di Gregor Nanger Inviato: luned? 15 maggio 2017 20:38 A: FreeSWITCH Users Help Oggetto: Re: [Freeswitch-users] Freeswitch Time of Day Routing XML_curl Even if FS makes http post, you can offload FS with all logic in server side, that you can return only small dial plan. On Mon, May 15, 2017, 19:29 Brian West wrote: ToD routing when using xml_curl makes no sense, You can already answer the request with exactly what needs to run for the current time the request is being made. /b On Mon, May 15, 2017 at 11:41 AM, Piccinin Francesco wrote: Hi all, we are trying to setup Freeswitch ToD routing using XML curl for XML dialplan dynamic generation. Please note that our dialplan is mainly stored on static file (plus lua support) with few exceptions like IVRs. In order to avoid freeswitch sending http post on every call (as would happen adding binding=directory on xml_curl configuration module), we would like to find a way to generate dynamic xml dialplan only on particular case like ToD. Can you please give us some clues about this? Thanks Regards Francesco _________________________________________________________________________ Professional FreeSWITCH Consulting Services: mailto:consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list mailto:FreeSWITCH-users at 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 mailto:brian at freeswitch.org Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com? http://www.freeswitchcookbook.com https://freeswitch.com/appointment Allison prompts for FreeSWITCH: https://www.gofundme.com/allison-prompts-for-freeswitch Got Bugs? Report them https://freeswitch.org/jira! | Reddit:?https://www.reddit.com/r/freeswitch T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) Skype:briankwest _________________________________________________________________________ Professional FreeSWITCH Consulting Services: mailto:consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list mailto:FreeSWITCH-users at 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? ??http://www.infomedia.si/ From gregor at infomedia.si Tue May 16 00:54:57 2017 From: gregor at infomedia.si (Gregor Nanger) Date: Mon, 15 May 2017 22:54:57 +0200 Subject: [Freeswitch-users] R: Freeswitch Time of Day Routing XML_curl In-Reply-To: References: Message-ID: I can't really comment regarding your architecture. But real power of FS shines when using with xml_curl. I am using FS with xml_curl to not let FS to make any decision or checking, but only to handle calls. For example, I even check on server side, where curl is generated, if extension is registered before transfering calls. In your case server side logic can check time of day and only tells FS even to play file and cancel or to transfer to extension. 2017-05-15 22:34 GMT+02:00 Piccinin Francesco : > Hi Gregor, > our platform is a cluster of FS server and all the config resides on a > centralized DB hosted on HA servers. > If I got it right, offloading logic on server side for all FS implies > replicating data (and keep them updated and sync) from DB to all fs server. > > Is it really a good solution for scalability? > > Thanks > Francesco > > Da: freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] Per conto di Gregor Nanger > Inviato: luned? 15 maggio 2017 20:38 > A: FreeSWITCH Users Help > Oggetto: Re: [Freeswitch-users] Freeswitch Time of Day Routing XML_curl > > Even if FS makes http post, you can offload FS with all logic in server > side, that you can return only small dial plan. > > On Mon, May 15, 2017, 19:29 Brian West > wrote: > ToD routing when using xml_curl makes no sense, You can already answer the > request with exactly what needs to run for the current time the request is > being made. > > /b > > > On Mon, May 15, 2017 at 11:41 AM, Piccinin Francesco francesco.piccinin at insiel.it> wrote: > Hi all, > we are trying to setup Freeswitch ToD routing using XML curl for XML > dialplan dynamic generation. > > Please note that our dialplan is mainly stored on static file (plus lua > support) with few exceptions like IVRs. > In order to avoid freeswitch sending http post on every call (as would > happen adding binding=directory on xml_curl configuration module), we would > like to find a way to generate dynamic xml dialplan only on particular case > like ToD. > > Can you please give us some clues about this? > > Thanks > Regards > > Francesco > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > mailto:consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > mailto:FreeSWITCH-users at 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 > mailto:brian at freeswitch.org > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > https://freeswitch.com/appointment > > Allison prompts for FreeSWITCH: > https://www.gofundme.com/allison-prompts-for-freeswitch > Got Bugs? Report them https://freeswitch.org/jira! | Reddit: > https://www.reddit.com/r/freeswitch > T:+19184209001 | F: > +19184209002 | M:+1918424WEST (9378) > Skype:briankwest > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > mailto:consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > mailto:FreeSWITCH-users at 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 > ? http://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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170515/8d64abd5/attachment-0001.html From krice at freeswitch.org Tue May 16 05:19:16 2017 From: krice at freeswitch.org (Ken Rice) Date: Mon, 15 May 2017 20:19:16 -0500 Subject: [Freeswitch-users] Question about enabling SCTP on freeswitch on ubuntu 16.04 from apt repository. Message-ID: <2646D0E2-8093-4808-8DCF-119701F2D197@freeswitch.org> Packages are not build with sctp support enabled. Sent from my iPhone > On May 15, 2017, at 17:56, Ravi sanyal wrote: > > Hi, > > I know this will be specific to my installation, so I'm running ubuntu 16.04 and downloaded freeswitch from the apt repository. Oh yeah, i also just did a clean install of freeswitch. I've only been using it for a few weeks. > > If i run the sofia dig command on the domain it only shows udp and tcp. > doing 'sofia status profile internal/external' does not have the line for sctp either. > > First thing, will i have to uninstall the repository and build it? i know that guides say to run the build with --enable-sctp i tried looking for sctp module (i know its a bad name, it's not a module) using apt however not familiar with what the name would be called. freeswitch-* is a mammoth list > > doing apt-cache search freeswitch | grep sctp doesn't find anything. > > Once i've set up freeswitch to handle it, i'll still need to do naptr and srv records on the domain. im running with rackspace. Has anyone used rackspace before? they say that naptr and srv can be added using their system (however there is no naptr record adding) > > Nathan > > -- > Virtual Guard Ltd > info at virtualguard.co.nz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170515/6dd37994/attachment.html From bipin at xbipin.com Tue May 16 09:25:36 2017 From: bipin at xbipin.com (Bipin Patel) Date: Tue, 16 May 2017 09:25:36 +0400 Subject: [Freeswitch-users] Fax result when using mail to fax In-Reply-To: <591A0E8C.6070700@telefaks.de> References: <15c0db7b410.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com> <591A0E8C.6070700@telefaks.de> Message-ID: hi, well considering im running this FS on the raspberry pi for a small office, ESL would be like a lot just for the purpose, if the console can display the fax result then why isnt it possible to get the same in dialplan or somewhere else where i can fire a script or something and let the sender know what happened to their fax. i tried using execute_on_fax_success and execute_on_fax_failure but they dont seem to work while sending a fax, i think its for receiving only Regards, Bipin ------------------------------------------------------------------------ -------- Original Message -------- Subject: Re: [Freeswitch-users] Fax result when using mail to fax From: Peter Steinbach To: FreeSWITCH Users Help Date: 5/16/2017, 12:24:44 AM > Hello Bipin, > > there are some events fired via ESL, if you subscribe to them. We > evaluate txfaxresult event and hangup event. > > The only thing which is critcal here, is that the contents of the 2 > events have to be joined, in order to have all informations. And in my > experience txfaxresult and hangup event do not always come in the same > order, dependend on which side of the channels hangs up first. > > Best regards > Peter > > On 05/15/17 22:06, Bipin Patel wrote: >> Hi, >> >> I have a python script that checks a mailbox for faxes to be sent, when it >> finds a new mail it downloads the PDF and converts to Tiff and sends via >> the originate command, now the problem is I can't seem to find a way to get >> the fax report, nor in cdr it says what happened and the originate just >> says ok followed by uuid of the bridge. I'm sending the fax over a fxo >> gateway device. Due to it going over fxo, I have no clue if the remote fax >> machine answered call and the only place the fax report comes is in the >> console. Is there a variable or anything else that can be resent to the >> python script telling what the result was. I tried the execute on fax >> successful and fail variable but they don't seem to work. >> >> My python script generates the originate command going to loopback which >> goes to dial plan where the actual bridge to fxo gateway happens. >> >> How are others dealing with fax report I would love to hear. >> >> 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 >> > > From lists at telefaks.de Tue May 16 11:38:18 2017 From: lists at telefaks.de (Peter Steinbach) Date: Tue, 16 May 2017 09:38:18 +0200 Subject: [Freeswitch-users] Fax result when using mail to fax In-Reply-To: References: <15c0db7b410.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com> <591A0E8C.6070700@telefaks.de> Message-ID: <591AAC6A.7060609@telefaks.de> Hello Bipin, the fax result is only available, long after the dialplan has beeen evaluated. So you will somehow have to use some event driven things. execute_on_fax_success would have been the next thing to propose, but id did not work in your case. In Linux you can use inotify. We have also used this method for getting certain infos from the Freeswitch logfile and then to fire actions (in our case firewall rules). Maybe this is an approach for you? Best regards Peter On 05/16/17 07:25, Bipin Patel wrote: > hi, > > well considering im running this FS on the raspberry pi for a small > office, ESL would be like a lot just for the purpose, if the console can > display the fax result then why isnt it possible to get the same in > dialplan or somewhere else where i can fire a script or something and > let the sender know what happened to their fax. > > i tried using execute_on_fax_success and execute_on_fax_failure but they > dont seem to work while sending a fax, i think its for receiving only > > > > Regards, > Bipin > > > > ------------------------------------------------------------------------ > > -------- Original Message -------- > Subject: Re: [Freeswitch-users] Fax result when using mail to fax > From: Peter Steinbach > To: FreeSWITCH Users Help > Date: 5/16/2017, 12:24:44 AM > >> Hello Bipin, >> >> there are some events fired via ESL, if you subscribe to them. We >> evaluate txfaxresult event and hangup event. >> >> The only thing which is critcal here, is that the contents of the 2 >> events have to be joined, in order to have all informations. And in my >> experience txfaxresult and hangup event do not always come in the same >> order, dependend on which side of the channels hangs up first. >> >> Best regards >> Peter >> >> On 05/15/17 22:06, Bipin Patel wrote: >>> Hi, >>> >>> I have a python script that checks a mailbox for faxes to be sent, when it >>> finds a new mail it downloads the PDF and converts to Tiff and sends via >>> the originate command, now the problem is I can't seem to find a way to get >>> the fax report, nor in cdr it says what happened and the originate just >>> says ok followed by uuid of the bridge. I'm sending the fax over a fxo >>> gateway device. Due to it going over fxo, I have no clue if the remote fax >>> machine answered call and the only place the fax report comes is in the >>> console. Is there a variable or anything else that can be resent to the >>> python script telling what the result was. I tried the execute on fax >>> successful and fail variable but they don't seem to work. >>> >>> My python script generates the originate command going to loopback which >>> goes to dial plan where the actual bridge to fxo gateway happens. >>> >>> How are others dealing with fax report I would love to hear. >>> >>> 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 > -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de From bipin at xbipin.com Tue May 16 13:32:57 2017 From: bipin at xbipin.com (Bipin Patel) Date: Tue, 16 May 2017 13:32:57 +0400 Subject: [Freeswitch-users] Fax result when using mail to fax In-Reply-To: <591AAC6A.7060609@telefaks.de> References: <15c0db7b410.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com> <591A0E8C.6070700@telefaks.de> <591AAC6A.7060609@telefaks.de> Message-ID: hi, i think i have one other plan, let me try that and ill get back with the result. i have on more issue, when i send a fax using the originate command, it results in 2 invites being sent to the gateway, one sends the fax and i get report for that but the other fails so i also get a failed report for that, any idea what could be causing this, my originate command is as below originate [ignore_early_media=true,fax_verbose=true,fax_enable_t38_request=true,fax_enable_t38=true,absolute_codec_string=PCMU]sofia/gateway/ma/123466789 &txfax(\'" + att_path + ".tiff\')" Regards, Bipin Talky Communications VoIP/SMS/DID Services +971-55-9270058 www.talkycom.com www.xbipin.com ------------------------------------------------------------------------ -------- Original Message -------- Subject: Re: [Freeswitch-users] Fax result when using mail to fax From: Peter Steinbach To: FreeSWITCH Users Help Date: 5/16/2017, 11:38:18 AM > Hello Bipin, > > the fax result is only available, long after the dialplan has beeen > evaluated. > > So you will somehow have to use some event driven things. > execute_on_fax_success would have been the next thing to propose, but id > did not work in your case. > > In Linux you can use inotify. We have also used this method for getting > certain infos from the Freeswitch logfile and then to fire actions (in > our case firewall rules). Maybe this is an approach for you? > > Best regards > Peter > > On 05/16/17 07:25, Bipin Patel wrote: >> hi, >> >> well considering im running this FS on the raspberry pi for a small >> office, ESL would be like a lot just for the purpose, if the console can >> display the fax result then why isnt it possible to get the same in >> dialplan or somewhere else where i can fire a script or something and >> let the sender know what happened to their fax. >> >> i tried using execute_on_fax_success and execute_on_fax_failure but they >> dont seem to work while sending a fax, i think its for receiving only >> >> >> >> Regards, >> Bipin >> >> >> >> ------------------------------------------------------------------------ >> >> -------- Original Message -------- >> Subject: Re: [Freeswitch-users] Fax result when using mail to fax >> From: Peter Steinbach >> To: FreeSWITCH Users Help >> Date: 5/16/2017, 12:24:44 AM >> >>> Hello Bipin, >>> >>> there are some events fired via ESL, if you subscribe to them. We >>> evaluate txfaxresult event and hangup event. >>> >>> The only thing which is critcal here, is that the contents of the 2 >>> events have to be joined, in order to have all informations. And in my >>> experience txfaxresult and hangup event do not always come in the same >>> order, dependend on which side of the channels hangs up first. >>> >>> Best regards >>> Peter >>> >>> On 05/15/17 22:06, Bipin Patel wrote: >>>> Hi, >>>> >>>> I have a python script that checks a mailbox for faxes to be sent, when it >>>> finds a new mail it downloads the PDF and converts to Tiff and sends via >>>> the originate command, now the problem is I can't seem to find a way to get >>>> the fax report, nor in cdr it says what happened and the originate just >>>> says ok followed by uuid of the bridge. I'm sending the fax over a fxo >>>> gateway device. Due to it going over fxo, I have no clue if the remote fax >>>> machine answered call and the only place the fax report comes is in the >>>> console. Is there a variable or anything else that can be resent to the >>>> python script telling what the result was. I tried the execute on fax >>>> successful and fail variable but they don't seem to work. >>>> >>>> My python script generates the originate command going to loopback which >>>> goes to dial plan where the actual bridge to fxo gateway happens. >>>> >>>> How are others dealing with fax report I would love to hear. >>>> >>>> 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 >> > > From yu at yu-boot.ru Tue May 16 16:31:42 2017 From: yu at yu-boot.ru (Yu Boot) Date: Tue, 16 May 2017 15:31:42 +0300 Subject: [Freeswitch-users] T.38 in new FS versions In-Reply-To: <15c0db7b410.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com> References: <15c0db7b410.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com> Message-ID: <81c86120-7679-4357-55a3-a66bf009870f@yu-boot.ru> Hi there. How exactly to enable "classic" T.38 faxes with reINVITE in new FS versions? Not transcoding, not fax-to-email etc. From rbetancor at gmail.com Tue May 16 17:14:29 2017 From: rbetancor at gmail.com (=?UTF-8?Q?Ra=C3=BAl_Alexis_Betancor_Santana?=) Date: Tue, 16 May 2017 14:14:29 +0100 Subject: [Freeswitch-users] Noob question about xml dialplan Message-ID: Hi all, till now I'm been working with pre-made setup files for FS, and now I'm trying to get a deep knowleadge of how the dialplan works. So I modifed my autoload_configs/sofia.conf.xml file and changed my dialplan param to something like this: The Idea is that it loads another .xml file especific task and also use a dialplan throught a socket to a daemon that handle the rest. On my t38_transcode.xml file ... very simple: But If I fire a call to FS like [number]@[FS_IP] from a sip testing client I get [INFO] switch_core_state_machine.c:311 No Route, Aborting What I'm doing wrong here? My target its just I want to ANY call that came in with a SIP header of X-T39-Transcode=True (market by a sip proxy elsewhere on the net) just do a T38->Ulaw transcoding saving the T38 trace, so I could inspect it later. The rest of calls coming in ... as they don't have the sip-header should end on other app. For doing the testing I disabled the socket_inline part of the dialplan string, so it have only this: Did I miss something? ... or maybe missundestood who the xml dialplan works ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170516/ea3501f7/attachment-0001.html From david.villasmil.work at gmail.com Tue May 16 17:22:21 2017 From: david.villasmil.work at gmail.com (David Villasmil) Date: Tue, 16 May 2017 13:22:21 +0000 Subject: [Freeswitch-users] Noob question about xml dialplan In-Reply-To: References: Message-ID: Take a look at your profile, it should be listening on the port you're sending to, and must have the context parameter set to your dialplan name. On Tue, May 16, 2017 at 3:15 PM Ra?l Alexis Betancor Santana < rbetancor at gmail.com> wrote: > > Hi all, till now I'm been working with pre-made setup files for FS, and > now I'm trying to get a deep knowleadge of how the dialplan works. > > So I modifed my autoload_configs/sofia.conf.xml file and changed my > dialplan param to something like this: > > value="XML:/etc/freeswitch/dialplans/t38_transcode.xml,inline:socket: > 127.0.0.1:8022 async full"/> > > The Idea is that it loads another .xml file especific task and also use a > dialplan throught a socket to a daemon that handle the rest. > > On my t38_transcode.xml file ... very simple: > > > > expresion="^(\+1|1)?([2-9]\d\d[2-9]\d{6})$"> > > > > > > > > > But If I fire a call to FS like [number]@[FS_IP] from a sip testing client > I get > > [INFO] switch_core_state_machine.c:311 No Route, Aborting > > What I'm doing wrong here? > > My target its just I want to ANY call that came in with a SIP header of > X-T39-Transcode=True (market by a sip proxy elsewhere on the net) just do a > T38->Ulaw transcoding saving the T38 trace, so I could inspect it later. > The rest of calls coming in ... as they don't have the sip-header should > end on other app. > > For doing the testing I disabled the socket_inline part of the dialplan > string, so it have only this: > > value="XML:/etc/freeswitch/dialplans/t38_transcode.xml"/> > > Did I miss something? ... or maybe missundestood who the xml dialplan > works ? > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170516/60f5eeee/attachment.html From rbetancor at gmail.com Tue May 16 17:30:54 2017 From: rbetancor at gmail.com (=?UTF-8?Q?Ra=C3=BAl_Alexis_Betancor_Santana?=) Date: Tue, 16 May 2017 14:30:54 +0100 Subject: [Freeswitch-users] Noob question about xml dialplan In-Reply-To: References: Message-ID: Only one profile defined on the sofia.conf.xml and I'm sending the traffict to the wright ports, if not I whould get the logs on the console, as that are the only ports enabled. 2017-05-16 14:22 GMT+01:00 David Villasmil : > Take a look at your profile, it should be listening on the port you're > sending to, and must have the context parameter set to your dialplan name. > On Tue, May 16, 2017 at 3:15 PM Ra?l Alexis Betancor Santana < > rbetancor at gmail.com> wrote: > >> >> Hi all, till now I'm been working with pre-made setup files for FS, and >> now I'm trying to get a deep knowleadge of how the dialplan works. >> >> So I modifed my autoload_configs/sofia.conf.xml file and changed my >> dialplan param to something like this: >> >> >> >> The Idea is that it loads another .xml file especific task and also use a >> dialplan throught a socket to a daemon that handle the rest. >> >> On my t38_transcode.xml file ... very simple: >> >> >> >> >> >> >> >> >> >> >> >> >> But If I fire a call to FS like [number]@[FS_IP] from a sip testing >> client I get >> >> [INFO] switch_core_state_machine.c:311 No Route, Aborting >> >> What I'm doing wrong here? >> >> My target its just I want to ANY call that came in with a SIP header of >> X-T39-Transcode=True (market by a sip proxy elsewhere on the net) just do a >> T38->Ulaw transcoding saving the T38 trace, so I could inspect it later. >> The rest of calls coming in ... as they don't have the sip-header should >> end on other app. >> >> For doing the testing I disabled the socket_inline part of the dialplan >> string, so it have only this: >> >> >> >> Did I miss something? ... or maybe missundestood who the xml dialplan >> works ? >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170516/9ade3132/attachment.html From j.peral at airenetworks.es Tue May 16 11:09:36 2017 From: j.peral at airenetworks.es (Joaquin Peral) Date: Tue, 16 May 2017 09:09:36 +0200 Subject: [Freeswitch-users] Change the signaling port after the ACK In-Reply-To: References: <1dc9b4a3-14a5-30e0-afde-4633000f4b4e@airenetworks.es> Message-ID: <4dcb448a-f77a-ffda-e824-66dcdfbcb6ae@airenetworks.es> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170516/14321f26/attachment-0001.html From 398572406 at 163.com Tue May 16 13:49:53 2017 From: 398572406 at 163.com (hancymoon) Date: Tue, 16 May 2017 02:49:53 -0700 (MST) Subject: [Freeswitch-users] REFER message with replaces from Alcate OmniPCX Server Message-ID: <1494928193178-7596313.post@n2.nabble.com> Hello everyone: I have a internal sip phone(A) registered to Freeswitch. If A call a number via siptrunk to Alcate's phone(B). B answered, then transfer this call to Alcate's phone(C), if C is still ringing, but B hangup, then C stop ringing and hangup, and there is no connection bwtween A and C. Please help me to check why A cannot connect to C. Here is the log: recv 688 bytes from udp/[160.7.237.4]:5060 at 12:18:20.272035: ------------------------------------------------------------------------ REFER sip:gw+siptrunk1 at 160.7.237.89:5080;transport=udp;gw=siptrunk1 SIP/2.0 Contact: sip:160.7.237.4 Supported: timer,path,100rel User-Agent: OmniPCX Enterprise R11.2.2 l2.300.33.a Refer-To: Referred-By: sip:1260 at 160.7.237.4 To: sip:66577160 at 160.7.237.89;tag=vaZK46cXrgeyQ From: sip:1260 at 160.7.237.4:5060;tag=85e354db02f8d15fe12b28a385cb8861 Call-ID: 5334d53d-b3c8-1235-dfbf-109836ab50bb CSeq: 1904933290 REFER Via: SIP/2.0/UDP 160.7.237.4;branch=z9hG4bK7d39c893d0c0fd0deab78b450e6efd31 Max-Forwards: 70 Content-Length: 0 ------------------------------------------------------------------------ 7bd68496-3925-11e7-9cfa-29f19fd18425 2017-05-15 12:18:20.270464 [DEBUG] sofia.c:8487 Process REFER to [7950 at 160.7.237.4] 7bd68496-3925-11e7-9cfa-29f19fd18425 2017-05-15 12:18:20.270464 [DEBUG] sofia.c:8513 Replaces: [5334d53d-b3c8-1235-dfbf-109836ab50bb] send 613 bytes to udp/[160.7.237.4]:5060 at 12:18:20.290107: ------------------------------------------------------------------------ SIP/2.0 202 Accepted Via: SIP/2.0/UDP 160.7.237.4;branch=z9hG4bK7d39c893d0c0fd0deab78b450e6efd31 From: sip:1260 at 160.7.237.4:5060;tag=85e354db02f8d15fe12b28a385cb8861 To: sip:66577160 at 160.7.237.89;tag=vaZK46cXrgeyQ Call-ID: 5334d53d-b3c8-1235-dfbf-109836ab50bb CSeq: 1904933290 REFER Contact: Expires: 60 User-Agent: FreeSWITCH-mod_sofia/1.6.15~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Length: 0 ------------------------------------------------------------------------ 7bd68496-3925-11e7-9cfa-29f19fd18425 2017-05-15 12:18:20.270464 [NOTICE] sofia.c:8710 Attended Transfer [791cb18a-3925-11e7-9cf0-29f19fd18425][791cb18a-3925-11e7-9cf0-29f19fd18425] 791cb18a-3925-11e7-9cf0-29f19fd18425 2017-05-15 12:18:20.270464 [DEBUG] switch_ivr_bridge.c:1982 (FreeTDM/1:21/1260) State Change CS_EXECUTE -> CS_HIBERNATE 791cb18a-3925-11e7-9cf0-29f19fd18425 2017-05-15 12:18:20.270464 [DEBUG] switch_ivr_bridge.c:1984 (FreeTDM/1:21/1260) State Change CS_HIBERNATE -> CS_CONSUME_MEDIA 7bd68496-3925-11e7-9cfa-29f19fd18425 2017-05-15 12:18:20.270464 [DEBUG] switch_channel.c:2029 (sofia/external/1260) Callstate Change HELD -> UNHELD 7bd68496-3925-11e7-9cfa-29f19fd18425 2017-05-15 12:18:20.270464 [DEBUG] sofia.c:8785 (sofia/external/1260) State Change CS_EXCHANGE_MEDIA -> CS_PARK send 774 bytes to udp/[160.7.237.4]:5060 at 12:18:20.290288: ------------------------------------------------------------------------ NOTIFY sip:160.7.237.4 SIP/2.0 Via: SIP/2.0/UDP 160.7.237.89:5080;rport;branch=z9hG4bKH8BaH8ejBemBB Max-Forwards: 70 From: "66577160" ;tag=vaZK46cXrgeyQ To: ;tag=85e354db02f8d15fe12b28a385cb8861 Call-ID: 5334d53d-b3c8-1235-dfbf-109836ab50bb CSeq: 107074877 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.6.15~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Event: refer;id=1904933290 Allow-Events: talk, hold, conference, refer Subscription-State: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Content-Length: 16 SIP/2.0 200 OK ------------------------------------------------------------------------ 791cb18a-3925-11e7-9cf0-29f19fd18425 2017-05-15 12:18:20.290465 [DEBUG] switch_ivr_play_say.c:1942 done playing file local_stream://moh recv 545 bytes from udp/[160.7.237.4]:5060 at 12:18:20.304851: ------------------------------------------------------------------------ SIP/2.0 200 OK Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, REFER, SUBSCRIBE, OPTIONS, UPDATE, INFO Contact: sip:160.7.237.4 Supported: replaces,timer,path,100rel User-Agent: OmniPCX Enterprise R11.2.2 l2.300.33.a To: ;tag=85e354db02f8d15fe12b28a385cb8861 From: "66577160" ;tag=vaZK46cXrgeyQ Call-ID: 5334d53d-b3c8-1235-dfbf-109836ab50bb CSeq: 107074877 NOTIFY Via: SIP/2.0/UDP 160.7.237.89:5080;received=160.7.237.89;rport=5080;branch=z9hG4bKH8BaH8ejBemBB Content-Length: 0 ------------------------------------------------------------------------ recv 470 bytes from udp/[160.7.237.4]:5060 at 12:18:20.310591: ------------------------------------------------------------------------ BYE sip:gw+siptrunk1 at 160.7.237.89:5080;transport=udp;gw=siptrunk1 SIP/2.0 Supported: replaces,timer,path,100rel User-Agent: OmniPCX Enterprise R11.2.2 l2.300.33.a To: sip:66577160 at 160.7.237.89;tag=vaZK46cXrgeyQ From: sip:1260 at 160.7.237.4:5060;tag=85e354db02f8d15fe12b28a385cb8861 Call-ID: 5334d53d-b3c8-1235-dfbf-109836ab50bb CSeq: 1904933291 BYE Via: SIP/2.0/UDP 160.7.237.4;branch=z9hG4bKf69a857f743ba04db1f37bb7af0337f8 Max-Forwards: 70 Content-Length: 0 ------------------------------------------------------------------------ 791cb18a-3925-11e7-9cf0-29f19fd18425 2017-05-15 12:18:20.310465 [DEBUG] switch_ivr_bridge.c:752 BRIDGE THREAD DONE [FreeTDM/1:21/1260] 791cb18a-3925-11e7-9cf0-29f19fd18425 2017-05-15 12:18:25.530465 [DEBUG] switch_ivr_bridge.c:1729 (FreeTDM/1:21/1260) State Change CS_CONSUME_MEDIA -> CS_RESET 7bd68496-3925-11e7-9cfa-29f19fd18425 2017-05-15 12:18:25.530465 [NOTICE] fssession.cpp:139 Hangup sofia/external/1260 [CS_PARK] [NORMAL_CLEARING] 2017-05-15 12:18:25.530465 [INFO] mod_v8.cpp:627 Javascript result: [true] 7bd68496-3925-11e7-9cfa-29f19fd18425 2017-05-15 12:18:25.530465 [DEBUG] switch_ivr_bridge.c:752 BRIDGE THREAD DONE [sofia/external/1260] 7bd68496-3925-11e7-9cfa-29f19fd18425 2017-05-15 12:18:25.530465 [DEBUG] switch_core_state_machine.c:653 (sofia/external/1260) State EXCHANGE_MEDIA going to sleep 7bd68496-3925-11e7-9cfa-29f19fd18425 2017-05-15 12:18:25.530465 [DEBUG] switch_core_state_machine.c:584 (sofia/external/1260) Running State Change CS_HANGUP (Cur 5 Tot 17168) 7bd68496-3925-11e7-9cfa-29f19fd18425 2017-05-15 12:18:25.530465 [DEBUG] sofia.c:1453 Channel is already hungup. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/REFER-message-with-replaces-from-Alcate-OmniPCX-Server-tp7596313.html Sent from the freeswitch-users mailing list archive at Nabble.com. From Agusti.Ubalde at enghouse.com Tue May 16 14:55:15 2017 From: Agusti.Ubalde at enghouse.com (Agusti Ubalde) Date: Tue, 16 May 2017 10:55:15 +0000 Subject: [Freeswitch-users] Verto calls Message-ID: <3e80ecf35e984499825b26c52173587a@UK-MAIL-001.edge.local> Hi all, I am trying to call from Verto extension to another Verto extension. Both are successfully registered (Verto status show the successfully register) but the call between is not established. The call remains in ring state. This is the last dialplan function executed (calling 1000 to 1001): EXECUTE verto.rtc/1001 bridge() Thanks, PRESENCE TECHNOLOGY An ENGHOUSE INTERACTIVE Company Agust? Ubalde Bellot Chief Developer C/ Comte Urgell 240 3?-A Barcelona 08036 aubalde at presenceco.com Ph: +34 93 10 10 322 Enghouse is listed on the Toronto Stock Exchange, stock symbol: ENGH -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170516/30eae6fd/attachment.html From virtualguard2015 at gmail.com Tue May 16 02:56:02 2017 From: virtualguard2015 at gmail.com (Ravi sanyal) Date: Tue, 16 May 2017 10:56:02 +1200 Subject: [Freeswitch-users] Question about enabling SCTP on freeswitch on ubuntu 16.04 from apt repository. Message-ID: Hi, I know this will be specific to my installation, so I'm running ubuntu 16.04 and downloaded freeswitch from the apt repository. Oh yeah, i also just did a clean install of freeswitch. I've only been using it for a few weeks. If i run the sofia dig command on the domain it only shows udp and tcp. doing 'sofia status profile internal/external' does not have the line for sctp either. First thing, will i have to uninstall the repository and build it? i know that guides say to run the build with --enable-sctp i tried looking for sctp module (i know its a bad name, it's not a module) using apt however not familiar with what the name would be called. freeswitch-* is a mammoth list doing apt-cache search freeswitch | grep sctp doesn't find anything. Once i've set up freeswitch to handle it, i'll still need to do naptr and srv records on the domain. im running with rackspace. Has anyone used rackspace before? they say that naptr and srv can be added using their system (however there is no naptr record adding) Nathan -- *Virtual Guard Ltd* *info at virtualguard.co.nz * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170516/772c682e/attachment-0001.html From virtualguard2015 at gmail.com Tue May 16 07:56:22 2017 From: virtualguard2015 at gmail.com (Ravi sanyal) Date: Tue, 16 May 2017 15:56:22 +1200 Subject: [Freeswitch-users] Question about enabling SCTP on freeswitch on ubuntu 16.04 from apt repository. In-Reply-To: <2646D0E2-8093-4808-8DCF-119701F2D197@freeswitch.org> References: <2646D0E2-8093-4808-8DCF-119701F2D197@freeswitch.org> Message-ID: Hello, Thanks :) that helps a lot. Nathan On Tue, May 16, 2017 at 1:19 PM, Ken Rice wrote: > Packages are not build with sctp support enabled. > > Sent from my iPhone > > On May 15, 2017, at 17:56, Ravi sanyal wrote: > > Hi, > > I know this will be specific to my installation, so I'm running ubuntu > 16.04 and downloaded freeswitch from the apt repository. Oh yeah, i also > just did a clean install of freeswitch. I've only been using it for a few > weeks. > > If i run the sofia dig command on the domain it only shows udp and tcp. > doing 'sofia status profile internal/external' does not have the line for > sctp either. > > First thing, will i have to uninstall the repository and build it? i know > that guides say to run the build with --enable-sctp i tried looking for > sctp module (i know its a bad name, it's not a module) using apt however > not familiar with what the name would be called. freeswitch-* is a mammoth > list > > doing apt-cache search freeswitch | grep sctp doesn't find anything. > > Once i've set up freeswitch to handle it, i'll still need to do naptr and > srv records on the domain. im running with rackspace. Has anyone used > rackspace before? they say that naptr and srv can be added using their > system (however there is no naptr record adding) > > Nathan > > -- > > *Virtual Guard Ltd* > *info at virtualguard.co.nz * > > -- *Virtual Guard Ltd* *info at virtualguard.co.nz * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170516/d2f28eae/attachment-0001.html From mike at jerris.com Tue May 16 17:41:30 2017 From: mike at jerris.com (Michael Jerris) Date: Tue, 16 May 2017 13:41:30 +0000 Subject: [Freeswitch-users] Fax result when using mail to fax In-Reply-To: References: <15c0db7b410.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com> <591A0E8C.6070700@telefaks.de> <591AAC6A.7060609@telefaks.de> Message-ID: a single originate will only create a single call. What you describe here isn't happening as you describe it. A log of the calls may tell you more. On Tue, May 16, 2017 at 5:36 AM Bipin Patel wrote: > hi, > > i think i have one other plan, let me try that and ill get back with the > result. > > i have on more issue, when i send a fax using the originate command, it > results in 2 invites being sent to the gateway, one sends the fax and i > get report for that but the other fails so i also get a failed report > for that, any idea what could be causing this, my originate command is > as below > > originate > > [ignore_early_media=true,fax_verbose=true,fax_enable_t38_request=true,fax_enable_t38=true,absolute_codec_string=PCMU]sofia/gateway/ma/123466789 > &txfax(\'" + att_path + ".tiff\')" > > > > Regards, > Bipin > > > Talky Communications > VoIP/SMS/DID Services > +971-55-9270058 > www.talkycom.com > www.xbipin.com > ------------------------------------------------------------------------ > > -------- Original Message -------- > Subject: Re: [Freeswitch-users] Fax result when using mail to fax > From: Peter Steinbach > To: FreeSWITCH Users Help > Date: 5/16/2017, 11:38:18 AM > > > Hello Bipin, > > > > the fax result is only available, long after the dialplan has beeen > > evaluated. > > > > So you will somehow have to use some event driven things. > > execute_on_fax_success would have been the next thing to propose, but id > > did not work in your case. > > > > In Linux you can use inotify. We have also used this method for getting > > certain infos from the Freeswitch logfile and then to fire actions (in > > our case firewall rules). Maybe this is an approach for you? > > > > Best regards > > Peter > > > > On 05/16/17 07:25, Bipin Patel wrote: > >> hi, > >> > >> well considering im running this FS on the raspberry pi for a small > >> office, ESL would be like a lot just for the purpose, if the console can > >> display the fax result then why isnt it possible to get the same in > >> dialplan or somewhere else where i can fire a script or something and > >> let the sender know what happened to their fax. > >> > >> i tried using execute_on_fax_success and execute_on_fax_failure but they > >> dont seem to work while sending a fax, i think its for receiving only > >> > >> > >> > >> Regards, > >> Bipin > >> > >> > >> > >> ------------------------------------------------------------------------ > >> > >> -------- Original Message -------- > >> Subject: Re: [Freeswitch-users] Fax result when using mail to fax > >> From: Peter Steinbach > >> To: FreeSWITCH Users Help > >> Date: 5/16/2017, 12:24:44 AM > >> > >>> Hello Bipin, > >>> > >>> there are some events fired via ESL, if you subscribe to them. We > >>> evaluate txfaxresult event and hangup event. > >>> > >>> The only thing which is critcal here, is that the contents of the 2 > >>> events have to be joined, in order to have all informations. And in my > >>> experience txfaxresult and hangup event do not always come in the same > >>> order, dependend on which side of the channels hangs up first. > >>> > >>> Best regards > >>> Peter > >>> > >>> On 05/15/17 22:06, Bipin Patel wrote: > >>>> Hi, > >>>> > >>>> I have a python script that checks a mailbox for faxes to be sent, > when it > >>>> finds a new mail it downloads the PDF and converts to Tiff and sends > via > >>>> the originate command, now the problem is I can't seem to find a way > to get > >>>> the fax report, nor in cdr it says what happened and the originate > just > >>>> says ok followed by uuid of the bridge. I'm sending the fax over a fxo > >>>> gateway device. Due to it going over fxo, I have no clue if the > remote fax > >>>> machine answered call and the only place the fax report comes is in > the > >>>> console. Is there a variable or anything else that can be resent to > the > >>>> python script telling what the result was. I tried the execute on fax > >>>> successful and fail variable but they don't seem to work. > >>>> > >>>> My python script generates the originate command going to loopback > which > >>>> goes to dial plan where the actual bridge to fxo gateway happens. > >>>> > >>>> How are others dealing with fax report I would love to hear. > >>>> > >>>> 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170516/61b0599e/attachment.html From mike at jerris.com Tue May 16 17:45:54 2017 From: mike at jerris.com (Michael Jerris) Date: Tue, 16 May 2017 13:45:54 +0000 Subject: [Freeswitch-users] Change the signaling port after the ACK In-Reply-To: <4dcb448a-f77a-ffda-e824-66dcdfbcb6ae@airenetworks.es> References: <1dc9b4a3-14a5-30e0-afde-4633000f4b4e@airenetworks.es> <4dcb448a-f77a-ffda-e824-66dcdfbcb6ae@airenetworks.es> Message-ID: you most likely never want aggressive nat detection on. Turning it on in most situations is a sign you don't have other nat config correct, or simply thought it might be good. It should not be used unless absolutely required, which it rarely is. On Tue, May 16, 2017 at 9:40 AM Joaquin Peral wrote: > FS assumes that the user is doing NAT. > > Solved with > > > > in a new profile. > > Thanks! > > > On 15/05/17 17:11, Michael Jerris wrote: > > I?d have to see the sip trace to answer this better? From the ladder > diagram, i don?t see the ack looking any difference than the invite. The > BYE is sent I assume to exactly where the contact tells us to send it to. > > > On May 12, 2017, at 10:46 AM, Joaquin Peral > wrote: > > Hi all, I have a special case, a user with a Cisco system. He uses the > standard 5060 port but uses NAT. Everything is correct until send the ACK > signaling and the port changes. Any ideas? > > ? > > external cfg: > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > -- > [Joaquin Peral Cascales] > Departamento de Telefon?a > 911090048 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 > > > -- > [Joaquin Peral Cascales] > Departamento de Telefon?a > 911090048 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170516/4058c6f2/attachment-0001.html From mike at jerris.com Tue May 16 17:47:38 2017 From: mike at jerris.com (Michael Jerris) Date: Tue, 16 May 2017 13:47:38 +0000 Subject: [Freeswitch-users] REFER message with replaces from Alcate OmniPCX Server In-Reply-To: <1494928193178-7596313.post@n2.nabble.com> References: <1494928193178-7596313.post@n2.nabble.com> Message-ID: On most sip phones, hanging up does not complete a transfer, it just hangs up. On Tue, May 16, 2017 at 9:40 AM hancymoon <398572406 at 163.com> wrote: > Hello everyone: > > I have a internal sip phone(A) registered to Freeswitch. If A call a number > via siptrunk to Alcate's phone(B). B answered, then transfer this call to > Alcate's phone(C), if C is still ringing, but B hangup, then C stop ringing > and hangup, and there is no connection bwtween A and C. > Please help me to check why A cannot connect to C. > > Here is the log: > recv 688 bytes from udp/[160.7.237.4]:5060 at 12:18:20.272035: > ------------------------------------------------------------------------ > REFER sip:gw+siptrunk1 at 160.7.237.89:5080;transport=udp;gw=siptrunk1 > SIP/2.0 > Contact: sip:160.7.237.4 > Supported: timer,path,100rel > User-Agent: OmniPCX Enterprise R11.2.2 l2.300.33.a > Refer-To: > ;user=phone?REPLACES=5334d53d-b3c8-1235-dfbf-109836ab50bb%3bto-tag%3dvaZK46cXrgeyQ%3bfrom-tag%3d85e354db02f8d15fe12b28a385cb8861> > Referred-By: sip:1260 at 160.7.237.4 > To: sip:66577160 at 160.7.237.89;tag=vaZK46cXrgeyQ > From: sip:1260 at 160.7.237.4:5060;tag=85e354db02f8d15fe12b28a385cb8861 > Call-ID: 5334d53d-b3c8-1235-dfbf-109836ab50bb > CSeq: 1904933290 REFER > Via: SIP/2.0/UDP > 160.7.237.4;branch=z9hG4bK7d39c893d0c0fd0deab78b450e6efd31 > Max-Forwards: 70 > Content-Length: 0 > > ------------------------------------------------------------------------ > 7bd68496-3925-11e7-9cfa-29f19fd18425 2017-05-15 12:18:20.270464 [DEBUG] > sofia.c:8487 Process REFER to [7950 at 160.7.237.4] > 7bd68496-3925-11e7-9cfa-29f19fd18425 2017-05-15 12:18:20.270464 [DEBUG] > sofia.c:8513 Replaces: [5334d53d-b3c8-1235-dfbf-109836ab50bb] > send 613 bytes to udp/[160.7.237.4]:5060 at 12:18:20.290107: > ------------------------------------------------------------------------ > SIP/2.0 202 Accepted > Via: SIP/2.0/UDP > 160.7.237.4;branch=z9hG4bK7d39c893d0c0fd0deab78b450e6efd31 > From: sip:1260 at 160.7.237.4:5060;tag=85e354db02f8d15fe12b28a385cb8861 > To: sip:66577160 at 160.7.237.89;tag=vaZK46cXrgeyQ > Call-ID: 5334d53d-b3c8-1235-dfbf-109836ab50bb > CSeq: 1904933290 REFER > Contact: 89:5080;transport=udp;gw=siptrunk1> > Expires: 60 > User-Agent: FreeSWITCH-mod_sofia/1.6.15~64bit > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, > REGISTER, REFER, NOTIFY > Supported: timer, path, replaces > Allow-Events: talk, hold, conference, refer > Content-Length: 0 > > ------------------------------------------------------------------------ > 7bd68496-3925-11e7-9cfa-29f19fd18425 2017-05-15 12:18:20.270464 [NOTICE] > sofia.c:8710 Attended Transfer > > [791cb18a-3925-11e7-9cf0-29f19fd18425][791cb18a-3925-11e7-9cf0-29f19fd18425] > 791cb18a-3925-11e7-9cf0-29f19fd18425 2017-05-15 12:18:20.270464 [DEBUG] > switch_ivr_bridge.c:1982 (FreeTDM/1:21/1260) State Change CS_EXECUTE -> > CS_HIBERNATE > 791cb18a-3925-11e7-9cf0-29f19fd18425 2017-05-15 12:18:20.270464 [DEBUG] > switch_ivr_bridge.c:1984 (FreeTDM/1:21/1260) State Change CS_HIBERNATE -> > CS_CONSUME_MEDIA > 7bd68496-3925-11e7-9cfa-29f19fd18425 2017-05-15 12:18:20.270464 [DEBUG] > switch_channel.c:2029 (sofia/external/1260) Callstate Change HELD -> UNHELD > 7bd68496-3925-11e7-9cfa-29f19fd18425 2017-05-15 12:18:20.270464 [DEBUG] > sofia.c:8785 (sofia/external/1260) State Change CS_EXCHANGE_MEDIA -> > CS_PARK > send 774 bytes to udp/[160.7.237.4]:5060 at 12:18:20.290288: > ------------------------------------------------------------------------ > NOTIFY sip:160.7.237.4 SIP/2.0 > Via: SIP/2.0/UDP 160.7.237.89:5080;rport;branch=z9hG4bKH8BaH8ejBemBB > Max-Forwards: 70 > From: "66577160" ;tag=vaZK46cXrgeyQ > To: ;tag=85e354db02f8d15fe12b28a385cb8861 > Call-ID: 5334d53d-b3c8-1235-dfbf-109836ab50bb > CSeq: 107074877 NOTIFY > Contact: 89:5080;transport=udp;gw=siptrunk1> > User-Agent: FreeSWITCH-mod_sofia/1.6.15~64bit > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, > REGISTER, REFER, NOTIFY > Supported: timer, path, replaces > Event: refer;id=1904933290 > Allow-Events: talk, hold, conference, refer > Subscription-State: terminated;reason=noresource > Content-Type: message/sipfrag;version=2.0 > Content-Length: 16 > > SIP/2.0 200 OK > ------------------------------------------------------------------------ > 791cb18a-3925-11e7-9cf0-29f19fd18425 2017-05-15 12:18:20.290465 [DEBUG] > switch_ivr_play_say.c:1942 done playing file local_stream://moh > recv 545 bytes from udp/[160.7.237.4]:5060 at 12:18:20.304851: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, REFER, SUBSCRIBE, > OPTIONS, UPDATE, INFO > Contact: sip:160.7.237.4 > Supported: replaces,timer,path,100rel > User-Agent: OmniPCX Enterprise R11.2.2 l2.300.33.a > To: ;tag=85e354db02f8d15fe12b28a385cb8861 > From: "66577160" ;tag=vaZK46cXrgeyQ > Call-ID: 5334d53d-b3c8-1235-dfbf-109836ab50bb > CSeq: 107074877 NOTIFY > Via: SIP/2.0/UDP > 160.7.237.89:5080 > ;received=160.7.237.89;rport=5080;branch=z9hG4bKH8BaH8ejBemBB > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 470 bytes from udp/[160.7.237.4]:5060 at 12:18:20.310591: > ------------------------------------------------------------------------ > BYE sip:gw+siptrunk1 at 160.7.237.89:5080;transport=udp;gw=siptrunk1 > SIP/2.0 > Supported: replaces,timer,path,100rel > User-Agent: OmniPCX Enterprise R11.2.2 l2.300.33.a > To: sip:66577160 at 160.7.237.89;tag=vaZK46cXrgeyQ > From: sip:1260 at 160.7.237.4:5060;tag=85e354db02f8d15fe12b28a385cb8861 > Call-ID: 5334d53d-b3c8-1235-dfbf-109836ab50bb > CSeq: 1904933291 BYE > Via: SIP/2.0/UDP > 160.7.237.4;branch=z9hG4bKf69a857f743ba04db1f37bb7af0337f8 > Max-Forwards: 70 > Content-Length: 0 > > ------------------------------------------------------------------------ > 791cb18a-3925-11e7-9cf0-29f19fd18425 2017-05-15 12:18:20.310465 [DEBUG] > switch_ivr_bridge.c:752 BRIDGE THREAD DONE [FreeTDM/1:21/1260] > 791cb18a-3925-11e7-9cf0-29f19fd18425 2017-05-15 12:18:25.530465 [DEBUG] > switch_ivr_bridge.c:1729 (FreeTDM/1:21/1260) State Change CS_CONSUME_MEDIA > -> CS_RESET > 7bd68496-3925-11e7-9cfa-29f19fd18425 2017-05-15 12:18:25.530465 [NOTICE] > fssession.cpp:139 Hangup sofia/external/1260 [CS_PARK] [NORMAL_CLEARING] > 2017-05-15 12:18:25.530465 [INFO] mod_v8.cpp:627 Javascript result: [true] > 7bd68496-3925-11e7-9cfa-29f19fd18425 2017-05-15 12:18:25.530465 [DEBUG] > switch_ivr_bridge.c:752 BRIDGE THREAD DONE [sofia/external/1260] > 7bd68496-3925-11e7-9cfa-29f19fd18425 2017-05-15 12:18:25.530465 [DEBUG] > switch_core_state_machine.c:653 (sofia/external/1260) State EXCHANGE_MEDIA > going to sleep > 7bd68496-3925-11e7-9cfa-29f19fd18425 2017-05-15 12:18:25.530465 [DEBUG] > switch_core_state_machine.c:584 (sofia/external/1260) Running State Change > CS_HANGUP (Cur 5 Tot 17168) > 7bd68496-3925-11e7-9cfa-29f19fd18425 2017-05-15 12:18:25.530465 [DEBUG] > sofia.c:1453 Channel is already hungup. > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/REFER-message-with-replaces-from-Alcate-OmniPCX-Server-tp7596313.html > Sent from the freeswitch-users mailing list archive at Nabble.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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170516/18b915f6/attachment.html From bipin at xbipin.com Tue May 16 18:24:22 2017 From: bipin at xbipin.com (Bipin Patel) Date: Tue, 16 May 2017 18:24:22 +0400 Subject: [Freeswitch-users] Fax result when using mail to fax In-Reply-To: References: <15c0db7b410.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com> <591A0E8C.6070700@telefaks.de> <591AAC6A.7060609@telefaks.de> Message-ID: <17851f77-08bf-ca7d-4284-e943631f347f@xbipin.com> hi, well i was also not sure what was happening but after doing some packet traces im seeing 2 invites being generated and both run parallel so one fails later on and one transmits the fax successfully, shall i paste a debug log? Regards, Bipin ------------------------------------------------------------------------ -------- Original Message -------- Subject: Re: [Freeswitch-users] Fax result when using mail to fax From: Michael Jerris To: FreeSWITCH Users Help Date: 5/16/2017, 5:41:30 PM > a single originate will only create a single call. What you describe > here isn't happening as you describe it. A log of the calls may tell > you more. > > On Tue, May 16, 2017 at 5:36 AM Bipin Patel > wrote: > > hi, > > i think i have one other plan, let me try that and ill get back > with the > result. > > i have on more issue, when i send a fax using the originate > command, it > results in 2 invites being sent to the gateway, one sends the fax > and i > get report for that but the other fails so i also get a failed report > for that, any idea what could be causing this, my originate command is > as below > > originate > [ignore_early_media=true,fax_verbose=true,fax_enable_t38_request=true,fax_enable_t38=true,absolute_codec_string=PCMU]sofia/gateway/ma/123466789 > &txfax(\'" + att_path + ".tiff\')" > > > > Regards, > Bipin > > > Talky Communications > VoIP/SMS/DID Services > +971-55-9270058 > www.talkycom.com > www.xbipin.com > ------------------------------------------------------------------------ > > -------- Original Message -------- > Subject: Re: [Freeswitch-users] Fax result when using mail to fax > From: Peter Steinbach > > To: FreeSWITCH Users Help > > Date: 5/16/2017, 11:38:18 AM > > > Hello Bipin, > > > > the fax result is only available, long after the dialplan has beeen > > evaluated. > > > > So you will somehow have to use some event driven things. > > execute_on_fax_success would have been the next thing to > propose, but id > > did not work in your case. > > > > In Linux you can use inotify. We have also used this method for > getting > > certain infos from the Freeswitch logfile and then to fire > actions (in > > our case firewall rules). Maybe this is an approach for you? > > > > Best regards > > Peter > > > > On 05/16/17 07:25, Bipin Patel wrote: > >> hi, > >> > >> well considering im running this FS on the raspberry pi for a small > >> office, ESL would be like a lot just for the purpose, if the > console can > >> display the fax result then why isnt it possible to get the same in > >> dialplan or somewhere else where i can fire a script or > something and > >> let the sender know what happened to their fax. > >> > >> i tried using execute_on_fax_success and execute_on_fax_failure > but they > >> dont seem to work while sending a fax, i think its for > receiving only > >> > >> > >> > >> Regards, > >> Bipin > >> > >> > >> > >> > ------------------------------------------------------------------------ > >> > >> -------- Original Message -------- > >> Subject: Re: [Freeswitch-users] Fax result when using mail to fax > >> From: Peter Steinbach > > >> To: FreeSWITCH Users Help > > > >> Date: 5/16/2017, 12:24:44 AM > >> > >>> Hello Bipin, > >>> > >>> there are some events fired via ESL, if you subscribe to them. We > >>> evaluate txfaxresult event and hangup event. > >>> > >>> The only thing which is critcal here, is that the contents of > the 2 > >>> events have to be joined, in order to have all informations. > And in my > >>> experience txfaxresult and hangup event do not always come in > the same > >>> order, dependend on which side of the channels hangs up first. > >>> > >>> Best regards > >>> Peter > >>> > >>> On 05/15/17 22:06, Bipin Patel wrote: > >>>> Hi, > >>>> > >>>> I have a python script that checks a mailbox for faxes to be > sent, when it > >>>> finds a new mail it downloads the PDF and converts to Tiff > and sends via > >>>> the originate command, now the problem is I can't seem to > find a way to get > >>>> the fax report, nor in cdr it says what happened and the > originate just > >>>> says ok followed by uuid of the bridge. I'm sending the fax > over a fxo > >>>> gateway device. Due to it going over fxo, I have no clue if > the remote fax > >>>> machine answered call and the only place the fax report comes > is in the > >>>> console. Is there a variable or anything else that can be > resent to the > >>>> python script telling what the result was. I tried the > execute on fax > >>>> successful and fail variable but they don't seem to work. > >>>> > >>>> My python script generates the originate command going to > loopback which > >>>> goes to dial plan where the actual bridge to fxo gateway happens. > >>>> > >>>> How are others dealing with fax report I would love to hear. > >>>> > >>>> 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170516/368dfe77/attachment-0001.html From david.villasmil.work at gmail.com Tue May 16 19:13:53 2017 From: david.villasmil.work at gmail.com (David Villasmil) Date: Tue, 16 May 2017 17:13:53 +0200 Subject: [Freeswitch-users] Noob question about xml dialplan In-Reply-To: References: Message-ID: No, you may be sending it to the right ports, but the profile attached to that port must have the context set to the correct dialplan. I just did exactly what you're doing and i also got "No Route, Aborting" I've never seen this type of "dialplan" value, tbh Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 On Tue, May 16, 2017 at 3:30 PM, Ra?l Alexis Betancor Santana < rbetancor at gmail.com> wrote: > Only one profile defined on the sofia.conf.xml and I'm sending the > traffict to the wright ports, if not I whould get the logs on the console, > as that are the only ports enabled. > > 2017-05-16 14:22 GMT+01:00 David Villasmil >: > >> Take a look at your profile, it should be listening on the port you're >> sending to, and must have the context parameter set to your dialplan name. >> On Tue, May 16, 2017 at 3:15 PM Ra?l Alexis Betancor Santana < >> rbetancor at gmail.com> wrote: >> >>> >>> Hi all, till now I'm been working with pre-made setup files for FS, and >>> now I'm trying to get a deep knowleadge of how the dialplan works. >>> >>> So I modifed my autoload_configs/sofia.conf.xml file and changed my >>> dialplan param to something like this: >>> >>> >>> >>> The Idea is that it loads another .xml file especific task and also use >>> a dialplan throught a socket to a daemon that handle the rest. >>> >>> On my t38_transcode.xml file ... very simple: >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> But If I fire a call to FS like [number]@[FS_IP] from a sip testing >>> client I get >>> >>> [INFO] switch_core_state_machine.c:311 No Route, Aborting >>> >>> What I'm doing wrong here? >>> >>> My target its just I want to ANY call that came in with a SIP header of >>> X-T39-Transcode=True (market by a sip proxy elsewhere on the net) just do a >>> T38->Ulaw transcoding saving the T38 trace, so I could inspect it later. >>> The rest of calls coming in ... as they don't have the sip-header should >>> end on other app. >>> >>> For doing the testing I disabled the socket_inline part of the dialplan >>> string, so it have only this: >>> >>> >>> >>> Did I miss something? ... or maybe missundestood who the xml dialplan >>> works ? >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170516/8f7560bf/attachment.html From Agusti.Ubalde at enghouse.com Tue May 16 19:07:55 2017 From: Agusti.Ubalde at enghouse.com (Agusti Ubalde) Date: Tue, 16 May 2017 15:07:55 +0000 Subject: [Freeswitch-users] Verto calls Message-ID: <72189bc7b6ce4503ae6430b6c7f4d905@UK-MAIL-001.edge.local> Hi all, I am trying to call from Verto extension to another Verto extension. Both are successfully registered (Verto status show the successfully register) but the call between is not established. The call remains in ring state. This is the last dialplan function executed (calling 1000 to 1001): EXECUTE verto.rtc/1001 bridge() Thanks, PRESENCE TECHNOLOGY An ENGHOUSE INTERACTIVE Company Agust? Ubalde Bellot Chief Developer C/ Comte Urgell 240 3?-A Barcelona 08036 aubalde at presenceco.com Ph: +34 93 10 10 322 Enghouse is listed on the Toronto Stock Exchange, stock symbol: ENGH -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170516/465c195d/attachment.html From mike at jerris.com Tue May 16 19:16:51 2017 From: mike at jerris.com (Michael Jerris) Date: Tue, 16 May 2017 11:16:51 -0400 Subject: [Freeswitch-users] Fax result when using mail to fax In-Reply-To: <17851f77-08bf-ca7d-4284-e943631f347f@xbipin.com> References: <15c0db7b410.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com> <591A0E8C.6070700@telefaks.de> <591AAC6A.7060609@telefaks.de> <17851f77-08bf-ca7d-4284-e943631f347f@xbipin.com> Message-ID: <364C278B-FB7D-4F6E-9864-05C55E945057@jerris.com> You should investigate in your logs to see what is really going on? I?d set the assumptions of what is going on to the side and review logs to see whats happening. > On May 16, 2017, at 10:24 AM, Bipin Patel wrote: > > hi, > > well i was also not sure what was happening but after doing some packet traces im seeing 2 invites being generated and both run parallel so one fails later on and one transmits the fax successfully, shall i paste a debug log? > > > Regards, > Bipin > > > -------- Original Message -------- > Subject: Re: [Freeswitch-users] Fax result when using mail to fax > From: Michael Jerris > To: FreeSWITCH Users Help > Date: 5/16/2017, 5:41:30 PM >> a single originate will only create a single call. What you describe here isn't happening as you describe it. A log of the calls may tell you more. >> >> On Tue, May 16, 2017 at 5:36 AM Bipin Patel > wrote: >> hi, >> >> i think i have one other plan, let me try that and ill get back with the >> result. >> >> i have on more issue, when i send a fax using the originate command, it >> results in 2 invites being sent to the gateway, one sends the fax and i >> get report for that but the other fails so i also get a failed report >> for that, any idea what could be causing this, my originate command is >> as below >> >> originate >> [ignore_early_media=true,fax_verbose=true,fax_enable_t38_request=true,fax_enable_t38=true,absolute_codec_string=PCMU]sofia/gateway/ma/123466789 >> &txfax(\'" + att_path + ".tiff\')" >> >> >> >> Regards, >> Bipin >> >> >> Talky Communications >> VoIP/SMS/DID Services >> +971-55-9270058 >> www.talkycom.com > >> www.xbipin.com > >> ------------------------------------------------------------------------ >> >> -------- Original Message -------- >> Subject: Re: [Freeswitch-users] Fax result when using mail to fax >> From: Peter Steinbach > >> To: FreeSWITCH Users Help > >> Date: 5/16/2017, 11:38:18 AM >> >> > Hello Bipin, >> > >> > the fax result is only available, long after the dialplan has beeen >> > evaluated. >> > >> > So you will somehow have to use some event driven things. >> > execute_on_fax_success would have been the next thing to propose, but id >> > did not work in your case. >> > >> > In Linux you can use inotify. We have also used this method for getting >> > certain infos from the Freeswitch logfile and then to fire actions (in >> > our case firewall rules). Maybe this is an approach for you? >> > >> > Best regards >> > Peter >> > >> > On 05/16/17 07:25, Bipin Patel wrote: >> >> hi, >> >> >> >> well considering im running this FS on the raspberry pi for a small >> >> office, ESL would be like a lot just for the purpose, if the console can >> >> display the fax result then why isnt it possible to get the same in >> >> dialplan or somewhere else where i can fire a script or something and >> >> let the sender know what happened to their fax. >> >> >> >> i tried using execute_on_fax_success and execute_on_fax_failure but they >> >> dont seem to work while sending a fax, i think its for receiving only >> >> >> >> >> >> >> >> Regards, >> >> Bipin >> >> >> >> >> >> >> >> ------------------------------------------------------------------------ >> >> >> >> -------- Original Message -------- >> >> Subject: Re: [Freeswitch-users] Fax result when using mail to fax >> >> From: Peter Steinbach > >> >> To: FreeSWITCH Users Help > >> >> Date: 5/16/2017, 12:24:44 AM >> >> >> >>> Hello Bipin, >> >>> >> >>> there are some events fired via ESL, if you subscribe to them. We >> >>> evaluate txfaxresult event and hangup event. >> >>> >> >>> The only thing which is critcal here, is that the contents of the 2 >> >>> events have to be joined, in order to have all informations. And in my >> >>> experience txfaxresult and hangup event do not always come in the same >> >>> order, dependend on which side of the channels hangs up first. >> >>> >> >>> Best regards >> >>> Peter >> >>> >> >>> On 05/15/17 22:06, Bipin Patel wrote: >> >>>> Hi, >> >>>> >> >>>> I have a python script that checks a mailbox for faxes to be sent, when it >> >>>> finds a new mail it downloads the PDF and converts to Tiff and sends via >> >>>> the originate command, now the problem is I can't seem to find a way to get >> >>>> the fax report, nor in cdr it says what happened and the originate just >> >>>> says ok followed by uuid of the bridge. I'm sending the fax over a fxo >> >>>> gateway device. Due to it going over fxo, I have no clue if the remote fax >> >>>> machine answered call and the only place the fax report comes is in the >> >>>> console. Is there a variable or anything else that can be resent to the >> >>>> python script telling what the result was. I tried the execute on fax >> >>>> successful and fail variable but they don't seem to work. >> >>>> >> >>>> My python script generates the originate command going to loopback which >> >>>> goes to dial plan where the actual bridge to fxo gateway happens. >> >>>> >> >>>> How are others dealing with fax report I would love to hear. >> >>>> >> >>>> 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170516/4f186c20/attachment-0001.html From mike at jerris.com Tue May 16 19:19:08 2017 From: mike at jerris.com (Michael Jerris) Date: Tue, 16 May 2017 11:19:08 -0400 Subject: [Freeswitch-users] Noob question about xml dialplan In-Reply-To: References: Message-ID: <8E1259F0-5E34-46D0-8DED-50CD2A4B5795@jerris.com> this is totally valid syntax. It means parse that exact static file. > On May 16, 2017, at 11:13 AM, David Villasmil wrote: > > No, you may be sending it to the right ports, but the profile attached to that port must have the context set to the correct dialplan. > > I just did exactly what you're doing and i also got "No Route, Aborting" > > I've never seen this type of "dialplan" value, tbh > > > > > On Tue, May 16, 2017 at 3:30 PM, Ra?l Alexis Betancor Santana > wrote: > Only one profile defined on the sofia.conf.xml and I'm sending the traffict to the wright ports, if not I whould get the logs on the console, as that are the only ports enabled. > > 2017-05-16 14:22 GMT+01:00 David Villasmil >: > Take a look at your profile, it should be listening on the port you're sending to, and must have the context parameter set to your dialplan name. > On Tue, May 16, 2017 at 3:15 PM Ra?l Alexis Betancor Santana > wrote: > > Hi all, till now I'm been working with pre-made setup files for FS, and now I'm trying to get a deep knowleadge of how the dialplan works. > > So I modifed my autoload_configs/sofia.conf.xml file and changed my dialplan param to something like this: > > > > The Idea is that it loads another .xml file especific task and also use a dialplan throught a socket to a daemon that handle the rest. > > On my t38_transcode.xml file ... very simple: > > > > > > > > > > > > But If I fire a call to FS like [number]@[FS_IP] from a sip testing client I get > > [INFO] switch_core_state_machine.c:311 No Route, Aborting > > What I'm doing wrong here? > > My target its just I want to ANY call that came in with a SIP header of X-T39-Transcode=True (market by a sip proxy elsewhere on the net) just do a T38->Ulaw transcoding saving the T38 trace, so I could inspect it later. > The rest of calls coming in ... as they don't have the sip-header should end on other app. > > For doing the testing I disabled the socket_inline part of the dialplan string, so it have only this: > > > > Did I miss something? ... or maybe missundestood who the xml dialplan works ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170516/08a3d02d/attachment.html From aubalde at presenceco.com Tue May 16 19:19:20 2017 From: aubalde at presenceco.com (=?UTF-8?Q?Agust=C3=AD_Ubalde?=) Date: Tue, 16 May 2017 17:19:20 +0200 Subject: [Freeswitch-users] Verto calls Message-ID: Hi all, I am trying to call from Verto extension to another Verto extension. *Both are successfully registered* (Verto status show the successfully register) but the call between is not established. The call remains in ring state. This is the last dialplan function executed (calling 1000 to 1001): *EXECUTE verto.rtc/1001 bridge()* Regards, PRESENCE TECHNOLOGY An ENGHOUSE INTERACTIVE Company Agust? Ubalde Bellot Chief Developer C/ Comte Urgell 240 3?-A Barcelona 08036 aubalde at presenceco.com Ph: +34 93 10 10 322 Enghouse is listed on the Toronto Stock Exchange, stock symbol: ENGH -- *Presence Technology - DisclaimerThis message, its content and any file attached thereto is for the intended recipient only and is confidential and /or privileged. If you have received this e-mail in error or had access to it, you should note that the information in it is private and any use thereof is unauthorized. In such an event please notify us by e-mail or by telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by whatsoever means and any transmission or dissemination thereof to other persons is prohibited. It should be deleted immediately from your system. Presence Technology reserves the right to take legal action against any persons unlawfully gaining access to the content of any external message it has emitted.* *For additional information, please visit our website **www.presenceco.com * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170516/9c51d1a4/attachment.html From rbetancor at gmail.com Tue May 16 19:26:12 2017 From: rbetancor at gmail.com (=?UTF-8?Q?Ra=C3=BAl_Alexis_Betancor_Santana?=) Date: Tue, 16 May 2017 16:26:12 +0100 Subject: [Freeswitch-users] Noob question about xml dialplan In-Reply-To: References: Message-ID: Ok, I think the whole point is that I don't undestand how the dialplan works. I started from 0 ... now on the sofia.conf.xml I have this: Then on the t38_transconde.xml file I have this. I could see on the /var/log/freeswitch/freeswitch.xml.fsxml file that the t38_transcode.xml file its included ... but still get the same error on on the console, so something terrible wrong I'm missing here ... :-( Also ... it's possible to directly send the calls sofia/$1$2@$${external} ? ... I mean, sending the call directly to a URI, with username/password and so on ... instead of having to define a profile for it. 2017-05-16 16:13 GMT+01:00 David Villasmil : > No, you may be sending it to the right ports, but the profile attached to > that port must have the context set to the correct dialplan. > > I just did exactly what you're doing and i also got "No Route, Aborting" > > I've never seen this type of "dialplan" value, tbh > > > > > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 <+34%20669%2044%2083%2037> > > On Tue, May 16, 2017 at 3:30 PM, Ra?l Alexis Betancor Santana < > rbetancor at gmail.com> wrote: > >> Only one profile defined on the sofia.conf.xml and I'm sending the >> traffict to the wright ports, if not I whould get the logs on the console, >> as that are the only ports enabled. >> >> 2017-05-16 14:22 GMT+01:00 David Villasmil > m>: >> >>> Take a look at your profile, it should be listening on the port you're >>> sending to, and must have the context parameter set to your dialplan name. >>> On Tue, May 16, 2017 at 3:15 PM Ra?l Alexis Betancor Santana < >>> rbetancor at gmail.com> wrote: >>> >>>> >>>> Hi all, till now I'm been working with pre-made setup files for FS, and >>>> now I'm trying to get a deep knowleadge of how the dialplan works. >>>> >>>> So I modifed my autoload_configs/sofia.conf.xml file and changed my >>>> dialplan param to something like this: >>>> >>>> >>>> >>>> The Idea is that it loads another .xml file especific task and also use >>>> a dialplan throught a socket to a daemon that handle the rest. >>>> >>>> On my t38_transcode.xml file ... very simple: >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> But If I fire a call to FS like [number]@[FS_IP] from a sip testing >>>> client I get >>>> >>>> [INFO] switch_core_state_machine.c:311 No Route, Aborting >>>> >>>> What I'm doing wrong here? >>>> >>>> My target its just I want to ANY call that came in with a SIP header of >>>> X-T39-Transcode=True (market by a sip proxy elsewhere on the net) just do a >>>> T38->Ulaw transcoding saving the T38 trace, so I could inspect it later. >>>> The rest of calls coming in ... as they don't have the sip-header >>>> should end on other app. >>>> >>>> For doing the testing I disabled the socket_inline part of the dialplan >>>> string, so it have only this: >>>> >>>> >>>> >>>> Did I miss something? ... or maybe missundestood who the xml dialplan >>>> works ? >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170516/c538e7d9/attachment-0001.html From bipin at xbipin.com Tue May 16 19:57:39 2017 From: bipin at xbipin.com (Bipin Patel) Date: Tue, 16 May 2017 19:57:39 +0400 Subject: [Freeswitch-users] Fax result when using mail to fax In-Reply-To: <364C278B-FB7D-4F6E-9864-05C55E945057@jerris.com> References: <15c0db7b410.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com> <591A0E8C.6070700@telefaks.de> <591AAC6A.7060609@telefaks.de> <17851f77-08bf-ca7d-4284-e943631f347f@xbipin.com> <364C278B-FB7D-4F6E-9864-05C55E945057@jerris.com> Message-ID: <96ce819b-f29c-a157-5cf4-13d76be1d9fb@xbipin.com> hi, after a lot of digging figured out the issue, python script wasnt behaving properly, also managed to get the FAX report when fax was sent and then email it to sender, ill post the details soon once i have everything cleaned up. one last thing i wanted to know is im using a lua script which in turn calls a python script to send the mail with the FAX result but how do we print all the variable available for use, actually i want to know the dialed number where the fax was sent so can also print that in the report, im using the below in the originate. Manged to get it working api_hangup_hook=\'lua /home/pi/hangup.lua\' Regards, Bipin ------------------------------------------------------------------------ -------- Original Message -------- Subject: Re: [Freeswitch-users] Fax result when using mail to fax From: Michael Jerris To: FreeSWITCH Users Help Date: 5/16/2017, 7:16:51 PM > You should investigate in your logs to see what is really going on? > I?d set the assumptions of what is going on to the side and review > logs to see whats happening. > >> On May 16, 2017, at 10:24 AM, Bipin Patel > > wrote: >> >> hi, >> >> well i was also not sure what was happening but after doing some >> packet traces im seeing 2 invites being generated and both run >> parallel so one fails later on and one transmits the fax >> successfully, shall i paste a debug log? >> >> >> Regards, >> Bipin >> >> >> ------------------------------------------------------------------------ >> -------- Original Message -------- >> Subject: Re: [Freeswitch-users] Fax result when using mail to fax >> From: Michael Jerris >> To: FreeSWITCH Users Help >> Date: 5/16/2017, 5:41:30 PM >>> a single originate will only create a single call. What you >>> describe here isn't happening as you describe it. A log of the >>> calls may tell you more. >>> >>> On Tue, May 16, 2017 at 5:36 AM Bipin Patel >> > wrote: >>> >>> hi, >>> >>> i think i have one other plan, let me try that and ill get back >>> with the >>> result. >>> >>> i have on more issue, when i send a fax using the originate >>> command, it >>> results in 2 invites being sent to the gateway, one sends the >>> fax and i >>> get report for that but the other fails so i also get a failed >>> report >>> for that, any idea what could be causing this, my originate >>> command is >>> as below >>> >>> originate >>> [ignore_early_media=true,fax_verbose=true,fax_enable_t38_request=true,fax_enable_t38=true,absolute_codec_string=PCMU]sofia/gateway/ma/123466789 >>> &txfax(\'" + att_path + ".tiff\')" >>> >>> >>> >>> Regards, >>> Bipin >>> >>> >>> Talky Communications >>> VoIP/SMS/DID Services >>> +971-55-9270058 >>> www.talkycom.com >>> >>> www.xbipin.com >>> ------------------------------------------------------------------------ >>> >>> -------- Original Message -------- >>> Subject: Re: [Freeswitch-users] Fax result when using mail to fax >>> From: Peter Steinbach > >>> To: FreeSWITCH Users Help >> > >>> Date: 5/16/2017, 11:38:18 AM >>> >>> > Hello Bipin, >>> > >>> > the fax result is only available, long after the dialplan has >>> beeen >>> > evaluated. >>> > >>> > So you will somehow have to use some event driven things. >>> > execute_on_fax_success would have been the next thing to >>> propose, but id >>> > did not work in your case. >>> > >>> > In Linux you can use inotify. We have also used this method >>> for getting >>> > certain infos from the Freeswitch logfile and then to fire >>> actions (in >>> > our case firewall rules). Maybe this is an approach for you? >>> > >>> > Best regards >>> > Peter >>> > >>> > On 05/16/17 07:25, Bipin Patel wrote: >>> >> hi, >>> >> >>> >> well considering im running this FS on the raspberry pi for a >>> small >>> >> office, ESL would be like a lot just for the purpose, if the >>> console can >>> >> display the fax result then why isnt it possible to get the >>> same in >>> >> dialplan or somewhere else where i can fire a script or >>> something and >>> >> let the sender know what happened to their fax. >>> >> >>> >> i tried using execute_on_fax_success and >>> execute_on_fax_failure but they >>> >> dont seem to work while sending a fax, i think its for >>> receiving only >>> >> >>> >> >>> >> >>> >> Regards, >>> >> Bipin >>> >> >>> >> >>> >> >>> >> >>> ------------------------------------------------------------------------ >>> >> >>> >> -------- Original Message -------- >>> >> Subject: Re: [Freeswitch-users] Fax result when using mail to fax >>> >> From: Peter Steinbach >> > >>> >> To: FreeSWITCH Users Help >>> >> > >>> >> Date: 5/16/2017, 12:24:44 AM >>> >> >>> >>> Hello Bipin, >>> >>> >>> >>> there are some events fired via ESL, if you subscribe to >>> them. We >>> >>> evaluate txfaxresult event and hangup event. >>> >>> >>> >>> The only thing which is critcal here, is that the contents >>> of the 2 >>> >>> events have to be joined, in order to have all informations. >>> And in my >>> >>> experience txfaxresult and hangup event do not always come >>> in the same >>> >>> order, dependend on which side of the channels hangs up first. >>> >>> >>> >>> Best regards >>> >>> Peter >>> >>> >>> >>> On 05/15/17 22:06, Bipin Patel wrote: >>> >>>> Hi, >>> >>>> >>> >>>> I have a python script that checks a mailbox for faxes to >>> be sent, when it >>> >>>> finds a new mail it downloads the PDF and converts to Tiff >>> and sends via >>> >>>> the originate command, now the problem is I can't seem to >>> find a way to get >>> >>>> the fax report, nor in cdr it says what happened and the >>> originate just >>> >>>> says ok followed by uuid of the bridge. I'm sending the fax >>> over a fxo >>> >>>> gateway device. Due to it going over fxo, I have no clue if >>> the remote fax >>> >>>> machine answered call and the only place the fax report >>> comes is in the >>> >>>> console. Is there a variable or anything else that can be >>> resent to the >>> >>>> python script telling what the result was. I tried the >>> execute on fax >>> >>>> successful and fail variable but they don't seem to work. >>> >>>> >>> >>>> My python script generates the originate command going to >>> loopback which >>> >>>> goes to dial plan where the actual bridge to fxo gateway >>> happens. >>> >>>> >>> >>>> How are others dealing with fax report I would love to hear. >>> >>>> >>> >>>> 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170516/68e2f51e/attachment-0001.html From david.villasmil.work at gmail.com Tue May 16 20:33:03 2017 From: david.villasmil.work at gmail.com (David Villasmil) Date: Tue, 16 May 2017 18:33:03 +0200 Subject: [Freeswitch-users] Noob question about xml dialplan In-Reply-To: References: Message-ID: Ok i've looked at the source code for mod_dialplan and in your xml, you need something like:
wrote: > Ok, I think the whole point is that I don't undestand how the dialplan > works. > > I started from 0 ... now on the sofia.conf.xml I have this: > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > Then on the t38_transconde.xml file I have this. > > > > > > > > > > > > > > > > > I could see on the /var/log/freeswitch/freeswitch.xml.fsxml file that the > t38_transcode.xml file its included ... but still get the same error on on > the console, so something terrible wrong I'm missing here ... :-( > > Also ... it's possible to directly send the calls sofia/$1$2@$${external} > ? ... I mean, sending the call directly to a URI, with username/password > and so on ... instead of having to define a profile for it. > > > 2017-05-16 16:13 GMT+01:00 David Villasmil >: > >> No, you may be sending it to the right ports, but the profile attached to >> that port must have the context set to the correct dialplan. >> >> I just did exactly what you're doing and i also got "No Route, Aborting" >> >> I've never seen this type of "dialplan" value, tbh >> >> >> >> >> Regards, >> >> David Villasmil >> email: david.villasmil.work at gmail.com >> phone: +34669448337 <+34%20669%2044%2083%2037> >> >> On Tue, May 16, 2017 at 3:30 PM, Ra?l Alexis Betancor Santana < >> rbetancor at gmail.com> wrote: >> >>> Only one profile defined on the sofia.conf.xml and I'm sending the >>> traffict to the wright ports, if not I whould get the logs on the console, >>> as that are the only ports enabled. >>> >>> 2017-05-16 14:22 GMT+01:00 David Villasmil < >>> david.villasmil.work at gmail.com>: >>> >>>> Take a look at your profile, it should be listening on the port you're >>>> sending to, and must have the context parameter set to your dialplan name. >>>> On Tue, May 16, 2017 at 3:15 PM Ra?l Alexis Betancor Santana < >>>> rbetancor at gmail.com> wrote: >>>> >>>>> >>>>> Hi all, till now I'm been working with pre-made setup files for FS, >>>>> and now I'm trying to get a deep knowleadge of how the dialplan works. >>>>> >>>>> So I modifed my autoload_configs/sofia.conf.xml file and changed my >>>>> dialplan param to something like this: >>>>> >>>>> >>>>> >>>>> The Idea is that it loads another .xml file especific task and also >>>>> use a dialplan throught a socket to a daemon that handle the rest. >>>>> >>>>> On my t38_transcode.xml file ... very simple: >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> But If I fire a call to FS like [number]@[FS_IP] from a sip testing >>>>> client I get >>>>> >>>>> [INFO] switch_core_state_machine.c:311 No Route, Aborting >>>>> >>>>> What I'm doing wrong here? >>>>> >>>>> My target its just I want to ANY call that came in with a SIP header >>>>> of X-T39-Transcode=True (market by a sip proxy elsewhere on the net) just >>>>> do a T38->Ulaw transcoding saving the T38 trace, so I could inspect it >>>>> later. >>>>> The rest of calls coming in ... as they don't have the sip-header >>>>> should end on other app. >>>>> >>>>> For doing the testing I disabled the socket_inline part of the >>>>> dialplan string, so it have only this: >>>>> >>>>> >>>>> >>>>> Did I miss something? ... or maybe missundestood who the xml dialplan >>>>> works ? >>>>> ____________________________________________________________ >>>>> _____________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-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 >>> >> >> ? >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170516/7fcba54e/attachment-0001.html From david.villasmil.work at gmail.com Tue May 16 20:34:54 2017 From: david.villasmil.work at gmail.com (David Villasmil) Date: Tue, 16 May 2017 18:34:54 +0200 Subject: [Freeswitch-users] Noob question about xml dialplan In-Reply-To: References: Message-ID: that context line is not necessary, btw. point is you need the section:
? Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 On Tue, May 16, 2017 at 6:33 PM, David Villasmil < david.villasmil.work at gmail.com> wrote: > Ok i've looked at the source code for mod_dialplan and in your xml, you > need something like: > > > >
> > > > > > > > > > >
? > > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 <+34%20669%2044%2083%2037> > > On Tue, May 16, 2017 at 5:26 PM, Ra?l Alexis Betancor Santana < > rbetancor at gmail.com> wrote: > >> Ok, I think the whole point is that I don't undestand how the dialplan >> works. >> >> I started from 0 ... now on the sofia.conf.xml I have this: >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> Then on the t38_transconde.xml file I have this. >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> I could see on the /var/log/freeswitch/freeswitch.xml.fsxml file that >> the t38_transcode.xml file its included ... but still get the same error on >> on the console, so something terrible wrong I'm missing here ... :-( >> >> Also ... it's possible to directly send the calls sofia/$1$2@$${external} >> ? ... I mean, sending the call directly to a URI, with username/password >> and so on ... instead of having to define a profile for it. >> >> >> 2017-05-16 16:13 GMT+01:00 David Villasmil > m>: >> >>> No, you may be sending it to the right ports, but the profile attached >>> to that port must have the context set to the correct dialplan. >>> >>> I just did exactly what you're doing and i also got "No Route, Aborting" >>> >>> I've never seen this type of "dialplan" value, tbh >>> >>> >>> >>> >>> Regards, >>> >>> David Villasmil >>> email: david.villasmil.work at gmail.com >>> phone: +34669448337 <+34%20669%2044%2083%2037> >>> >>> On Tue, May 16, 2017 at 3:30 PM, Ra?l Alexis Betancor Santana < >>> rbetancor at gmail.com> wrote: >>> >>>> Only one profile defined on the sofia.conf.xml and I'm sending the >>>> traffict to the wright ports, if not I whould get the logs on the console, >>>> as that are the only ports enabled. >>>> >>>> 2017-05-16 14:22 GMT+01:00 David Villasmil < >>>> david.villasmil.work at gmail.com>: >>>> >>>>> Take a look at your profile, it should be listening on the port you're >>>>> sending to, and must have the context parameter set to your dialplan name. >>>>> On Tue, May 16, 2017 at 3:15 PM Ra?l Alexis Betancor Santana < >>>>> rbetancor at gmail.com> wrote: >>>>> >>>>>> >>>>>> Hi all, till now I'm been working with pre-made setup files for FS, >>>>>> and now I'm trying to get a deep knowleadge of how the dialplan works. >>>>>> >>>>>> So I modifed my autoload_configs/sofia.conf.xml file and changed my >>>>>> dialplan param to something like this: >>>>>> >>>>>> >>>>>> >>>>>> The Idea is that it loads another .xml file especific task and also >>>>>> use a dialplan throught a socket to a daemon that handle the rest. >>>>>> >>>>>> On my t38_transcode.xml file ... very simple: >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> But If I fire a call to FS like [number]@[FS_IP] from a sip testing >>>>>> client I get >>>>>> >>>>>> [INFO] switch_core_state_machine.c:311 No Route, Aborting >>>>>> >>>>>> What I'm doing wrong here? >>>>>> >>>>>> My target its just I want to ANY call that came in with a SIP header >>>>>> of X-T39-Transcode=True (market by a sip proxy elsewhere on the net) just >>>>>> do a T38->Ulaw transcoding saving the T38 trace, so I could inspect it >>>>>> later. >>>>>> The rest of calls coming in ... as they don't have the sip-header >>>>>> should end on other app. >>>>>> >>>>>> For doing the testing I disabled the socket_inline part of the >>>>>> dialplan string, so it have only this: >>>>>> >>>>>> >>>>>> >>>>>> Did I miss something? ... or maybe missundestood who the xml dialplan >>>>>> works ? >>>>>> ____________________________________________________________ >>>>>> _____________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-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/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 >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170516/17d53358/attachment-0001.html From mike at jerris.com Tue May 16 20:38:17 2017 From: mike at jerris.com (Michael Jerris) Date: Tue, 16 May 2017 12:38:17 -0400 Subject: [Freeswitch-users] Fax result when using mail to fax In-Reply-To: <96ce819b-f29c-a157-5cf4-13d76be1d9fb@xbipin.com> References: <15c0db7b410.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com> <591A0E8C.6070700@telefaks.de> <591AAC6A.7060609@telefaks.de> <17851f77-08bf-ca7d-4284-e943631f347f@xbipin.com> <364C278B-FB7D-4F6E-9864-05C55E945057@jerris.com> <96ce819b-f29c-a157-5cf4-13d76be1d9fb@xbipin.com> Message-ID: <5E33DE8A-C31E-4FBF-997A-02FB41103AB2@jerris.com> https://freeswitch.org/confluence/display/FREESWITCH/Lua+API+Reference#LuaAPIReference-SpecialCase:envobject > On May 16, 2017, at 11:57 AM, Bipin Patel wrote: > > hi, > > after a lot of digging figured out the issue, python script wasnt behaving properly, also managed to get the FAX report when fax was sent and then email it to sender, ill post the details soon once i have everything cleaned up. > > one last thing i wanted to know is im using a lua script which in turn calls a python script to send the mail with the FAX result but how do we print all the variable available for use, actually i want to know the dialed number where the fax was sent so can also print that in the report, im using the below in the originate. Manged to get it working > > api_hangup_hook=\'lua /home/pi/hangup.lua\' > > > Regards, > Bipin > > > -------- Original Message -------- > Subject: Re: [Freeswitch-users] Fax result when using mail to fax > From: Michael Jerris > To: FreeSWITCH Users Help > Date: 5/16/2017, 7:16:51 PM >> You should investigate in your logs to see what is really going on? I?d set the assumptions of what is going on to the side and review logs to see whats happening. >> >>> On May 16, 2017, at 10:24 AM, Bipin Patel > wrote: >>> >>> hi, >>> >>> well i was also not sure what was happening but after doing some packet traces im seeing 2 invites being generated and both run parallel so one fails later on and one transmits the fax successfully, shall i paste a debug log? >>> >>> >>> Regards, >>> Bipin >>> >>> >>> -------- Original Message -------- >>> Subject: Re: [Freeswitch-users] Fax result when using mail to fax >>> From: Michael Jerris >>> To: FreeSWITCH Users Help >>> Date: 5/16/2017, 5:41:30 PM >>>> a single originate will only create a single call. What you describe here isn't happening as you describe it. A log of the calls may tell you more. >>>> >>>> On Tue, May 16, 2017 at 5:36 AM Bipin Patel > wrote: >>>> hi, >>>> >>>> i think i have one other plan, let me try that and ill get back with the >>>> result. >>>> >>>> i have on more issue, when i send a fax using the originate command, it >>>> results in 2 invites being sent to the gateway, one sends the fax and i >>>> get report for that but the other fails so i also get a failed report >>>> for that, any idea what could be causing this, my originate command is >>>> as below >>>> >>>> originate >>>> [ignore_early_media=true,fax_verbose=true,fax_enable_t38_request=true,fax_enable_t38=true,absolute_codec_string=PCMU]sofia/gateway/ma/123466789 >>>> &txfax(\'" + att_path + ".tiff\')" >>>> >>>> >>>> -------- Original Message -------- >>>> Subject: Re: [Freeswitch-users] Fax result when using mail to fax >>>> From: Peter Steinbach > >>>> To: FreeSWITCH Users Help > >>>> Date: 5/16/2017, 11:38:18 AM >>>> >>>> > Hello Bipin, >>>> > >>>> > the fax result is only available, long after the dialplan has beeen >>>> > evaluated. >>>> > >>>> > So you will somehow have to use some event driven things. >>>> > execute_on_fax_success would have been the next thing to propose, but id >>>> > did not work in your case. >>>> > >>>> > In Linux you can use inotify. We have also used this method for getting >>>> > certain infos from the Freeswitch logfile and then to fire actions (in >>>> > our case firewall rules). Maybe this is an approach for you? >>>> > >>>> > Best regards >>>> > Peter >>>> > >>>> > On 05/16/17 07:25, Bipin Patel wrote: >>>> >> hi, >>>> >> >>>> >> well considering im running this FS on the raspberry pi for a small >>>> >> office, ESL would be like a lot just for the purpose, if the console can >>>> >> display the fax result then why isnt it possible to get the same in >>>> >> dialplan or somewhere else where i can fire a script or something and >>>> >> let the sender know what happened to their fax. >>>> >> >>>> >> i tried using execute_on_fax_success and execute_on_fax_failure but they >>>> >> dont seem to work while sending a fax, i think its for receiving only >>>> >> >>>> >> >>>> >> -------- Original Message -------- >>>> >> Subject: Re: [Freeswitch-users] Fax result when using mail to fax >>>> >> From: Peter Steinbach > >>>> >> To: FreeSWITCH Users Help > >>>> >> Date: 5/16/2017, 12:24:44 AM >>>> >> >>>> >>> Hello Bipin, >>>> >>> >>>> >>> there are some events fired via ESL, if you subscribe to them. We >>>> >>> evaluate txfaxresult event and hangup event. >>>> >>> >>>> >>> The only thing which is critcal here, is that the contents of the 2 >>>> >>> events have to be joined, in order to have all informations. And in my >>>> >>> experience txfaxresult and hangup event do not always come in the same >>>> >>> order, dependend on which side of the channels hangs up first. >>>> >>> >>>> >>> Best regards >>>> >>> Peter >>>> >>> >>>> >>> On 05/15/17 22:06, Bipin Patel wrote: >>>> >>>> Hi, >>>> >>>> >>>> >>>> I have a python script that checks a mailbox for faxes to be sent, when it >>>> >>>> finds a new mail it downloads the PDF and converts to Tiff and sends via >>>> >>>> the originate command, now the problem is I can't seem to find a way to get >>>> >>>> the fax report, nor in cdr it says what happened and the originate just >>>> >>>> says ok followed by uuid of the bridge. I'm sending the fax over a fxo >>>> >>>> gateway device. Due to it going over fxo, I have no clue if the remote fax >>>> >>>> machine answered call and the only place the fax report comes is in the >>>> >>>> console. Is there a variable or anything else that can be resent to the >>>> >>>> python script telling what the result was. I tried the execute on fax >>>> >>>> successful and fail variable but they don't seem to work. >>>> >>>> >>>> >>>> My python script generates the originate command going to loopback which >>>> >>>> goes to dial plan where the actual bridge to fxo gateway happens. >>>> >>>> >>>> >>>> How are others dealing with fax report I would love to hear. >>>> >>>> >>>> >>> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170516/adc22ebd/attachment.html From rbetancor at gmail.com Tue May 16 20:38:59 2017 From: rbetancor at gmail.com (=?UTF-8?Q?Ra=C3=BAl_Alexis_Betancor_Santana?=) Date: Tue, 16 May 2017 17:38:59 +0100 Subject: [Freeswitch-users] Noob question about xml dialplan In-Reply-To: References: Message-ID: I don't get the point of calling a user/$1 ... as there is no users registered against this FS ... I just want to use it as a T38->Ulaw gateway. 2017-05-16 17:34 GMT+01:00 David Villasmil : > that context line is not necessary, btw. point is you need the section: > > > >
> > > > > > >
> ? > > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 <+34%20669%2044%2083%2037> > > On Tue, May 16, 2017 at 6:33 PM, David Villasmil < > david.villasmil.work at gmail.com> wrote: > >> Ok i've looked at the source code for mod_dialplan and in your xml, you >> need something like: >> >> >> >>
>> >> >> >> >> >> >> >> >> >> >>
> ? >> >> Regards, >> >> David Villasmil >> email: david.villasmil.work at gmail.com >> phone: +34669448337 <+34%20669%2044%2083%2037> >> >> On Tue, May 16, 2017 at 5:26 PM, Ra?l Alexis Betancor Santana < >> rbetancor at gmail.com> wrote: >> >>> Ok, I think the whole point is that I don't undestand how the dialplan >>> works. >>> >>> I started from 0 ... now on the sofia.conf.xml I have this: >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> Then on the t38_transconde.xml file I have this. >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> I could see on the /var/log/freeswitch/freeswitch.xml.fsxml file that >>> the t38_transcode.xml file its included ... but still get the same error on >>> on the console, so something terrible wrong I'm missing here ... :-( >>> >>> Also ... it's possible to directly send the calls sofia/$1$2@$${external} >>> ? ... I mean, sending the call directly to a URI, with username/password >>> and so on ... instead of having to define a profile for it. >>> >>> >>> 2017-05-16 16:13 GMT+01:00 David Villasmil < >>> david.villasmil.work at gmail.com>: >>> >>>> No, you may be sending it to the right ports, but the profile attached >>>> to that port must have the context set to the correct dialplan. >>>> >>>> I just did exactly what you're doing and i also got "No Route, Aborting" >>>> >>>> I've never seen this type of "dialplan" value, tbh >>>> >>>> >>>> >>>> >>>> Regards, >>>> >>>> David Villasmil >>>> email: david.villasmil.work at gmail.com >>>> phone: +34669448337 <+34%20669%2044%2083%2037> >>>> >>>> On Tue, May 16, 2017 at 3:30 PM, Ra?l Alexis Betancor Santana < >>>> rbetancor at gmail.com> wrote: >>>> >>>>> Only one profile defined on the sofia.conf.xml and I'm sending the >>>>> traffict to the wright ports, if not I whould get the logs on the console, >>>>> as that are the only ports enabled. >>>>> >>>>> 2017-05-16 14:22 GMT+01:00 David Villasmil < >>>>> david.villasmil.work at gmail.com>: >>>>> >>>>>> Take a look at your profile, it should be listening on the port >>>>>> you're sending to, and must have the context parameter set to your dialplan >>>>>> name. >>>>>> On Tue, May 16, 2017 at 3:15 PM Ra?l Alexis Betancor Santana < >>>>>> rbetancor at gmail.com> wrote: >>>>>> >>>>>>> >>>>>>> Hi all, till now I'm been working with pre-made setup files for FS, >>>>>>> and now I'm trying to get a deep knowleadge of how the dialplan works. >>>>>>> >>>>>>> So I modifed my autoload_configs/sofia.conf.xml file and changed my >>>>>>> dialplan param to something like this: >>>>>>> >>>>>>> >>>>>>> >>>>>>> The Idea is that it loads another .xml file especific task and also >>>>>>> use a dialplan throught a socket to a daemon that handle the rest. >>>>>>> >>>>>>> On my t38_transcode.xml file ... very simple: >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> But If I fire a call to FS like [number]@[FS_IP] from a sip testing >>>>>>> client I get >>>>>>> >>>>>>> [INFO] switch_core_state_machine.c:311 No Route, Aborting >>>>>>> >>>>>>> What I'm doing wrong here? >>>>>>> >>>>>>> My target its just I want to ANY call that came in with a SIP header >>>>>>> of X-T39-Transcode=True (market by a sip proxy elsewhere on the net) just >>>>>>> do a T38->Ulaw transcoding saving the T38 trace, so I could inspect it >>>>>>> later. >>>>>>> The rest of calls coming in ... as they don't have the sip-header >>>>>>> should end on other app. >>>>>>> >>>>>>> For doing the testing I disabled the socket_inline part of the >>>>>>> dialplan string, so it have only this: >>>>>>> >>>>>>> >>>>>>> >>>>>>> Did I miss something? ... or maybe missundestood who the xml >>>>>>> dialplan works ? >>>>>>> ____________________________________________________________ >>>>>>> _____________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://confluence.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-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/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 >>> >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170516/5639c61d/attachment-0001.html From david.villasmil.work at gmail.com Tue May 16 20:40:18 2017 From: david.villasmil.work at gmail.com (David Villasmil) Date: Tue, 16 May 2017 18:40:18 +0200 Subject: [Freeswitch-users] Noob question about xml dialplan In-Reply-To: References: Message-ID: That was my test extension, you can replace it for whatever you need :) ? Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 On Tue, May 16, 2017 at 6:38 PM, Ra?l Alexis Betancor Santana < rbetancor at gmail.com> wrote: > I don't get the point of calling a user/$1 ... as there is no users > registered against this FS ... I just want to use it as a T38->Ulaw gateway. > > 2017-05-16 17:34 GMT+01:00 David Villasmil >: > >> that context line is not necessary, btw. point is you need the section: >> >> >> >>
>> >> >> >> >> >> >>
>> ? >> >> Regards, >> >> David Villasmil >> email: david.villasmil.work at gmail.com >> phone: +34669448337 <+34%20669%2044%2083%2037> >> >> On Tue, May 16, 2017 at 6:33 PM, David Villasmil < >> david.villasmil.work at gmail.com> wrote: >> >>> Ok i've looked at the source code for mod_dialplan and in your xml, you >>> need something like: >>> >>> >>> >>>
>>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>>
>> ? >>> >>> Regards, >>> >>> David Villasmil >>> email: david.villasmil.work at gmail.com >>> phone: +34669448337 <+34%20669%2044%2083%2037> >>> >>> On Tue, May 16, 2017 at 5:26 PM, Ra?l Alexis Betancor Santana < >>> rbetancor at gmail.com> wrote: >>> >>>> Ok, I think the whole point is that I don't undestand how the dialplan >>>> works. >>>> >>>> I started from 0 ... now on the sofia.conf.xml I have this: >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> Then on the t38_transconde.xml file I have this. >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> I could see on the /var/log/freeswitch/freeswitch.xml.fsxml file that >>>> the t38_transcode.xml file its included ... but still get the same error on >>>> on the console, so something terrible wrong I'm missing here ... :-( >>>> >>>> Also ... it's possible to directly send the calls sofia/$1$2@$${external} >>>> ? ... I mean, sending the call directly to a URI, with username/password >>>> and so on ... instead of having to define a profile for it. >>>> >>>> >>>> 2017-05-16 16:13 GMT+01:00 David Villasmil < >>>> david.villasmil.work at gmail.com>: >>>> >>>>> No, you may be sending it to the right ports, but the profile attached >>>>> to that port must have the context set to the correct dialplan. >>>>> >>>>> I just did exactly what you're doing and i also got "No Route, >>>>> Aborting" >>>>> >>>>> I've never seen this type of "dialplan" value, tbh >>>>> >>>>> >>>>> >>>>> >>>>> Regards, >>>>> >>>>> David Villasmil >>>>> email: david.villasmil.work at gmail.com >>>>> phone: +34669448337 <+34%20669%2044%2083%2037> >>>>> >>>>> On Tue, May 16, 2017 at 3:30 PM, Ra?l Alexis Betancor Santana < >>>>> rbetancor at gmail.com> wrote: >>>>> >>>>>> Only one profile defined on the sofia.conf.xml and I'm sending the >>>>>> traffict to the wright ports, if not I whould get the logs on the console, >>>>>> as that are the only ports enabled. >>>>>> >>>>>> 2017-05-16 14:22 GMT+01:00 David Villasmil < >>>>>> david.villasmil.work at gmail.com>: >>>>>> >>>>>>> Take a look at your profile, it should be listening on the port >>>>>>> you're sending to, and must have the context parameter set to your dialplan >>>>>>> name. >>>>>>> On Tue, May 16, 2017 at 3:15 PM Ra?l Alexis Betancor Santana < >>>>>>> rbetancor at gmail.com> wrote: >>>>>>> >>>>>>>> >>>>>>>> Hi all, till now I'm been working with pre-made setup files for FS, >>>>>>>> and now I'm trying to get a deep knowleadge of how the dialplan works. >>>>>>>> >>>>>>>> So I modifed my autoload_configs/sofia.conf.xml file and changed >>>>>>>> my dialplan param to something like this: >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> The Idea is that it loads another .xml file especific task and also >>>>>>>> use a dialplan throught a socket to a daemon that handle the rest. >>>>>>>> >>>>>>>> On my t38_transcode.xml file ... very simple: >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>> expresion="^(\+1|1)?([2-9]\d\d[2-9]\d{6})$"> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> But If I fire a call to FS like [number]@[FS_IP] from a sip testing >>>>>>>> client I get >>>>>>>> >>>>>>>> [INFO] switch_core_state_machine.c:311 No Route, Aborting >>>>>>>> >>>>>>>> What I'm doing wrong here? >>>>>>>> >>>>>>>> My target its just I want to ANY call that came in with a SIP >>>>>>>> header of X-T39-Transcode=True (market by a sip proxy elsewhere on the net) >>>>>>>> just do a T38->Ulaw transcoding saving the T38 trace, so I could inspect it >>>>>>>> later. >>>>>>>> The rest of calls coming in ... as they don't have the sip-header >>>>>>>> should end on other app. >>>>>>>> >>>>>>>> For doing the testing I disabled the socket_inline part of the >>>>>>>> dialplan string, so it have only this: >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> Did I miss something? ... or maybe missundestood who the xml >>>>>>>> dialplan works ? >>>>>>>> ____________________________________________________________ >>>>>>>> _____________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting at freeswitch.org >>>>>>>> http://www.freeswitchsolutions.com >>>>>>>> >>>>>>>> Official FreeSWITCH Sites >>>>>>>> http://www.freeswitch.org >>>>>>>> http://confluence.freeswitch.org >>>>>>>> http://www.cluecon.com >>>>>>>> >>>>>>>> FreeSWITCH-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/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/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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170516/73a7ab53/attachment-0001.html From rbetancor at gmail.com Tue May 16 20:48:21 2017 From: rbetancor at gmail.com (=?UTF-8?Q?Ra=C3=BAl_Alexis_Betancor_Santana?=) Date: Tue, 16 May 2017 17:48:21 +0100 Subject: [Freeswitch-users] Noob question about xml dialplan In-Reply-To: References: Message-ID: Same result ... No Route, Aborting Lets see if I found some step-by-step tutorial that explains who the dialplan works. The doc on the Wiki it's not clear enought for me. 2017-05-16 17:40 GMT+01:00 David Villasmil : > That was my test extension, you can replace it for whatever you need :) > ? > > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 <+34%20669%2044%2083%2037> > > On Tue, May 16, 2017 at 6:38 PM, Ra?l Alexis Betancor Santana < > rbetancor at gmail.com> wrote: > >> I don't get the point of calling a user/$1 ... as there is no users >> registered against this FS ... I just want to use it as a T38->Ulaw gateway. >> >> 2017-05-16 17:34 GMT+01:00 David Villasmil > m>: >> >>> that context line is not necessary, btw. point is you need the section: >>> >>> >>> >>>
>>> >>> >>> >>> >>> >>> >>>
>>> ? >>> >>> Regards, >>> >>> David Villasmil >>> email: david.villasmil.work at gmail.com >>> phone: +34669448337 <+34%20669%2044%2083%2037> >>> >>> On Tue, May 16, 2017 at 6:33 PM, David Villasmil < >>> david.villasmil.work at gmail.com> wrote: >>> >>>> Ok i've looked at the source code for mod_dialplan and in your xml, you >>>> need something like: >>>> >>>> >>>> >>>>
>>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>>
>>> ? >>>> >>>> Regards, >>>> >>>> David Villasmil >>>> email: david.villasmil.work at gmail.com >>>> phone: +34669448337 <+34%20669%2044%2083%2037> >>>> >>>> On Tue, May 16, 2017 at 5:26 PM, Ra?l Alexis Betancor Santana < >>>> rbetancor at gmail.com> wrote: >>>> >>>>> Ok, I think the whole point is that I don't undestand how the dialplan >>>>> works. >>>>> >>>>> I started from 0 ... now on the sofia.conf.xml I have this: >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> Then on the t38_transconde.xml file I have this. >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> I could see on the /var/log/freeswitch/freeswitch.xml.fsxml file that >>>>> the t38_transcode.xml file its included ... but still get the same error on >>>>> on the console, so something terrible wrong I'm missing here ... :-( >>>>> >>>>> Also ... it's possible to directly send the calls sofia/$1$2@$${external} >>>>> ? ... I mean, sending the call directly to a URI, with username/password >>>>> and so on ... instead of having to define a profile for it. >>>>> >>>>> >>>>> 2017-05-16 16:13 GMT+01:00 David Villasmil < >>>>> david.villasmil.work at gmail.com>: >>>>> >>>>>> No, you may be sending it to the right ports, but the profile >>>>>> attached to that port must have the context set to the correct dialplan. >>>>>> >>>>>> I just did exactly what you're doing and i also got "No Route, >>>>>> Aborting" >>>>>> >>>>>> I've never seen this type of "dialplan" value, tbh >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Regards, >>>>>> >>>>>> David Villasmil >>>>>> email: david.villasmil.work at gmail.com >>>>>> phone: +34669448337 <+34%20669%2044%2083%2037> >>>>>> >>>>>> On Tue, May 16, 2017 at 3:30 PM, Ra?l Alexis Betancor Santana < >>>>>> rbetancor at gmail.com> wrote: >>>>>> >>>>>>> Only one profile defined on the sofia.conf.xml and I'm sending the >>>>>>> traffict to the wright ports, if not I whould get the logs on the console, >>>>>>> as that are the only ports enabled. >>>>>>> >>>>>>> 2017-05-16 14:22 GMT+01:00 David Villasmil < >>>>>>> david.villasmil.work at gmail.com>: >>>>>>> >>>>>>>> Take a look at your profile, it should be listening on the port >>>>>>>> you're sending to, and must have the context parameter set to your dialplan >>>>>>>> name. >>>>>>>> On Tue, May 16, 2017 at 3:15 PM Ra?l Alexis Betancor Santana < >>>>>>>> rbetancor at gmail.com> wrote: >>>>>>>> >>>>>>>>> >>>>>>>>> Hi all, till now I'm been working with pre-made setup files for >>>>>>>>> FS, and now I'm trying to get a deep knowleadge of how the dialplan works. >>>>>>>>> >>>>>>>>> So I modifed my autoload_configs/sofia.conf.xml file and changed >>>>>>>>> my dialplan param to something like this: >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> The Idea is that it loads another .xml file especific task and >>>>>>>>> also use a dialplan throught a socket to a daemon that handle the rest. >>>>>>>>> >>>>>>>>> On my t38_transcode.xml file ... very simple: >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>> expresion="^(\+1|1)?([2-9]\d\d[2-9]\d{6})$"> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> But If I fire a call to FS like [number]@[FS_IP] from a sip >>>>>>>>> testing client I get >>>>>>>>> >>>>>>>>> [INFO] switch_core_state_machine.c:311 No Route, Aborting >>>>>>>>> >>>>>>>>> What I'm doing wrong here? >>>>>>>>> >>>>>>>>> My target its just I want to ANY call that came in with a SIP >>>>>>>>> header of X-T39-Transcode=True (market by a sip proxy elsewhere on the net) >>>>>>>>> just do a T38->Ulaw transcoding saving the T38 trace, so I could inspect it >>>>>>>>> later. >>>>>>>>> The rest of calls coming in ... as they don't have the sip-header >>>>>>>>> should end on other app. >>>>>>>>> >>>>>>>>> For doing the testing I disabled the socket_inline part of the >>>>>>>>> dialplan string, so it have only this: >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> Did I miss something? ... or maybe missundestood who the xml >>>>>>>>> dialplan works ? >>>>>>>>> ____________________________________________________________ >>>>>>>>> _____________ >>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>> consulting at freeswitch.org >>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>> >>>>>>>>> Official FreeSWITCH Sites >>>>>>>>> http://www.freeswitch.org >>>>>>>>> http://confluence.freeswitch.org >>>>>>>>> http://www.cluecon.com >>>>>>>>> >>>>>>>>> FreeSWITCH-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/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/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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170516/6958f8cf/attachment-0001.html From ksrigo at gmail.com Tue May 16 20:50:15 2017 From: ksrigo at gmail.com (Srigo Kana) Date: Tue, 16 May 2017 18:50:15 +0200 Subject: [Freeswitch-users] Conditional call forward In-Reply-To: References: Message-ID: Hi, Is the callforward configured on the phone? If you get 302 redirect from a phone, you can jst catch it in a dialplan and do whatever you want. Srigo Sent from my iPhone > On 13 May 2017, at 19:14, Ashwin Rath wrote: > > Hi > > I have an extension which has call forward setup BUT i would like the call forward to work only when dialed from a certain number and not from another numbers. Can this be achieved ? > > -- > Ashwin Kumar Rath > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 rbetancor at gmail.com Tue May 16 21:52:35 2017 From: rbetancor at gmail.com (=?UTF-8?Q?Ra=C3=BAl_Alexis_Betancor_Santana?=) Date: Tue, 16 May 2017 18:52:35 +0100 Subject: [Freeswitch-users] Noob question about xml dialplan In-Reply-To: References: Message-ID: Just for the shake of simplicity ... on the Asterisk world would be something like: exten => _1NXXNXXXXXX,1,Noop("T38 Gateway Call") exten => _1NXXNXXXXXX,n,Set(FAXOPT(gateway)=yes) exten => _1NXXNXXXXXX,n,Dial(SIP/${EXTEN}@proxy.server.tld) I just want to make the same with FS, but not allowing T38 passthrought at all ... just T38 (a leg) to Ulaw (b leg) 2017-05-16 17:48 GMT+01:00 Ra?l Alexis Betancor Santana : > Same result ... No Route, Aborting > > Lets see if I found some step-by-step tutorial that explains who the > dialplan works. The doc on the Wiki it's not clear enought for me. > > 2017-05-16 17:40 GMT+01:00 David Villasmil >: > >> That was my test extension, you can replace it for whatever you need :) >> ? >> >> Regards, >> >> David Villasmil >> email: david.villasmil.work at gmail.com >> phone: +34669448337 <+34%20669%2044%2083%2037> >> >> On Tue, May 16, 2017 at 6:38 PM, Ra?l Alexis Betancor Santana < >> rbetancor at gmail.com> wrote: >> >>> I don't get the point of calling a user/$1 ... as there is no users >>> registered against this FS ... I just want to use it as a T38->Ulaw gateway. >>> >>> 2017-05-16 17:34 GMT+01:00 David Villasmil < >>> david.villasmil.work at gmail.com>: >>> >>>> that context line is not necessary, btw. point is you need the section: >>>> >>>> >>>> >>>>
>>>> >>>> >>>> >>>> >>>> >>>> >>>>
>>>> ? >>>> >>>> Regards, >>>> >>>> David Villasmil >>>> email: david.villasmil.work at gmail.com >>>> phone: +34669448337 <+34%20669%2044%2083%2037> >>>> >>>> On Tue, May 16, 2017 at 6:33 PM, David Villasmil < >>>> david.villasmil.work at gmail.com> wrote: >>>> >>>>> Ok i've looked at the source code for mod_dialplan and in your xml, >>>>> you need something like: >>>>> >>>>> >>>>> >>>>>
>>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>>
>>>> ? >>>>> >>>>> Regards, >>>>> >>>>> David Villasmil >>>>> email: david.villasmil.work at gmail.com >>>>> phone: +34669448337 <+34%20669%2044%2083%2037> >>>>> >>>>> On Tue, May 16, 2017 at 5:26 PM, Ra?l Alexis Betancor Santana < >>>>> rbetancor at gmail.com> wrote: >>>>> >>>>>> Ok, I think the whole point is that I don't undestand how the >>>>>> dialplan works. >>>>>> >>>>>> I started from 0 ... now on the sofia.conf.xml I have this: >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> value="generous"/> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Then on the t38_transconde.xml file I have this. >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> I could see on the /var/log/freeswitch/freeswitch.xml.fsxml file >>>>>> that the t38_transcode.xml file its included ... but still get the same >>>>>> error on on the console, so something terrible wrong I'm missing here ... >>>>>> :-( >>>>>> >>>>>> Also ... it's possible to directly send the calls sofia/$1$2@$${external} >>>>>> ? ... I mean, sending the call directly to a URI, with username/password >>>>>> and so on ... instead of having to define a profile for it. >>>>>> >>>>>> >>>>>> 2017-05-16 16:13 GMT+01:00 David Villasmil < >>>>>> david.villasmil.work at gmail.com>: >>>>>> >>>>>>> No, you may be sending it to the right ports, but the profile >>>>>>> attached to that port must have the context set to the correct dialplan. >>>>>>> >>>>>>> I just did exactly what you're doing and i also got "No Route, >>>>>>> Aborting" >>>>>>> >>>>>>> I've never seen this type of "dialplan" value, tbh >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> Regards, >>>>>>> >>>>>>> David Villasmil >>>>>>> email: david.villasmil.work at gmail.com >>>>>>> phone: +34669448337 <+34%20669%2044%2083%2037> >>>>>>> >>>>>>> On Tue, May 16, 2017 at 3:30 PM, Ra?l Alexis Betancor Santana < >>>>>>> rbetancor at gmail.com> wrote: >>>>>>> >>>>>>>> Only one profile defined on the sofia.conf.xml and I'm sending the >>>>>>>> traffict to the wright ports, if not I whould get the logs on the console, >>>>>>>> as that are the only ports enabled. >>>>>>>> >>>>>>>> 2017-05-16 14:22 GMT+01:00 David Villasmil < >>>>>>>> david.villasmil.work at gmail.com>: >>>>>>>> >>>>>>>>> Take a look at your profile, it should be listening on the port >>>>>>>>> you're sending to, and must have the context parameter set to your dialplan >>>>>>>>> name. >>>>>>>>> On Tue, May 16, 2017 at 3:15 PM Ra?l Alexis Betancor Santana < >>>>>>>>> rbetancor at gmail.com> wrote: >>>>>>>>> >>>>>>>>>> >>>>>>>>>> Hi all, till now I'm been working with pre-made setup files for >>>>>>>>>> FS, and now I'm trying to get a deep knowleadge of how the dialplan works. >>>>>>>>>> >>>>>>>>>> So I modifed my autoload_configs/sofia.conf.xml file and changed >>>>>>>>>> my dialplan param to something like this: >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> The Idea is that it loads another .xml file especific task and >>>>>>>>>> also use a dialplan throught a socket to a daemon that handle the rest. >>>>>>>>>> >>>>>>>>>> On my t38_transcode.xml file ... very simple: >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>> expresion="^(\+1|1)?([2-9]\d\d[2-9]\d{6})$"> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> But If I fire a call to FS like [number]@[FS_IP] from a sip >>>>>>>>>> testing client I get >>>>>>>>>> >>>>>>>>>> [INFO] switch_core_state_machine.c:311 No Route, Aborting >>>>>>>>>> >>>>>>>>>> What I'm doing wrong here? >>>>>>>>>> >>>>>>>>>> My target its just I want to ANY call that came in with a SIP >>>>>>>>>> header of X-T39-Transcode=True (market by a sip proxy elsewhere on the net) >>>>>>>>>> just do a T38->Ulaw transcoding saving the T38 trace, so I could inspect it >>>>>>>>>> later. >>>>>>>>>> The rest of calls coming in ... as they don't have the sip-header >>>>>>>>>> should end on other app. >>>>>>>>>> >>>>>>>>>> For doing the testing I disabled the socket_inline part of the >>>>>>>>>> dialplan string, so it have only this: >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> Did I miss something? ... or maybe missundestood who the xml >>>>>>>>>> dialplan works ? >>>>>>>>>> ____________________________________________________________ >>>>>>>>>> _____________ >>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>> consulting at freeswitch.org >>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>> >>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>> http://www.freeswitch.org >>>>>>>>>> http://confluence.freeswitch.org >>>>>>>>>> http://www.cluecon.com >>>>>>>>>> >>>>>>>>>> FreeSWITCH-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/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/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 >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170516/759df545/attachment-0001.html From david.villasmil.work at gmail.com Tue May 16 22:01:43 2017 From: david.villasmil.work at gmail.com (David Villasmil) Date: Tue, 16 May 2017 18:01:43 +0000 Subject: [Freeswitch-users] Noob question about xml dialplan In-Reply-To: References: Message-ID: Hello, You need to start by understanding freeswitch before comparing the two, i work with both and there's just no comparison, as far as I'm concerned. When you create a profile, this profile binds to a port. In the profile config there's two params, context and diaplan (though tbh until today i never ever touched the dialplan param, as i haven't had any need to do it. If i need a simple diaplan, i just remove all xmls included in said diaplan and have just one file. Now, the "context" param, along with the includes, is what fs uses to hunt for extensions. Any file loaded from the profile which includes the context name, will be use to hunt for an extension. You haven't posted your profile, so it's hard to figure out what could be wrong, as when i tried it, it worked perfectly. I should note i commented out the context param. For the sake of simplicity, can you change your extension to answer EVERYTHING, (i.e.: "^(.*)") ? On Tue, May 16, 2017 at 7:53 PM Ra?l Alexis Betancor Santana < rbetancor at gmail.com> wrote: > Just for the shake of simplicity ... on the Asterisk world would be > something like: > > exten => _1NXXNXXXXXX,1,Noop("T38 Gateway Call") > exten => _1NXXNXXXXXX,n,Set(FAXOPT(gateway)=yes) > exten => _1NXXNXXXXXX,n,Dial(SIP/${EXTEN}@proxy.server.tld) > > I just want to make the same with FS, but not allowing T38 passthrought at > all ... just T38 (a leg) to Ulaw (b leg) > > 2017-05-16 17:48 GMT+01:00 Ra?l Alexis Betancor Santana < > rbetancor at gmail.com>: > >> Same result ... No Route, Aborting >> >> Lets see if I found some step-by-step tutorial that explains who the >> dialplan works. The doc on the Wiki it's not clear enought for me. >> >> 2017-05-16 17:40 GMT+01:00 David Villasmil < >> david.villasmil.work at gmail.com>: >> >>> That was my test extension, you can replace it for whatever you need :) >>> ? >>> >>> Regards, >>> >>> David Villasmil >>> email: david.villasmil.work at gmail.com >>> phone: +34669448337 <+34%20669%2044%2083%2037> >>> >>> On Tue, May 16, 2017 at 6:38 PM, Ra?l Alexis Betancor Santana < >>> rbetancor at gmail.com> wrote: >>> >>>> I don't get the point of calling a user/$1 ... as there is no users >>>> registered against this FS ... I just want to use it as a T38->Ulaw gateway. >>>> >>>> 2017-05-16 17:34 GMT+01:00 David Villasmil < >>>> david.villasmil.work at gmail.com>: >>>> >>>>> that context line is not necessary, btw. point is you need the section: >>>>> >>>>> >>>>> >>>>>
>>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>>
>>>>> ? >>>>> >>>>> Regards, >>>>> >>>>> David Villasmil >>>>> email: david.villasmil.work at gmail.com >>>>> phone: +34669448337 <+34%20669%2044%2083%2037> >>>>> >>>>> On Tue, May 16, 2017 at 6:33 PM, David Villasmil < >>>>> david.villasmil.work at gmail.com> wrote: >>>>> >>>>>> Ok i've looked at the source code for mod_dialplan and in your xml, >>>>>> you need something like: >>>>>> >>>>>> >>>>>> >>>>>>
>>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>
>>>>> ? >>>>>> >>>>>> Regards, >>>>>> >>>>>> David Villasmil >>>>>> email: david.villasmil.work at gmail.com >>>>>> phone: +34669448337 <+34%20669%2044%2083%2037> >>>>>> >>>>>> On Tue, May 16, 2017 at 5:26 PM, Ra?l Alexis Betancor Santana < >>>>>> rbetancor at gmail.com> wrote: >>>>>> >>>>>>> Ok, I think the whole point is that I don't undestand how the >>>>>>> dialplan works. >>>>>>> >>>>>>> I started from 0 ... now on the sofia.conf.xml I have this: >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> value="udp:$${homer_ip}:$${homer_port};hep=3;capture_id=2001"/> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> value="generous"/> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> value="$${base_dir}/conf/ssl"/> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> Then on the t38_transconde.xml file I have this. >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> expresion="^(\+1|1)?([2-9]\d\d[2-9]\d{6})$"> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> I could see on the /var/log/freeswitch/freeswitch.xml.fsxml file >>>>>>> that the t38_transcode.xml file its included ... but still get the same >>>>>>> error on on the console, so something terrible wrong I'm missing here ... >>>>>>> :-( >>>>>>> >>>>>>> Also ... it's possible to directly send the calls sofia/$1$2@$${external} >>>>>>> ? ... I mean, sending the call directly to a URI, with username/password >>>>>>> and so on ... instead of having to define a profile for it. >>>>>>> >>>>>>> >>>>>>> 2017-05-16 16:13 GMT+01:00 David Villasmil < >>>>>>> david.villasmil.work at gmail.com>: >>>>>>> >>>>>>>> No, you may be sending it to the right ports, but the profile >>>>>>>> attached to that port must have the context set to the correct dialplan. >>>>>>>> >>>>>>>> I just did exactly what you're doing and i also got "No Route, >>>>>>>> Aborting" >>>>>>>> >>>>>>>> I've never seen this type of "dialplan" value, tbh >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> Regards, >>>>>>>> >>>>>>>> David Villasmil >>>>>>>> email: david.villasmil.work at gmail.com >>>>>>>> phone: +34669448337 <+34%20669%2044%2083%2037> >>>>>>>> >>>>>>>> On Tue, May 16, 2017 at 3:30 PM, Ra?l Alexis Betancor Santana < >>>>>>>> rbetancor at gmail.com> wrote: >>>>>>>> >>>>>>>>> Only one profile defined on the sofia.conf.xml and I'm sending the >>>>>>>>> traffict to the wright ports, if not I whould get the logs on the console, >>>>>>>>> as that are the only ports enabled. >>>>>>>>> >>>>>>>>> 2017-05-16 14:22 GMT+01:00 David Villasmil < >>>>>>>>> david.villasmil.work at gmail.com>: >>>>>>>>> >>>>>>>>>> Take a look at your profile, it should be listening on the port >>>>>>>>>> you're sending to, and must have the context parameter set to your dialplan >>>>>>>>>> name. >>>>>>>>>> On Tue, May 16, 2017 at 3:15 PM Ra?l Alexis Betancor Santana < >>>>>>>>>> rbetancor at gmail.com> wrote: >>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> Hi all, till now I'm been working with pre-made setup files for >>>>>>>>>>> FS, and now I'm trying to get a deep knowleadge of how the dialplan works. >>>>>>>>>>> >>>>>>>>>>> So I modifed my autoload_configs/sofia.conf.xml file and changed >>>>>>>>>>> my dialplan param to something like this: >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>> value="XML:/etc/freeswitch/dialplans/t38_transcode.xml,inline:socket: >>>>>>>>>>> 127.0.0.1:8022 async full"/> >>>>>>>>>>> >>>>>>>>>>> The Idea is that it loads another .xml file especific task and >>>>>>>>>>> also use a dialplan throught a socket to a daemon that handle the rest. >>>>>>>>>>> >>>>>>>>>>> On my t38_transcode.xml file ... very simple: >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>> expresion="^(\+1|1)?([2-9]\d\d[2-9]\d{6})$"> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>> data="sip_execute_on_image=t38_gateway peer nocng"/> >>>>>>>>>>> >>>>>>>>>> data="sofia/external/$1$2 at proxy.server.tld"/> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> But If I fire a call to FS like [number]@[FS_IP] from a sip >>>>>>>>>>> testing client I get >>>>>>>>>>> >>>>>>>>>>> [INFO] switch_core_state_machine.c:311 No Route, Aborting >>>>>>>>>>> >>>>>>>>>>> What I'm doing wrong here? >>>>>>>>>>> >>>>>>>>>>> My target its just I want to ANY call that came in with a SIP >>>>>>>>>>> header of X-T39-Transcode=True (market by a sip proxy elsewhere on the net) >>>>>>>>>>> just do a T38->Ulaw transcoding saving the T38 trace, so I could inspect it >>>>>>>>>>> later. >>>>>>>>>>> The rest of calls coming in ... as they don't have the >>>>>>>>>>> sip-header should end on other app. >>>>>>>>>>> >>>>>>>>>>> For doing the testing I disabled the socket_inline part of the >>>>>>>>>>> dialplan string, so it have only this: >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>> value="XML:/etc/freeswitch/dialplans/t38_transcode.xml"/> >>>>>>>>>>> >>>>>>>>>>> Did I miss something? ... or maybe missundestood who the xml >>>>>>>>>>> dialplan works ? >>>>>>>>>>> >>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>> consulting at freeswitch.org >>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>> >>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>> http://confluence.freeswitch.org >>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>> >>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>> FreeSWITCH-users at 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 >>> >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170516/843a4b04/attachment-0001.html From rbetancor at gmail.com Tue May 16 22:12:08 2017 From: rbetancor at gmail.com (=?UTF-8?Q?Ra=C3=BAl_Alexis_Betancor_Santana?=) Date: Tue, 16 May 2017 19:12:08 +0100 Subject: [Freeswitch-users] Noob question about xml dialplan In-Reply-To: References: Message-ID: Thats what I'm trying ... to undestand how it works. I posted the profile I'm using a couple of messasges earlier, anyway: I changed the the extension pattern to ^(.*) ... still the same result Could you please share an example freeswitch.xml.fsxml file ? ... maybe the problem is how the includes are handled, so something is not on the xml node it needs to. I miss a command like the 'show dialplan' ... where you could check how an specific call would behalf 2017-05-16 19:01 GMT+01:00 David Villasmil : > Hello, > > You need to start by understanding freeswitch before comparing the two, i > work with both and there's just no comparison, as far as I'm concerned. > > When you create a profile, this profile binds to a port. In the profile > config there's two params, context and diaplan (though tbh until today i > never ever touched the dialplan param, as i haven't had any need to do it. > If i need a simple diaplan, i just remove all xmls included in said diaplan > and have just one file. > > Now, the "context" param, along with the includes, is what fs uses to hunt > for extensions. Any file loaded from the profile which includes the context > name, will be use to hunt for an extension. > > You haven't posted your profile, so it's hard to figure out what could be > wrong, as when i tried it, it worked perfectly. I should note i commented > out the context param. > > For the sake of simplicity, can you change your extension to answer > EVERYTHING, (i.e.: "^(.*)") ? > > > > On Tue, May 16, 2017 at 7:53 PM Ra?l Alexis Betancor Santana < > rbetancor at gmail.com> wrote: > >> Just for the shake of simplicity ... on the Asterisk world would be >> something like: >> >> exten => _1NXXNXXXXXX,1,Noop("T38 Gateway Call") >> exten => _1NXXNXXXXXX,n,Set(FAXOPT(gateway)=yes) >> exten => _1NXXNXXXXXX,n,Dial(SIP/${EXTEN}@proxy.server.tld) >> >> I just want to make the same with FS, but not allowing T38 passthrought >> at all ... just T38 (a leg) to Ulaw (b leg) >> >> 2017-05-16 17:48 GMT+01:00 Ra?l Alexis Betancor Santana < >> rbetancor at gmail.com>: >> >>> Same result ... No Route, Aborting >>> >>> Lets see if I found some step-by-step tutorial that explains who the >>> dialplan works. The doc on the Wiki it's not clear enought for me. >>> >>> 2017-05-16 17:40 GMT+01:00 David Villasmil >> com>: >>> >>>> That was my test extension, you can replace it for whatever you need :) >>>> ? >>>> >>>> Regards, >>>> >>>> David Villasmil >>>> email: david.villasmil.work at gmail.com >>>> phone: +34669448337 <+34%20669%2044%2083%2037> >>>> >>>> On Tue, May 16, 2017 at 6:38 PM, Ra?l Alexis Betancor Santana < >>>> rbetancor at gmail.com> wrote: >>>> >>>>> I don't get the point of calling a user/$1 ... as there is no users >>>>> registered against this FS ... I just want to use it as a T38->Ulaw gateway. >>>>> >>>>> 2017-05-16 17:34 GMT+01:00 David Villasmil < >>>>> david.villasmil.work at gmail.com>: >>>>> >>>>>> that context line is not necessary, btw. point is you need the >>>>>> section: >>>>>> >>>>>> >>>>>> >>>>>>
>>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>
>>>>>> ? >>>>>> >>>>>> Regards, >>>>>> >>>>>> David Villasmil >>>>>> email: david.villasmil.work at gmail.com >>>>>> phone: +34669448337 <+34%20669%2044%2083%2037> >>>>>> >>>>>> On Tue, May 16, 2017 at 6:33 PM, David Villasmil < >>>>>> david.villasmil.work at gmail.com> wrote: >>>>>> >>>>>>> Ok i've looked at the source code for mod_dialplan and in your xml, >>>>>>> you need something like: >>>>>>> >>>>>>> >>>>>>> >>>>>>>
>>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>>
>>>>>> ? >>>>>>> >>>>>>> Regards, >>>>>>> >>>>>>> David Villasmil >>>>>>> email: david.villasmil.work at gmail.com >>>>>>> phone: +34669448337 <+34%20669%2044%2083%2037> >>>>>>> >>>>>>> On Tue, May 16, 2017 at 5:26 PM, Ra?l Alexis Betancor Santana < >>>>>>> rbetancor at gmail.com> wrote: >>>>>>> >>>>>>>> Ok, I think the whole point is that I don't undestand how the >>>>>>>> dialplan works. >>>>>>>> >>>>>>>> I started from 0 ... now on the sofia.conf.xml I have this: >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>> value="generous"/> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> Then on the t38_transconde.xml file I have this. >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> I could see on the /var/log/freeswitch/freeswitch.xml.fsxml file >>>>>>>> that the t38_transcode.xml file its included ... but still get the same >>>>>>>> error on on the console, so something terrible wrong I'm missing here ... >>>>>>>> :-( >>>>>>>> >>>>>>>> Also ... it's possible to directly send the calls sofia/$1$2@$${external} >>>>>>>> ? ... I mean, sending the call directly to a URI, with username/password >>>>>>>> and so on ... instead of having to define a profile for it. >>>>>>>> >>>>>>>> >>>>>>>> 2017-05-16 16:13 GMT+01:00 David Villasmil < >>>>>>>> david.villasmil.work at gmail.com>: >>>>>>>> >>>>>>>>> No, you may be sending it to the right ports, but the profile >>>>>>>>> attached to that port must have the context set to the correct dialplan. >>>>>>>>> >>>>>>>>> I just did exactly what you're doing and i also got "No Route, >>>>>>>>> Aborting" >>>>>>>>> >>>>>>>>> I've never seen this type of "dialplan" value, tbh >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> Regards, >>>>>>>>> >>>>>>>>> David Villasmil >>>>>>>>> email: david.villasmil.work at gmail.com >>>>>>>>> phone: +34669448337 <+34%20669%2044%2083%2037> >>>>>>>>> >>>>>>>>> On Tue, May 16, 2017 at 3:30 PM, Ra?l Alexis Betancor Santana < >>>>>>>>> rbetancor at gmail.com> wrote: >>>>>>>>> >>>>>>>>>> Only one profile defined on the sofia.conf.xml and I'm sending >>>>>>>>>> the traffict to the wright ports, if not I whould get the logs on the >>>>>>>>>> console, as that are the only ports enabled. >>>>>>>>>> >>>>>>>>>> 2017-05-16 14:22 GMT+01:00 David Villasmil < >>>>>>>>>> david.villasmil.work at gmail.com>: >>>>>>>>>> >>>>>>>>>>> Take a look at your profile, it should be listening on the port >>>>>>>>>>> you're sending to, and must have the context parameter set to your dialplan >>>>>>>>>>> name. >>>>>>>>>>> On Tue, May 16, 2017 at 3:15 PM Ra?l Alexis Betancor Santana < >>>>>>>>>>> rbetancor at gmail.com> wrote: >>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> Hi all, till now I'm been working with pre-made setup files for >>>>>>>>>>>> FS, and now I'm trying to get a deep knowleadge of how the dialplan works. >>>>>>>>>>>> >>>>>>>>>>>> So I modifed my autoload_configs/sofia.conf.xml file and >>>>>>>>>>>> changed my dialplan param to something like this: >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> The Idea is that it loads another .xml file especific task and >>>>>>>>>>>> also use a dialplan throught a socket to a daemon that handle the rest. >>>>>>>>>>>> >>>>>>>>>>>> On my t38_transcode.xml file ... very simple: >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>> expresion="^(\+1|1)?([2-9]\d\d[2-9]\d{6})$"> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> But If I fire a call to FS like [number]@[FS_IP] from a sip >>>>>>>>>>>> testing client I get >>>>>>>>>>>> >>>>>>>>>>>> [INFO] switch_core_state_machine.c:311 No Route, Aborting >>>>>>>>>>>> >>>>>>>>>>>> What I'm doing wrong here? >>>>>>>>>>>> >>>>>>>>>>>> My target its just I want to ANY call that came in with a SIP >>>>>>>>>>>> header of X-T39-Transcode=True (market by a sip proxy elsewhere on the net) >>>>>>>>>>>> just do a T38->Ulaw transcoding saving the T38 trace, so I could inspect it >>>>>>>>>>>> later. >>>>>>>>>>>> The rest of calls coming in ... as they don't have the >>>>>>>>>>>> sip-header should end on other app. >>>>>>>>>>>> >>>>>>>>>>>> For doing the testing I disabled the socket_inline part of the >>>>>>>>>>>> dialplan string, so it have only this: >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> Did I miss something? ... or maybe missundestood who the xml >>>>>>>>>>>> dialplan works ? >>>>>>>>>>>> ____________________________________________________________ >>>>>>>>>>>> _____________ >>>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>>> consulting at freeswitch.org >>>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>>> >>>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>> http://confluence.freeswitch.org >>>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>>> >>>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>>> FreeSWITCH-users at 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 >>>> >>> >>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170516/180658c4/attachment-0001.html From david.villasmil.work at gmail.com Tue May 16 22:15:24 2017 From: david.villasmil.work at gmail.com (David Villasmil) Date: Tue, 16 May 2017 18:15:24 +0000 Subject: [Freeswitch-users] Noob question about xml dialplan In-Reply-To: References: Message-ID: Yeah that command doesn't exists. But there is another command, xml_locate or something like that. Remove the context param and try again. Post the latest version of the extension with the section i pointed out before. I can't send you the fsxml as I'm not in my computer right now. I'll send it in a few minutes. On Tue, May 16, 2017 at 8:12 PM Ra?l Alexis Betancor Santana < rbetancor at gmail.com> wrote: > Thats what I'm trying ... to undestand how it works. > > I posted the profile I'm using a couple of messasges earlier, anyway: > > > > > > > > > > > > > > > > > value="udp:$${homer_ip}:$${homer_port};hep=3;capture_id=2001"/> > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > I changed the the extension pattern to ^(.*) ... still the same result > > Could you please share an example freeswitch.xml.fsxml file ? ... maybe > the problem is how the includes are handled, so something is not on the xml > node it needs to. > > I miss a command like the 'show dialplan' ... where you could check how an > specific call would behalf > > > 2017-05-16 19:01 GMT+01:00 David Villasmil >: > >> Hello, >> >> You need to start by understanding freeswitch before comparing the two, i >> work with both and there's just no comparison, as far as I'm concerned. >> >> When you create a profile, this profile binds to a port. In the profile >> config there's two params, context and diaplan (though tbh until today i >> never ever touched the dialplan param, as i haven't had any need to do it. >> If i need a simple diaplan, i just remove all xmls included in said diaplan >> and have just one file. >> >> Now, the "context" param, along with the includes, is what fs uses to >> hunt for extensions. Any file loaded from the profile which includes the >> context name, will be use to hunt for an extension. >> >> You haven't posted your profile, so it's hard to figure out what could be >> wrong, as when i tried it, it worked perfectly. I should note i commented >> out the context param. >> >> For the sake of simplicity, can you change your extension to answer >> EVERYTHING, (i.e.: "^(.*)") ? >> >> >> >> On Tue, May 16, 2017 at 7:53 PM Ra?l Alexis Betancor Santana < >> rbetancor at gmail.com> wrote: >> >>> Just for the shake of simplicity ... on the Asterisk world would be >>> something like: >>> >>> exten => _1NXXNXXXXXX,1,Noop("T38 Gateway Call") >>> exten => _1NXXNXXXXXX,n,Set(FAXOPT(gateway)=yes) >>> exten => _1NXXNXXXXXX,n,Dial(SIP/${EXTEN}@proxy.server.tld) >>> >>> I just want to make the same with FS, but not allowing T38 passthrought >>> at all ... just T38 (a leg) to Ulaw (b leg) >>> >>> 2017-05-16 17:48 GMT+01:00 Ra?l Alexis Betancor Santana < >>> rbetancor at gmail.com>: >>> >>>> Same result ... No Route, Aborting >>>> >>>> Lets see if I found some step-by-step tutorial that explains who the >>>> dialplan works. The doc on the Wiki it's not clear enought for me. >>>> >>>> 2017-05-16 17:40 GMT+01:00 David Villasmil < >>>> david.villasmil.work at gmail.com>: >>>> >>>>> That was my test extension, you can replace it for whatever you need :) >>>>> ? >>>>> >>>>> Regards, >>>>> >>>>> David Villasmil >>>>> email: david.villasmil.work at gmail.com >>>>> phone: +34669448337 <+34%20669%2044%2083%2037> >>>>> >>>>> On Tue, May 16, 2017 at 6:38 PM, Ra?l Alexis Betancor Santana < >>>>> rbetancor at gmail.com> wrote: >>>>> >>>>>> I don't get the point of calling a user/$1 ... as there is no users >>>>>> registered against this FS ... I just want to use it as a T38->Ulaw gateway. >>>>>> >>>>>> 2017-05-16 17:34 GMT+01:00 David Villasmil < >>>>>> david.villasmil.work at gmail.com>: >>>>>> >>>>>>> that context line is not necessary, btw. point is you need the >>>>>>> section: >>>>>>> >>>>>>> >>>>>>> >>>>>>>
>>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>>
>>>>>>> ? >>>>>>> >>>>>>> Regards, >>>>>>> >>>>>>> David Villasmil >>>>>>> email: david.villasmil.work at gmail.com >>>>>>> phone: +34669448337 <+34%20669%2044%2083%2037> >>>>>>> >>>>>>> On Tue, May 16, 2017 at 6:33 PM, David Villasmil < >>>>>>> david.villasmil.work at gmail.com> wrote: >>>>>>> >>>>>>>> Ok i've looked at the source code for mod_dialplan and in your xml, >>>>>>>> you need something like: >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>>
>>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>>
>>>>>>> ? >>>>>>>> >>>>>>>> Regards, >>>>>>>> >>>>>>>> David Villasmil >>>>>>>> email: david.villasmil.work at gmail.com >>>>>>>> phone: +34669448337 <+34%20669%2044%2083%2037> >>>>>>>> >>>>>>>> On Tue, May 16, 2017 at 5:26 PM, Ra?l Alexis Betancor Santana < >>>>>>>> rbetancor at gmail.com> wrote: >>>>>>>> >>>>>>>>> Ok, I think the whole point is that I don't undestand how the >>>>>>>>> dialplan works. >>>>>>>>> >>>>>>>>> I started from 0 ... now on the sofia.conf.xml I have this: >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>> value="udp:$${homer_ip}:$${homer_port};hep=3;capture_id=2001"/> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>> value="generous"/> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>> value="$${base_dir}/conf/ssl"/> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> Then on the t38_transconde.xml file I have this. >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>> expresion="^(\+1|1)?([2-9]\d\d[2-9]\d{6})$"> >>>>>>>>> >>>>>>>>> >>>>>>>> data="sip_execute_on_image=t38_gateway peer nocng"/> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> I could see on the /var/log/freeswitch/freeswitch.xml.fsxml file >>>>>>>>> that the t38_transcode.xml file its included ... but still get the same >>>>>>>>> error on on the console, so something terrible wrong I'm missing here ... >>>>>>>>> :-( >>>>>>>>> >>>>>>>>> Also ... it's possible to directly send the calls sofia/$1$2@$${external} >>>>>>>>> ? ... I mean, sending the call directly to a URI, with username/password >>>>>>>>> and so on ... instead of having to define a profile for it. >>>>>>>>> >>>>>>>>> >>>>>>>>> 2017-05-16 16:13 GMT+01:00 David Villasmil < >>>>>>>>> david.villasmil.work at gmail.com>: >>>>>>>>> >>>>>>>>>> No, you may be sending it to the right ports, but the profile >>>>>>>>>> attached to that port must have the context set to the correct dialplan. >>>>>>>>>> >>>>>>>>>> I just did exactly what you're doing and i also got "No Route, >>>>>>>>>> Aborting" >>>>>>>>>> >>>>>>>>>> I've never seen this type of "dialplan" value, tbh >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> Regards, >>>>>>>>>> >>>>>>>>>> David Villasmil >>>>>>>>>> email: david.villasmil.work at gmail.com >>>>>>>>>> phone: +34669448337 <+34%20669%2044%2083%2037> >>>>>>>>>> >>>>>>>>>> On Tue, May 16, 2017 at 3:30 PM, Ra?l Alexis Betancor Santana < >>>>>>>>>> rbetancor at gmail.com> wrote: >>>>>>>>>> >>>>>>>>>>> Only one profile defined on the sofia.conf.xml and I'm sending >>>>>>>>>>> the traffict to the wright ports, if not I whould get the logs on the >>>>>>>>>>> console, as that are the only ports enabled. >>>>>>>>>>> >>>>>>>>>>> 2017-05-16 14:22 GMT+01:00 David Villasmil < >>>>>>>>>>> david.villasmil.work at gmail.com>: >>>>>>>>>>> >>>>>>>>>>>> Take a look at your profile, it should be listening on the port >>>>>>>>>>>> you're sending to, and must have the context parameter set to your dialplan >>>>>>>>>>>> name. >>>>>>>>>>>> On Tue, May 16, 2017 at 3:15 PM Ra?l Alexis Betancor Santana < >>>>>>>>>>>> rbetancor at gmail.com> wrote: >>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> Hi all, till now I'm been working with pre-made setup files >>>>>>>>>>>>> for FS, and now I'm trying to get a deep knowleadge of how the dialplan >>>>>>>>>>>>> works. >>>>>>>>>>>>> >>>>>>>>>>>>> So I modifed my autoload_configs/sofia.conf.xml file and >>>>>>>>>>>>> changed my dialplan param to something like this: >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>> value="XML:/etc/freeswitch/dialplans/t38_transcode.xml,inline:socket: >>>>>>>>>>>>> 127.0.0.1:8022 async full"/> >>>>>>>>>>>>> >>>>>>>>>>>>> The Idea is that it loads another .xml file especific task and >>>>>>>>>>>>> also use a dialplan throught a socket to a daemon that handle the rest. >>>>>>>>>>>>> >>>>>>>>>>>>> On my t38_transcode.xml file ... very simple: >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>> expresion="^(\+1|1)?([2-9]\d\d[2-9]\d{6})$"> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>> data="sip_execute_on_image=t38_gateway peer nocng"/> >>>>>>>>>>>>> >>>>>>>>>>>> data="sofia/external/$1$2 at proxy.server.tld"/> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> But If I fire a call to FS like [number]@[FS_IP] from a sip >>>>>>>>>>>>> testing client I get >>>>>>>>>>>>> >>>>>>>>>>>>> [INFO] switch_core_state_machine.c:311 No Route, Aborting >>>>>>>>>>>>> >>>>>>>>>>>>> What I'm doing wrong here? >>>>>>>>>>>>> >>>>>>>>>>>>> My target its just I want to ANY call that came in with a SIP >>>>>>>>>>>>> header of X-T39-Transcode=True (market by a sip proxy elsewhere on the net) >>>>>>>>>>>>> just do a T38->Ulaw transcoding saving the T38 trace, so I could inspect it >>>>>>>>>>>>> later. >>>>>>>>>>>>> The rest of calls coming in ... as they don't have the >>>>>>>>>>>>> sip-header should end on other app. >>>>>>>>>>>>> >>>>>>>>>>>>> For doing the testing I disabled the socket_inline part of the >>>>>>>>>>>>> dialplan string, so it have only this: >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>> value="XML:/etc/freeswitch/dialplans/t38_transcode.xml"/> >>>>>>>>>>>>> >>>>>>>>>>>>> Did I miss something? ... or maybe missundestood who the xml >>>>>>>>>>>>> dialplan works ? >>>>>>>>>>>>> >>>>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>>>> consulting at freeswitch.org >>>>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>>>> >>>>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>>> http://confluence.freeswitch.org >>>>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>>>> >>>>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>>>> FreeSWITCH-users at 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 >>>>> >>>> >>>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170516/363cb2d4/attachment-0001.html From john.nash778 at gmail.com Tue May 16 22:42:54 2017 From: john.nash778 at gmail.com (John Nash) Date: Wed, 17 May 2017 00:12:54 +0530 Subject: [Freeswitch-users] Limit module using hiredis Message-ID: I am using limit module as per https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+Limit and using it to limit number of concurrent calls for a given resource (Unique ID for a group of Source IPs). I am using hiredis module and my redis server is on another box. It all seemed to run fine when I implemented but after few days of running (Freeswitch was restarted but not redis server) I saw that redis entries have minus values against keys. In what condition they will cause entries to be negative?. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170517/909c8804/attachment.html From david.villasmil.work at gmail.com Tue May 16 22:48:13 2017 From: david.villasmil.work at gmail.com (David Villasmil) Date: Tue, 16 May 2017 20:48:13 +0200 Subject: [Freeswitch-users] Noob question about xml dialplan In-Reply-To: References: Message-ID: "xml_locate dialplan" will give you the whole xml for the dialplan "xml_locate configuration configuration name sofia.conf" will give you your sofia.conf ? Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 On Tue, May 16, 2017 at 8:15 PM, David Villasmil < david.villasmil.work at gmail.com> wrote: > Yeah that command doesn't exists. But there is another command, xml_locate > or something like that. > Remove the context param and try again. Post the latest version of the > extension with the section i pointed out before. > > I can't send you the fsxml as I'm not in my computer right now. I'll send > it in a few minutes. > > On Tue, May 16, 2017 at 8:12 PM Ra?l Alexis Betancor Santana < > rbetancor at gmail.com> wrote: > >> Thats what I'm trying ... to undestand how it works. >> >> I posted the profile I'm using a couple of messasges earlier, anyway: >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> I changed the the extension pattern to ^(.*) ... still the same result >> >> Could you please share an example freeswitch.xml.fsxml file ? ... maybe >> the problem is how the includes are handled, so something is not on the xml >> node it needs to. >> >> I miss a command like the 'show dialplan' ... where you could check how >> an specific call would behalf >> >> >> 2017-05-16 19:01 GMT+01:00 David Villasmil > com>: >> >>> Hello, >>> >>> You need to start by understanding freeswitch before comparing the two, >>> i work with both and there's just no comparison, as far as I'm concerned. >>> >>> When you create a profile, this profile binds to a port. In the profile >>> config there's two params, context and diaplan (though tbh until today i >>> never ever touched the dialplan param, as i haven't had any need to do it. >>> If i need a simple diaplan, i just remove all xmls included in said diaplan >>> and have just one file. >>> >>> Now, the "context" param, along with the includes, is what fs uses to >>> hunt for extensions. Any file loaded from the profile which includes the >>> context name, will be use to hunt for an extension. >>> >>> You haven't posted your profile, so it's hard to figure out what could >>> be wrong, as when i tried it, it worked perfectly. I should note i >>> commented out the context param. >>> >>> For the sake of simplicity, can you change your extension to answer >>> EVERYTHING, (i.e.: "^(.*)") ? >>> >>> >>> >>> On Tue, May 16, 2017 at 7:53 PM Ra?l Alexis Betancor Santana < >>> rbetancor at gmail.com> wrote: >>> >>>> Just for the shake of simplicity ... on the Asterisk world would be >>>> something like: >>>> >>>> exten => _1NXXNXXXXXX,1,Noop("T38 Gateway Call") >>>> exten => _1NXXNXXXXXX,n,Set(FAXOPT(gateway)=yes) >>>> exten => _1NXXNXXXXXX,n,Dial(SIP/${EXTEN}@proxy.server.tld) >>>> >>>> I just want to make the same with FS, but not allowing T38 passthrought >>>> at all ... just T38 (a leg) to Ulaw (b leg) >>>> >>>> 2017-05-16 17:48 GMT+01:00 Ra?l Alexis Betancor Santana < >>>> rbetancor at gmail.com>: >>>> >>>>> Same result ... No Route, Aborting >>>>> >>>>> Lets see if I found some step-by-step tutorial that explains who the >>>>> dialplan works. The doc on the Wiki it's not clear enought for me. >>>>> >>>>> 2017-05-16 17:40 GMT+01:00 David Villasmil < >>>>> david.villasmil.work at gmail.com>: >>>>> >>>>>> That was my test extension, you can replace it for whatever you need >>>>>> :) >>>>>> ? >>>>>> >>>>>> Regards, >>>>>> >>>>>> David Villasmil >>>>>> email: david.villasmil.work at gmail.com >>>>>> phone: +34669448337 <+34%20669%2044%2083%2037> >>>>>> >>>>>> On Tue, May 16, 2017 at 6:38 PM, Ra?l Alexis Betancor Santana < >>>>>> rbetancor at gmail.com> wrote: >>>>>> >>>>>>> I don't get the point of calling a user/$1 ... as there is no users >>>>>>> registered against this FS ... I just want to use it as a T38->Ulaw gateway. >>>>>>> >>>>>>> 2017-05-16 17:34 GMT+01:00 David Villasmil < >>>>>>> david.villasmil.work at gmail.com>: >>>>>>> >>>>>>>> that context line is not necessary, btw. point is you need the >>>>>>>> section: >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>>
>>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>>
>>>>>>>> ? >>>>>>>> >>>>>>>> Regards, >>>>>>>> >>>>>>>> David Villasmil >>>>>>>> email: david.villasmil.work at gmail.com >>>>>>>> phone: +34669448337 <+34%20669%2044%2083%2037> >>>>>>>> >>>>>>>> On Tue, May 16, 2017 at 6:33 PM, David Villasmil < >>>>>>>> david.villasmil.work at gmail.com> wrote: >>>>>>>> >>>>>>>>> Ok i've looked at the source code for mod_dialplan and in your >>>>>>>>> xml, you need something like: >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>>
>>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>>
>>>>>>>> ? >>>>>>>>> >>>>>>>>> Regards, >>>>>>>>> >>>>>>>>> David Villasmil >>>>>>>>> email: david.villasmil.work at gmail.com >>>>>>>>> phone: +34669448337 <+34%20669%2044%2083%2037> >>>>>>>>> >>>>>>>>> On Tue, May 16, 2017 at 5:26 PM, Ra?l Alexis Betancor Santana < >>>>>>>>> rbetancor at gmail.com> wrote: >>>>>>>>> >>>>>>>>>> Ok, I think the whole point is that I don't undestand how the >>>>>>>>>> dialplan works. >>>>>>>>>> >>>>>>>>>> I started from 0 ... now on the sofia.conf.xml I have this: >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>> value="udp:$${homer_ip}:$${homer_port};hep=3;capture_id=2001"/> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>> value="localnet.auto"/> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>> value="generous"/> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>> value="false"/> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>> value="$${base_dir}/conf/ssl"/> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> Then on the t38_transconde.xml file I have this. >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>> expresion="^(\+1|1)?([2-9]\d\d[2-9]\d{6})$"> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> I could see on the /var/log/freeswitch/freeswitch.xml.fsxml file >>>>>>>>>> that the t38_transcode.xml file its included ... but still get the same >>>>>>>>>> error on on the console, so something terrible wrong I'm missing here ... >>>>>>>>>> :-( >>>>>>>>>> >>>>>>>>>> Also ... it's possible to directly send the calls sofia/$1$2@$${external} >>>>>>>>>> ? ... I mean, sending the call directly to a URI, with username/password >>>>>>>>>> and so on ... instead of having to define a profile for it. >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> 2017-05-16 16:13 GMT+01:00 David Villasmil < >>>>>>>>>> david.villasmil.work at gmail.com>: >>>>>>>>>> >>>>>>>>>>> No, you may be sending it to the right ports, but the profile >>>>>>>>>>> attached to that port must have the context set to the correct dialplan. >>>>>>>>>>> >>>>>>>>>>> I just did exactly what you're doing and i also got "No Route, >>>>>>>>>>> Aborting" >>>>>>>>>>> >>>>>>>>>>> I've never seen this type of "dialplan" value, tbh >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> Regards, >>>>>>>>>>> >>>>>>>>>>> David Villasmil >>>>>>>>>>> email: david.villasmil.work at gmail.com >>>>>>>>>>> phone: +34669448337 <+34%20669%2044%2083%2037> >>>>>>>>>>> >>>>>>>>>>> On Tue, May 16, 2017 at 3:30 PM, Ra?l Alexis Betancor Santana < >>>>>>>>>>> rbetancor at gmail.com> wrote: >>>>>>>>>>> >>>>>>>>>>>> Only one profile defined on the sofia.conf.xml and I'm sending >>>>>>>>>>>> the traffict to the wright ports, if not I whould get the logs on the >>>>>>>>>>>> console, as that are the only ports enabled. >>>>>>>>>>>> >>>>>>>>>>>> 2017-05-16 14:22 GMT+01:00 David Villasmil < >>>>>>>>>>>> david.villasmil.work at gmail.com>: >>>>>>>>>>>> >>>>>>>>>>>>> Take a look at your profile, it should be listening on the >>>>>>>>>>>>> port you're sending to, and must have the context parameter set to your >>>>>>>>>>>>> dialplan name. >>>>>>>>>>>>> On Tue, May 16, 2017 at 3:15 PM Ra?l Alexis Betancor Santana < >>>>>>>>>>>>> rbetancor at gmail.com> wrote: >>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> Hi all, till now I'm been working with pre-made setup files >>>>>>>>>>>>>> for FS, and now I'm trying to get a deep knowleadge of how the dialplan >>>>>>>>>>>>>> works. >>>>>>>>>>>>>> >>>>>>>>>>>>>> So I modifed my autoload_configs/sofia.conf.xml file and >>>>>>>>>>>>>> changed my dialplan param to something like this: >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> The Idea is that it loads another .xml file especific task >>>>>>>>>>>>>> and also use a dialplan throught a socket to a daemon that handle the rest. >>>>>>>>>>>>>> >>>>>>>>>>>>>> On my t38_transcode.xml file ... very simple: >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>> expresion="^(\+1|1)?([2-9]\d\d[2-9]\d{6})$"> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> But If I fire a call to FS like [number]@[FS_IP] from a sip >>>>>>>>>>>>>> testing client I get >>>>>>>>>>>>>> >>>>>>>>>>>>>> [INFO] switch_core_state_machine.c:311 No Route, Aborting >>>>>>>>>>>>>> >>>>>>>>>>>>>> What I'm doing wrong here? >>>>>>>>>>>>>> >>>>>>>>>>>>>> My target its just I want to ANY call that came in with a SIP >>>>>>>>>>>>>> header of X-T39-Transcode=True (market by a sip proxy elsewhere on the net) >>>>>>>>>>>>>> just do a T38->Ulaw transcoding saving the T38 trace, so I could inspect it >>>>>>>>>>>>>> later. >>>>>>>>>>>>>> The rest of calls coming in ... as they don't have the >>>>>>>>>>>>>> sip-header should end on other app. >>>>>>>>>>>>>> >>>>>>>>>>>>>> For doing the testing I disabled the socket_inline part of >>>>>>>>>>>>>> the dialplan string, so it have only this: >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> Did I miss something? ... or maybe missundestood who the xml >>>>>>>>>>>>>> dialplan works ? >>>>>>>>>>>>>> ____________________________________________________________ >>>>>>>>>>>>>> _____________ >>>>>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>>>>> consulting at freeswitch.org >>>>>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>>>>> >>>>>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>>>> http://confluence.freeswitch.org >>>>>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>>>>> >>>>>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>>>>> FreeSWITCH-users at 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 >>>>>> >>>>> >>>>> >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170516/c4c7d0c6/attachment-0001.html From hunterj91 at hotmail.com Tue May 16 23:53:55 2017 From: hunterj91 at hotmail.com (Jonathan Hunter) Date: Tue, 16 May 2017 19:53:55 +0000 Subject: [Freeswitch-users] voicemail_ivr examples when checking voicemail In-Reply-To: References: <719512DD-9FEF-45BB-9E25-058B33BD278D@jerris.com>, , , Message-ID: Hi Michael, Do you know how I can stop the following being called; I just dont want caller to have this option. Thanks Jon ________________________________ From: Jonathan Hunter Sent: 12 May 2017 09:25 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] voicemail_ivr examples when checking voicemail Hi Michael, Thanks for the response, so I can just exclude phrases in; etc/freeswitch/lang/en/vm/sounds.xml to stop it reading out particular options, as essentially I dont want the user ringing in to check their voicemail to not have options to record multiple greetings, just want them to be able to record one, thats only change I need ? Jon ---------- Forwarded message ---------- From: Michael Jerris > Date: Wed, May 10, 2017 at 7:49 PM Subject: Re: [Freeswitch-users] voicemail_ivr examples when checking voicemail To: FreeSWITCH Users Help > On May 10, 2017, at 2:34 PM, Jonathan Hunter > wrote: Hey guys, Sorry for the noise! I am looking to use voicemail_ivr so I can ensure my own phrases are played and I can avoid giving the user options to record just one voicemail message (as they have the option to upload a file from our portal too). You can do all this with regular voicemail app by modifying the phrases involved and options I think. No need to do your own completely. Take a look at the phrase configs used for voicemail and voicemail config file and let us know where it might be an issue. Can this be achieved using voicemail_ivr and if so are there any examples as confluence is good but doesnt have many configuration examples if you want to miss out phrases etc. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Jonathan Hunter Technical Director /Telephony Developer M:(+44) 7917 190 438 Email:jhunter at voxboxcoms.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170516/709cc26c/attachment.html From mike at jerris.com Wed May 17 00:01:08 2017 From: mike at jerris.com (Michael Jerris) Date: Tue, 16 May 2017 16:01:08 -0400 Subject: [Freeswitch-users] voicemail_ivr examples when checking voicemail In-Reply-To: References: <719512DD-9FEF-45BB-9E25-058B33BD278D@jerris.com> Message-ID: you can just change the phrase to do whatever you want, or nothing at all. > On May 16, 2017, at 3:53 PM, Jonathan Hunter wrote: > > Hi Michael, > > Do you know how I can stop the following being called; > > > > > > > > > > I just dont want caller to have this option. > > Thanks > > Jon > > > From: Jonathan Hunter > > Sent: 12 May 2017 09:25 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] voicemail_ivr examples when checking voicemail > > Hi Michael, > > Thanks for the response, so I can just exclude phrases in; > > etc/freeswitch/lang/en/vm/sounds.xml to stop it reading out particular options, as essentially I dont want the user ringing in to check their voicemail to not have options to record multiple greetings, just want them to be able to record one, thats only change I need ? > > Jon > ---------- Forwarded message ---------- > From: Michael Jerris > > Date: Wed, May 10, 2017 at 7:49 PM > Subject: Re: [Freeswitch-users] voicemail_ivr examples when checking voicemail > To: FreeSWITCH Users Help > > > > >> On May 10, 2017, at 2:34 PM, Jonathan Hunter > wrote: >> >> Hey guys, >> >> Sorry for the noise! >> >> I am looking to use voicemail_ivr so I can ensure my own phrases are played and I can avoid giving the user options to record just one voicemail message (as they have the option to upload a file from our portal too). > > You can do all this with regular voicemail app by modifying the phrases involved and options I think. No need to do your own completely. Take a look at the phrase configs used for voicemail and voicemail config file and let us know where it might be an issue. > >> >> Can this be achieved using voicemail_ivr and if so are there any examples as confluence is good but doesnt have many configuration examples if you want to miss out phrases etc. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > Jonathan Hunter > Technical Director /Telephony Developer > M:(+44) 7917 190 438 > Email:jhunter at voxboxcoms.co.uk > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170516/3a0d1c2a/attachment-0001.html From mike at jerris.com Wed May 17 00:05:30 2017 From: mike at jerris.com (Michael Jerris) Date: Tue, 16 May 2017 16:05:30 -0400 Subject: [Freeswitch-users] voicemail_ivr examples when checking voicemail In-Reply-To: References: <719512DD-9FEF-45BB-9E25-058B33BD278D@jerris.com> Message-ID: you?ll have to adjust voicemail_config_menu too? Could also with some modifications to code and the phrases just make it so you can not set or set to blank the choose/record greeting keys that would skip the one you are talking about automatically, and you?d need to do something to sort the config_menu macro > On May 16, 2017, at 4:01 PM, Michael Jerris wrote: > > you can just change the phrase to do whatever you want, or nothing at all. > >> On May 16, 2017, at 3:53 PM, Jonathan Hunter > wrote: >> >> Hi Michael, >> >> Do you know how I can stop the following being called; >> >> >> >> >> >> >> >> >> >> I just dont want caller to have this option. >> >> Thanks >> >> Jon >> >> >> From: Jonathan Hunter > >> Sent: 12 May 2017 09:25 >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] voicemail_ivr examples when checking voicemail >> >> Hi Michael, >> >> Thanks for the response, so I can just exclude phrases in; >> >> etc/freeswitch/lang/en/vm/sounds.xml to stop it reading out particular options, as essentially I dont want the user ringing in to check their voicemail to not have options to record multiple greetings, just want them to be able to record one, thats only change I need ? >> >> Jon >> ---------- Forwarded message ---------- >> From: Michael Jerris > >> Date: Wed, May 10, 2017 at 7:49 PM >> Subject: Re: [Freeswitch-users] voicemail_ivr examples when checking voicemail >> To: FreeSWITCH Users Help > >> >> >> >>> On May 10, 2017, at 2:34 PM, Jonathan Hunter > wrote: >>> >>> Hey guys, >>> >>> Sorry for the noise! >>> >>> I am looking to use voicemail_ivr so I can ensure my own phrases are played and I can avoid giving the user options to record just one voicemail message (as they have the option to upload a file from our portal too). >> >> You can do all this with regular voicemail app by modifying the phrases involved and options I think. No need to do your own completely. Take a look at the phrase configs used for voicemail and voicemail config file and let us know where it might be an issue. >> >>> >>> Can this be achieved using voicemail_ivr and if so are there any examples as confluence is good but doesnt have many configuration examples if you want to miss out phrases etc. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> -- >> Jonathan Hunter >> Technical Director /Telephony Developer >> M:(+44) 7917 190 438 >> Email:jhunter at voxboxcoms.co.uk >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170516/e04f6f5f/attachment-0001.html From brian at freeswitch.org Wed May 17 00:37:46 2017 From: brian at freeswitch.org (Brian West) Date: Tue, 16 May 2017 15:37:46 -0500 Subject: [Freeswitch-users] T.38 in new FS versions In-Reply-To: <81c86120-7679-4357-55a3-a66bf009870f@yu-boot.ru> References: <15c0db7b410.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com> <81c86120-7679-4357-55a3-a66bf009870f@yu-boot.ru> Message-ID: This hasn't changed really, the only difference now is we don't blindly accept reinvites unless you specifically enable T.38. What kind of problem are you experiencing? /b On Tue, May 16, 2017 at 7:31 AM, Yu Boot wrote: > Hi there. How exactly to enable "classic" T.38 faxes with reINVITE in > new FS versions? Not transcoding, not fax-to-email etc. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Book a phone call (CST) Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170516/435be148/attachment.html From david.villasmil.work at gmail.com Wed May 17 02:28:51 2017 From: david.villasmil.work at gmail.com (David Villasmil) Date: Tue, 16 May 2017 22:28:51 +0000 Subject: [Freeswitch-users] Change the signaling port after the ACK In-Reply-To: References: <1dc9b4a3-14a5-30e0-afde-4633000f4b4e@airenetworks.es> <4dcb448a-f77a-ffda-e824-66dcdfbcb6ae@airenetworks.es> Message-ID: I wonder on what case would it be required? (Never used it myself) On Tue, May 16, 2017 at 3:46 PM Michael Jerris wrote: > you most likely never want aggressive nat detection on. Turning it on in > most situations is a sign you don't have other nat config correct, or > simply thought it might be good. It should not be used unless absolutely > required, which it rarely is. > > On Tue, May 16, 2017 at 9:40 AM Joaquin Peral > wrote: > >> FS assumes that the user is doing NAT. >> >> Solved with >> >> >> >> in a new profile. >> >> Thanks! >> >> >> On 15/05/17 17:11, Michael Jerris wrote: >> >> I?d have to see the sip trace to answer this better? From the ladder >> diagram, i don?t see the ack looking any difference than the invite. The >> BYE is sent I assume to exactly where the contact tells us to send it to. >> >> >> On May 12, 2017, at 10:46 AM, Joaquin Peral >> wrote: >> >> Hi all, I have a special case, a user with a Cisco system. He uses the >> standard 5060 port but uses NAT. Everything is correct until send the ACK >> signaling and the port changes. Any ideas? >> >> ? >> >> external cfg: >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> -- >> [Joaquin Peral Cascales] >> Departamento de Telefon?a >> 911090048 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 >> >> >> -- >> [Joaquin Peral Cascales] >> Departamento de Telefon?a >> 911090048 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170516/42cab4f3/attachment-0001.html From bipin at xbipin.com Wed May 17 10:23:29 2017 From: bipin at xbipin.com (Bipin Patel) Date: Wed, 17 May 2017 10:23:29 +0400 Subject: [Freeswitch-users] Fax result when using mail to fax In-Reply-To: <5E33DE8A-C31E-4FBF-997A-02FB41103AB2@jerris.com> References: <15c0db7b410.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com> <591A0E8C.6070700@telefaks.de> <591AAC6A.7060609@telefaks.de> <17851f77-08bf-ca7d-4284-e943631f347f@xbipin.com> <364C278B-FB7D-4F6E-9864-05C55E945057@jerris.com> <96ce819b-f29c-a157-5cf4-13d76be1d9fb@xbipin.com> <5E33DE8A-C31E-4FBF-997A-02FB41103AB2@jerris.com> Message-ID: <306da264-0765-c205-f288-34b2c69b2763@xbipin.com> hi, finally got the fax report to work, heres more info for those willing to do it without using any ESL stuff first in the originate command include the below (change the script path and name to whatever u use, this will be called when the fax session is over): api_hangup_hook=lua /home/pi/hangup.lua in your hangup.lua script u can get the spandsp variables as below: fax_success = env:getHeader("fax_success") fax_result_code = env:getHeader("fax_result_code") fax_result_text = env:getHeader("fax_result_text") these i just pass them to a python script where i send the mail to sender with a properly formatted fax report letting him know if it was successful or not Regards, Bipin ------------------------------------------------------------------------ -------- Original Message -------- Subject: Re: [Freeswitch-users] Fax result when using mail to fax From: Michael Jerris To: FreeSWITCH Users Help Date: 5/16/2017, 8:38:17 PM > https://freeswitch.org/confluence/display/FREESWITCH/Lua+API+Reference#LuaAPIReference-SpecialCase:envobject > > >> On May 16, 2017, at 11:57 AM, Bipin Patel > > wrote: >> >> hi, >> >> after a lot of digging figured out the issue, python script wasnt >> behaving properly, also managed to get the FAX report when fax was >> sent and then email it to sender, ill post the details soon once i >> have everything cleaned up. >> >> one last thing i wanted to know is im using a lua script which in >> turn calls a python script to send the mail with the FAX result but >> how do we print all the variable available for use, actually i want >> to know the dialed number where the fax was sent so can also print >> that in the report, im using the below in the originate. Manged to >> get it working >> >> api_hangup_hook=\'lua /home/pi/hangup.lua\' >> >> >> Regards, >> Bipin >> >> >> ------------------------------------------------------------------------ >> -------- Original Message -------- >> Subject: Re: [Freeswitch-users] Fax result when using mail to fax >> From: Michael Jerris >> To: FreeSWITCH Users Help >> Date: 5/16/2017, 7:16:51 PM >>> You should investigate in your logs to see what is really going on? >>> I?d set the assumptions of what is going on to the side and review >>> logs to see whats happening. >>> >>>> On May 16, 2017, at 10:24 AM, Bipin Patel >>> > wrote: >>>> >>>> hi, >>>> >>>> well i was also not sure what was happening but after doing some >>>> packet traces im seeing 2 invites being generated and both run >>>> parallel so one fails later on and one transmits the fax >>>> successfully, shall i paste a debug log? >>>> >>>> >>>> Regards, >>>> Bipin >>>> >>>> >>>> ------------------------------------------------------------------------ >>>> -------- Original Message -------- >>>> Subject: Re: [Freeswitch-users] Fax result when using mail to fax >>>> From: Michael Jerris >>>> To: FreeSWITCH Users Help >>>> Date: 5/16/2017, 5:41:30 PM >>>>> a single originate will only create a single call. What you >>>>> describe here isn't happening as you describe it. A log of the >>>>> calls may tell you more. >>>>> >>>>> On Tue, May 16, 2017 at 5:36 AM Bipin Patel >>>> > wrote: >>>>> >>>>> hi, >>>>> >>>>> i think i have one other plan, let me try that and ill get >>>>> back with the >>>>> result. >>>>> >>>>> i have on more issue, when i send a fax using the originate >>>>> command, it >>>>> results in 2 invites being sent to the gateway, one sends the >>>>> fax and i >>>>> get report for that but the other fails so i also get a failed >>>>> report >>>>> for that, any idea what could be causing this, my originate >>>>> command is >>>>> as below >>>>> >>>>> originate >>>>> [ignore_early_media=true,fax_verbose=true,fax_enable_t38_request=true,fax_enable_t38=true,absolute_codec_string=PCMU]sofia/gateway/ma/123466789 >>>>> &txfax(\'" + att_path + ".tiff\')" >>>>> >>>>> >>>>> -------- Original Message -------- >>>>> Subject: Re: [Freeswitch-users] Fax result when using mail to fax >>>>> From: Peter Steinbach >>>> > >>>>> To: FreeSWITCH Users Help >>>>> >>>> > >>>>> Date: 5/16/2017, 11:38:18 AM >>>>> >>>>> > Hello Bipin, >>>>> > >>>>> > the fax result is only available, long after the dialplan >>>>> has beeen >>>>> > evaluated. >>>>> > >>>>> > So you will somehow have to use some event driven things. >>>>> > execute_on_fax_success would have been the next thing to >>>>> propose, but id >>>>> > did not work in your case. >>>>> > >>>>> > In Linux you can use inotify. We have also used this method >>>>> for getting >>>>> > certain infos from the Freeswitch logfile and then to fire >>>>> actions (in >>>>> > our case firewall rules). Maybe this is an approach for you? >>>>> > >>>>> > Best regards >>>>> > Peter >>>>> > >>>>> > On 05/16/17 07:25, Bipin Patel wrote: >>>>> >> hi, >>>>> >> >>>>> >> well considering im running this FS on the raspberry pi for >>>>> a small >>>>> >> office, ESL would be like a lot just for the purpose, if >>>>> the console can >>>>> >> display the fax result then why isnt it possible to get the >>>>> same in >>>>> >> dialplan or somewhere else where i can fire a script or >>>>> something and >>>>> >> let the sender know what happened to their fax. >>>>> >> >>>>> >> i tried using execute_on_fax_success and >>>>> execute_on_fax_failure but they >>>>> >> dont seem to work while sending a fax, i think its for >>>>> receiving only >>>>> >> >>>>> >> >>>>> >> -------- Original Message -------- >>>>> >> Subject: Re: [Freeswitch-users] Fax result when using mail >>>>> to fax >>>>> >> From: Peter Steinbach >>>> > >>>>> >> To: FreeSWITCH Users Help >>>>> >>>> > >>>>> >> Date: 5/16/2017, 12:24:44 AM >>>>> >> >>>>> >>> Hello Bipin, >>>>> >>> >>>>> >>> there are some events fired via ESL, if you subscribe to >>>>> them. We >>>>> >>> evaluate txfaxresult event and hangup event. >>>>> >>> >>>>> >>> The only thing which is critcal here, is that the contents >>>>> of the 2 >>>>> >>> events have to be joined, in order to have all >>>>> informations. And in my >>>>> >>> experience txfaxresult and hangup event do not always come >>>>> in the same >>>>> >>> order, dependend on which side of the channels hangs up first. >>>>> >>> >>>>> >>> Best regards >>>>> >>> Peter >>>>> >>> >>>>> >>> On 05/15/17 22:06, Bipin Patel wrote: >>>>> >>>> Hi, >>>>> >>>> >>>>> >>>> I have a python script that checks a mailbox for faxes to >>>>> be sent, when it >>>>> >>>> finds a new mail it downloads the PDF and converts to >>>>> Tiff and sends via >>>>> >>>> the originate command, now the problem is I can't seem to >>>>> find a way to get >>>>> >>>> the fax report, nor in cdr it says what happened and the >>>>> originate just >>>>> >>>> says ok followed by uuid of the bridge. I'm sending the >>>>> fax over a fxo >>>>> >>>> gateway device. Due to it going over fxo, I have no clue >>>>> if the remote fax >>>>> >>>> machine answered call and the only place the fax report >>>>> comes is in the >>>>> >>>> console. Is there a variable or anything else that can be >>>>> resent to the >>>>> >>>> python script telling what the result was. I tried the >>>>> execute on fax >>>>> >>>> successful and fail variable but they don't seem to work. >>>>> >>>> >>>>> >>>> My python script generates the originate command going to >>>>> loopback which >>>>> >>>> goes to dial plan where the actual bridge to fxo gateway >>>>> happens. >>>>> >>>> >>>>> >>>> How are others dealing with fax report I would love to hear. >>>>> >>>> >>>>> >>> >>>>> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170517/d7ee7d24/attachment-0001.html From yu at yu-boot.ru Wed May 17 13:26:14 2017 From: yu at yu-boot.ru (Yu Boot) Date: Wed, 17 May 2017 12:26:14 +0300 Subject: [Freeswitch-users] T.38 in new FS versions In-Reply-To: References: <15c0db7b410.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com> <81c86120-7679-4357-55a3-a66bf009870f@yu-boot.ru> Message-ID: <668ce63c-f5a2-05e1-25de-fd997611c4a6@yu-boot.ru> I just can't establish T.38 via FS like I did before. FS sends "488 not acceptable here" on T.38 reINVITE from B-leg and that's all. I use late codec negotiation and "inherit codec" option. How to revert previous T.38 behaviour? 16.05.2017 23:37, Brian West ?????: > This hasn't changed really, the only difference now is we don't > blindly accept reinvites unless you specifically enable T.38. What > kind of problem are you experiencing? > > /b > > > On Tue, May 16, 2017 at 7:31 AM, Yu Boot > wrote: > > Hi there. How exactly to enable "classic" T.38 faxes with reINVITE in > new FS versions? Not transcoding, not fax-to-email etc. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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.org > > */Twitter: @FreeSWITCH , @briankwest/* > > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > Book a phone call (CST) > > Allison prompts for FreeSWITCH: > > *https://www.gofundme.com/allison-prompts-for-freeswitch* > > Got Bugs? Report them here ! | Reddit: > /r/freeswitch > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *Skype:*briankwest > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170517/304466de/attachment.html From bipin at xbipin.com Wed May 17 14:03:11 2017 From: bipin at xbipin.com (Bipin Patel) Date: Wed, 17 May 2017 14:03:11 +0400 Subject: [Freeswitch-users] T.38 in new FS versions In-Reply-To: <668ce63c-f5a2-05e1-25de-fd997611c4a6@yu-boot.ru> References: <15c0db7b410.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com> <81c86120-7679-4357-55a3-a66bf009870f@yu-boot.ru> <668ce63c-f5a2-05e1-25de-fd997611c4a6@yu-boot.ru> Message-ID: <8adc43b9-3b5c-5056-3f04-ce5dc7755451@xbipin.com> hi, just set these 2 and ull get previous behavior fax_enable_t38_request=true fax_enable_t38=true Regards, Bipin ------------------------------------------------------------------------ -------- Original Message -------- Subject: Re: [Freeswitch-users] T.38 in new FS versions From: Yu Boot To: freeswitch-users at lists.freeswitch.org Date: 5/17/2017, 1:26:14 PM > > I just can't establish T.38 via FS like I did before. FS sends "488 > not acceptable here" on T.38 reINVITE from B-leg and that's all. I use > late codec negotiation and "inherit codec" option. How to revert > previous T.38 behaviour? > > > 16.05.2017 23:37, Brian West ?????: >> This hasn't changed really, the only difference now is we don't >> blindly accept reinvites unless you specifically enable T.38. What >> kind of problem are you experiencing? >> >> /b >> >> >> On Tue, May 16, 2017 at 7:31 AM, Yu Boot > > wrote: >> >> Hi there. How exactly to enable "classic" T.38 faxes with reINVITE in >> new FS versions? Not transcoding, not fax-to-email etc. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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.org >> >> */Twitter: @FreeSWITCH , @briankwest/* >> >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> Book a phone call (CST) >> >> Allison prompts for FreeSWITCH: >> >> *https://www.gofundme.com/allison-prompts-for-freeswitch* >> >> Got Bugs? Report them here ! | Reddit: >> /r/freeswitch >> >> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >> *Skype:*briankwest >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170517/283c5aca/attachment-0001.html From yu at yu-boot.ru Wed May 17 14:19:22 2017 From: yu at yu-boot.ru (Yu Boot) Date: Wed, 17 May 2017 13:19:22 +0300 Subject: [Freeswitch-users] T.38 in new FS versions In-Reply-To: <8adc43b9-3b5c-5056-3f04-ce5dc7755451@xbipin.com> References: <15c0db7b410.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com> <81c86120-7679-4357-55a3-a66bf009870f@yu-boot.ru> <668ce63c-f5a2-05e1-25de-fd997611c4a6@yu-boot.ru> <8adc43b9-3b5c-5056-3f04-ce5dc7755451@xbipin.com> Message-ID: Where to add this? vars.xml? ? 17.05.2017 13:03, Bipin Patel ?????: > hi, > > just set these 2 and ull get previous behavior > > fax_enable_t38_request=true > fax_enable_t38=true > > > Regards, > Bipin > > > ------------------------------------------------------------------------ > -------- Original Message -------- > Subject: Re: [Freeswitch-users] T.38 in new FS versions > From: Yu Boot > To: freeswitch-users at lists.freeswitch.org > Date: 5/17/2017, 1:26:14 PM >> >> I just can't establish T.38 via FS like I did before. FS sends "488 >> not acceptable here" on T.38 reINVITE from B-leg and that's all. I >> use late codec negotiation and "inherit codec" option. How to revert >> previous T.38 behaviour? >> >> >> 16.05.2017 23:37, Brian West ?????: >>> This hasn't changed really, the only difference now is we don't >>> blindly accept reinvites unless you specifically enable T.38. What >>> kind of problem are you experiencing? >>> >>> /b >>> >>> >>> On Tue, May 16, 2017 at 7:31 AM, Yu Boot >> > wrote: >>> >>> Hi there. How exactly to enable "classic" T.38 faxes with >>> reINVITE in >>> new FS versions? Not transcoding, not fax-to-email etc. >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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.org >>> >>> */Twitter: @FreeSWITCH , @briankwest/* >>> >>> http://www.freeswitchbook.com >>> http://www.freeswitchcookbook.com >>> >>> Book a phone call (CST) >>> >>> Allison prompts for FreeSWITCH: >>> >>> *https://www.gofundme.com/allison-prompts-for-freeswitch* >>> >>> Got Bugs? Report them here ! | Reddit: >>> /r/freeswitch >>> >>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>> *Skype:*briankwest >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170517/05336690/attachment.html From bipin at xbipin.com Wed May 17 15:38:27 2017 From: bipin at xbipin.com (Bipin Patel) Date: Wed, 17 May 2017 15:38:27 +0400 Subject: [Freeswitch-users] T.38 in new FS versions In-Reply-To: References: <15c0db7b410.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com> <81c86120-7679-4357-55a3-a66bf009870f@yu-boot.ru> <668ce63c-f5a2-05e1-25de-fd997611c4a6@yu-boot.ru> <8adc43b9-3b5c-5056-3f04-ce5dc7755451@xbipin.com> Message-ID: <8b443173-47ea-be71-568c-c99d3c3c67b9@xbipin.com> hi, yes or u can set them just before the bridge in the dialplan or if ur using originate command u can set them inside it as well Regards, Bipin ------------------------------------------------------------------------ -------- Original Message -------- Subject: Re: [Freeswitch-users] T.38 in new FS versions From: Yu Boot To: freeswitch-users at lists.freeswitch.org Date: 5/17/2017, 2:19:22 PM > > Where to add this? vars.xml? > > > > ? > > > > > 17.05.2017 13:03, Bipin Patel ?????: >> hi, >> >> just set these 2 and ull get previous behavior >> >> fax_enable_t38_request=true >> fax_enable_t38=true >> >> >> Regards, >> Bipin >> >> >> ------------------------------------------------------------------------ >> -------- Original Message -------- >> Subject: Re: [Freeswitch-users] T.38 in new FS versions >> From: Yu Boot >> To: freeswitch-users at lists.freeswitch.org >> Date: 5/17/2017, 1:26:14 PM >>> >>> I just can't establish T.38 via FS like I did before. FS sends "488 >>> not acceptable here" on T.38 reINVITE from B-leg and that's all. I >>> use late codec negotiation and "inherit codec" option. How to revert >>> previous T.38 behaviour? >>> >>> >>> 16.05.2017 23:37, Brian West ?????: >>>> This hasn't changed really, the only difference now is we don't >>>> blindly accept reinvites unless you specifically enable T.38. What >>>> kind of problem are you experiencing? >>>> >>>> /b >>>> >>>> >>>> On Tue, May 16, 2017 at 7:31 AM, Yu Boot >>> > wrote: >>>> >>>> Hi there. How exactly to enable "classic" T.38 faxes with >>>> reINVITE in >>>> new FS versions? Not transcoding, not fax-to-email etc. >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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.org >>>> >>>> */Twitter: @FreeSWITCH , @briankwest/* >>>> >>>> http://www.freeswitchbook.com >>>> http://www.freeswitchcookbook.com >>>> >>>> Book a phone call (CST) >>>> >>>> Allison prompts for FreeSWITCH: >>>> >>>> *https://www.gofundme.com/allison-prompts-for-freeswitch* >>>> >>>> Got Bugs? Report them here ! | Reddit: >>>> /r/freeswitch >>>> >>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>>> *Skype:*briankwest >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170517/a4253402/attachment-0001.html From brian at freeswitch.org Wed May 17 17:52:52 2017 From: brian at freeswitch.org (Brian West) Date: Wed, 17 May 2017 08:52:52 -0500 Subject: [Freeswitch-users] T.38 in new FS versions In-Reply-To: <8b443173-47ea-be71-568c-c99d3c3c67b9@xbipin.com> References: <15c0db7b410.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com> <81c86120-7679-4357-55a3-a66bf009870f@yu-boot.ru> <668ce63c-f5a2-05e1-25de-fd997611c4a6@yu-boot.ru> <8adc43b9-3b5c-5056-3f04-ce5dc7755451@xbipin.com> <8b443173-47ea-be71-568c-c99d3c3c67b9@xbipin.com> Message-ID: I sent out an email to the list warning everyone of this behavior change when it happened. Any pointers on what I can do to get this information where list members would actually see it? On Wed, May 17, 2017 at 6:38 AM, Bipin Patel wrote: > hi, > > yes or u can set them just before the bridge in the dialplan or if ur > using originate command u can set them inside it as well > > > Regards, > Bipin > > > ------------------------------ > -------- Original Message -------- > Subject: Re: [Freeswitch-users] T.38 in new FS versions > From: Yu Boot > To: freeswitch-users at lists.freeswitch.org > Date: 5/17/2017, 2:19:22 PM > > Where to add this? vars.xml? > > > > ? > > > > > 17.05.2017 13:03, Bipin Patel ?????: > > hi, > > just set these 2 and ull get previous behavior > > fax_enable_t38_request=true > fax_enable_t38=true > > > Regards, > Bipin > > > ------------------------------ > -------- Original Message -------- > Subject: Re: [Freeswitch-users] T.38 in new FS versions > From: Yu Boot > To: freeswitch-users at lists.freeswitch.org > Date: 5/17/2017, 1:26:14 PM > > I just can't establish T.38 via FS like I did before. FS sends "488 not > acceptable here" on T.38 reINVITE from B-leg and that's all. I use late > codec negotiation and "inherit codec" option. How to revert previous T.38 > behaviour? > > 16.05.2017 23:37, Brian West ?????: > > This hasn't changed really, the only difference now is we don't blindly > accept reinvites unless you specifically enable T.38. What kind of problem > are you experiencing? > > /b > > > On Tue, May 16, 2017 at 7:31 AM, Yu Boot wrote: > >> Hi there. How exactly to enable "classic" T.38 faxes with reINVITE in >> new FS versions? Not transcoding, not fax-to-email etc. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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.org > > *Twitter: @FreeSWITCH , @briankwest* > > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > Book a phone call (CST) > > Allison prompts for FreeSWITCH: > > *https://www.gofundme.com/allison-prompts-for-freeswitch* > > > Got Bugs? Report them here ! | Reddit: > /r/freeswitch > > *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002> > | *M:*+1918424WEST (9378) > *Skype:*briankwest > > > _________________________________________________________________________ > 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.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.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.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 > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Book a phone call (CST) Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170517/2a32cf5c/attachment-0001.html From asilva at wirelessmundi.com Wed May 17 18:01:44 2017 From: asilva at wirelessmundi.com (Antonio Silva) Date: Wed, 17 May 2017 16:01:44 +0200 Subject: [Freeswitch-users] T.38 in new FS versions In-Reply-To: References: <15c0db7b410.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com> <81c86120-7679-4357-55a3-a66bf009870f@yu-boot.ru> <668ce63c-f5a2-05e1-25de-fd997611c4a6@yu-boot.ru> <8adc43b9-3b5c-5056-3f04-ce5dc7755451@xbipin.com> <8b443173-47ea-be71-568c-c99d3c3c67b9@xbipin.com> Message-ID: Hi , we already have lot of info about t38 in fs located at https://freeswitch.org/confluence/display/FREESWITCH/mod_spandsp note sure but we probably could have "an attention note" to the change introduced here.. then when searching on google it will point to confluence :) Regards, Ant?nio On 05/17/2017 03:52 PM, Brian West wrote: > I sent out an email to the list warning everyone of this behavior > change when it happened. Any pointers on what I can do to get this > information where list members would actually see it? > > On Wed, May 17, 2017 at 6:38 AM, Bipin Patel > wrote: > > hi, > > yes or u can set them just before the bridge in the dialplan or if > ur using originate command u can set them inside it as well > > > Regards, > Bipin > > > ------------------------------------------------------------------------ > -------- Original Message -------- > Subject: Re: [Freeswitch-users] T.38 in new FS versions > From: Yu Boot > To: freeswitch-users at lists.freeswitch.org > > Date: 5/17/2017, 2:19:22 PM >> >> Where to add this? vars.xml? >> >> >> >> ? >> >> >> >> >> 17.05.2017 13:03, Bipin Patel ?????: >>> hi, >>> >>> just set these 2 and ull get previous behavior >>> >>> fax_enable_t38_request=true >>> fax_enable_t38=true >>> >>> >>> Regards, >>> Bipin >>> >>> >>> ------------------------------------------------------------------------ >>> -------- Original Message -------- >>> Subject: Re: [Freeswitch-users] T.38 in new FS versions >>> From: Yu Boot >>> To: freeswitch-users at lists.freeswitch.org >>> >>> Date: 5/17/2017, 1:26:14 PM >>>> >>>> I just can't establish T.38 via FS like I did before. FS sends >>>> "488 not acceptable here" on T.38 reINVITE from B-leg and >>>> that's all. I use late codec negotiation and "inherit codec" >>>> option. How to revert previous T.38 behaviour? >>>> >>>> >>>> 16.05.2017 23:37, Brian West ?????: >>>>> This hasn't changed really, the only difference now is we >>>>> don't blindly accept reinvites unless you specifically enable >>>>> T.38. What kind of problem are you experiencing? >>>>> >>>>> /b >>>>> >>>>> >>>>> On Tue, May 16, 2017 at 7:31 AM, Yu Boot >>>> > wrote: >>>>> >>>>> Hi there. How exactly to enable "classic" T.38 faxes with >>>>> reINVITE in >>>>> new FS versions? Not transcoding, not fax-to-email etc. >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at 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.org >>>>> >>>>> */Twitter: @FreeSWITCH , @briankwest/* >>>>> >>>>> http://www.freeswitchbook.com >>>>> http://www.freeswitchcookbook.com >>>>> >>>>> >>>>> Book a phone call (CST) >>>>> >>>>> Allison prompts for FreeSWITCH: >>>>> >>>>> *https://www.gofundme.com/allison-prompts-for-freeswitch* >>>>> >>>>> >>>>> Got Bugs? Report them here ! | >>>>> Reddit: /r/freeswitch >>>>> >>>>> *T:*+19184209001 | *F:*+19184209002 >>>>> | *M:*+1918424WEST (9378) >>>>> *Skype:*briankwest >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at 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 > > > > > -- > > */Brian West/* > brian at freeswitch.org > > */Twitter: @FreeSWITCH , @briankwest/* > > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > Book a phone call (CST) > > Allison prompts for FreeSWITCH: > > *https://www.gofundme.com/allison-prompts-for-freeswitch* > > Got Bugs? Report them here ! | Reddit: > /r/freeswitch > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *Skype:*briankwest > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170517/2ad7f358/attachment-0001.html From brian at freeswitch.org Wed May 17 19:45:30 2017 From: brian at freeswitch.org (Brian West) Date: Wed, 17 May 2017 10:45:30 -0500 Subject: [Freeswitch-users] T.38 in new FS versions In-Reply-To: References: <15c0db7b410.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com> <81c86120-7679-4357-55a3-a66bf009870f@yu-boot.ru> <668ce63c-f5a2-05e1-25de-fd997611c4a6@yu-boot.ru> <8adc43b9-3b5c-5056-3f04-ce5dc7755451@xbipin.com> <8b443173-47ea-be71-568c-c99d3c3c67b9@xbipin.com> Message-ID: What I find somewhat funny is Yu Boot actually was involved in the thread where I announced the behavior change, so maybe it wasn't well understood what the change was. /b On Wed, May 17, 2017 at 8:52 AM, Brian West wrote: > I sent out an email to the list warning everyone of this behavior change > when it happened. Any pointers on what I can do to get this information > where list members would actually see it? > > On Wed, May 17, 2017 at 6:38 AM, Bipin Patel wrote: > >> hi, >> >> yes or u can set them just before the bridge in the dialplan or if ur >> using originate command u can set them inside it as well >> >> >> Regards, >> Bipin >> >> >> ------------------------------ >> -------- Original Message -------- >> Subject: Re: [Freeswitch-users] T.38 in new FS versions >> From: Yu Boot >> To: freeswitch-users at lists.freeswitch.org >> Date: 5/17/2017, 2:19:22 PM >> >> Where to add this? vars.xml? >> >> >> >> ? >> >> >> >> >> 17.05.2017 13:03, Bipin Patel ?????: >> >> hi, >> >> just set these 2 and ull get previous behavior >> >> fax_enable_t38_request=true >> fax_enable_t38=true >> >> >> Regards, >> Bipin >> >> >> ------------------------------ >> -------- Original Message -------- >> Subject: Re: [Freeswitch-users] T.38 in new FS versions >> From: Yu Boot >> To: freeswitch-users at lists.freeswitch.org >> Date: 5/17/2017, 1:26:14 PM >> >> I just can't establish T.38 via FS like I did before. FS sends "488 not >> acceptable here" on T.38 reINVITE from B-leg and that's all. I use late >> codec negotiation and "inherit codec" option. How to revert previous T.38 >> behaviour? >> >> 16.05.2017 23:37, Brian West ?????: >> >> This hasn't changed really, the only difference now is we don't blindly >> accept reinvites unless you specifically enable T.38. What kind of problem >> are you experiencing? >> >> /b >> >> >> On Tue, May 16, 2017 at 7:31 AM, Yu Boot wrote: >> >>> Hi there. How exactly to enable "classic" T.38 faxes with reINVITE in >>> new FS versions? Not transcoding, not fax-to-email etc. >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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.org >> >> *Twitter: @FreeSWITCH , @briankwest* >> >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> Book a phone call (CST) >> >> Allison prompts for FreeSWITCH: >> >> *https://www.gofundme.com/allison-prompts-for-freeswitch* >> >> >> Got Bugs? Report them here ! | Reddit: >> /r/freeswitch >> >> *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002> >> | *M:*+1918424WEST (9378) >> *Skype:*briankwest >> >> >> _________________________________________________________________________ >> 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.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.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.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 >> > > > > -- > > *Brian West* > brian at freeswitch.org > > *Twitter: @FreeSWITCH , @briankwest* > > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > Book a phone call (CST) > > Allison prompts for FreeSWITCH: > > *https://www.gofundme.com/allison-prompts-for-freeswitch* > > > Got Bugs? Report them here ! | Reddit: > /r/freeswitch > > *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002> > | *M:*+1918424WEST (9378) > *Skype:*briankwest > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Book a phone call (CST) Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170517/290d5ad3/attachment-0001.html From rick at magicmail.mooo.com Wed May 17 20:02:46 2017 From: rick at magicmail.mooo.com (Rick Jarvis) Date: Wed, 17 May 2017 17:02:46 +0100 Subject: [Freeswitch-users] Paging Message-ID: <8AE4D7F5-1AE6-44F8-B85D-13C4FADC22CA@magicmail.mooo.com> Looking to set up paging (not multicast). What?s the best way of achieving this? Specifically, I want to have the receiving handset(s) answer muted for privacy reasons, so it?s literally like a PA system rather than just auto answer?? From rbetancor at gmail.com Thu May 18 11:22:03 2017 From: rbetancor at gmail.com (=?UTF-8?Q?Ra=C3=BAl_Alexis_Betancor_Santana?=) Date: Thu, 18 May 2017 08:22:03 +0100 Subject: [Freeswitch-users] Noob question about xml dialplan In-Reply-To: References: Message-ID: That command give me the clue ... if I do xml_locate dialplan it returns me an empty response ... so I assume that I'm doing something wrong and it's not loading any dialplan. 2017-05-16 19:48 GMT+01:00 David Villasmil : > "xml_locate dialplan" > > will give you the whole xml for the dialplan > > "xml_locate configuration configuration name sofia.conf" > > will give you your sofia.conf > ? > > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 <+34%20669%2044%2083%2037> > > On Tue, May 16, 2017 at 8:15 PM, David Villasmil < > david.villasmil.work at gmail.com> wrote: > >> Yeah that command doesn't exists. But there is another command, >> xml_locate or something like that. >> Remove the context param and try again. Post the latest version of the >> extension with the section i pointed out before. >> >> I can't send you the fsxml as I'm not in my computer right now. I'll send >> it in a few minutes. >> >> On Tue, May 16, 2017 at 8:12 PM Ra?l Alexis Betancor Santana < >> rbetancor at gmail.com> wrote: >> >>> Thats what I'm trying ... to undestand how it works. >>> >>> I posted the profile I'm using a couple of messasges earlier, anyway: >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> I changed the the extension pattern to ^(.*) ... still the same result >>> >>> Could you please share an example freeswitch.xml.fsxml file ? ... maybe >>> the problem is how the includes are handled, so something is not on the xml >>> node it needs to. >>> >>> I miss a command like the 'show dialplan' ... where you could check how >>> an specific call would behalf >>> >>> >>> 2017-05-16 19:01 GMT+01:00 David Villasmil < >>> david.villasmil.work at gmail.com>: >>> >>>> Hello, >>>> >>>> You need to start by understanding freeswitch before comparing the two, >>>> i work with both and there's just no comparison, as far as I'm concerned. >>>> >>>> When you create a profile, this profile binds to a port. In the profile >>>> config there's two params, context and diaplan (though tbh until today i >>>> never ever touched the dialplan param, as i haven't had any need to do it. >>>> If i need a simple diaplan, i just remove all xmls included in said diaplan >>>> and have just one file. >>>> >>>> Now, the "context" param, along with the includes, is what fs uses to >>>> hunt for extensions. Any file loaded from the profile which includes the >>>> context name, will be use to hunt for an extension. >>>> >>>> You haven't posted your profile, so it's hard to figure out what could >>>> be wrong, as when i tried it, it worked perfectly. I should note i >>>> commented out the context param. >>>> >>>> For the sake of simplicity, can you change your extension to answer >>>> EVERYTHING, (i.e.: "^(.*)") ? >>>> >>>> >>>> >>>> On Tue, May 16, 2017 at 7:53 PM Ra?l Alexis Betancor Santana < >>>> rbetancor at gmail.com> wrote: >>>> >>>>> Just for the shake of simplicity ... on the Asterisk world would be >>>>> something like: >>>>> >>>>> exten => _1NXXNXXXXXX,1,Noop("T38 Gateway Call") >>>>> exten => _1NXXNXXXXXX,n,Set(FAXOPT(gateway)=yes) >>>>> exten => _1NXXNXXXXXX,n,Dial(SIP/${EXTEN}@proxy.server.tld) >>>>> >>>>> I just want to make the same with FS, but not allowing T38 >>>>> passthrought at all ... just T38 (a leg) to Ulaw (b leg) >>>>> >>>>> 2017-05-16 17:48 GMT+01:00 Ra?l Alexis Betancor Santana < >>>>> rbetancor at gmail.com>: >>>>> >>>>>> Same result ... No Route, Aborting >>>>>> >>>>>> Lets see if I found some step-by-step tutorial that explains who the >>>>>> dialplan works. The doc on the Wiki it's not clear enought for me. >>>>>> >>>>>> 2017-05-16 17:40 GMT+01:00 David Villasmil < >>>>>> david.villasmil.work at gmail.com>: >>>>>> >>>>>>> That was my test extension, you can replace it for whatever you need >>>>>>> :) >>>>>>> ? >>>>>>> >>>>>>> Regards, >>>>>>> >>>>>>> David Villasmil >>>>>>> email: david.villasmil.work at gmail.com >>>>>>> phone: +34669448337 <+34%20669%2044%2083%2037> >>>>>>> >>>>>>> On Tue, May 16, 2017 at 6:38 PM, Ra?l Alexis Betancor Santana < >>>>>>> rbetancor at gmail.com> wrote: >>>>>>> >>>>>>>> I don't get the point of calling a user/$1 ... as there is no users >>>>>>>> registered against this FS ... I just want to use it as a T38->Ulaw gateway. >>>>>>>> >>>>>>>> 2017-05-16 17:34 GMT+01:00 David Villasmil < >>>>>>>> david.villasmil.work at gmail.com>: >>>>>>>> >>>>>>>>> that context line is not necessary, btw. point is you need the >>>>>>>>> section: >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>>
>>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>>
>>>>>>>>> ? >>>>>>>>> >>>>>>>>> Regards, >>>>>>>>> >>>>>>>>> David Villasmil >>>>>>>>> email: david.villasmil.work at gmail.com >>>>>>>>> phone: +34669448337 <+34%20669%2044%2083%2037> >>>>>>>>> >>>>>>>>> On Tue, May 16, 2017 at 6:33 PM, David Villasmil < >>>>>>>>> david.villasmil.work at gmail.com> wrote: >>>>>>>>> >>>>>>>>>> Ok i've looked at the source code for mod_dialplan and in your >>>>>>>>>> xml, you need something like: >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>>
>>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>>
>>>>>>>>> ? >>>>>>>>>> >>>>>>>>>> Regards, >>>>>>>>>> >>>>>>>>>> David Villasmil >>>>>>>>>> email: david.villasmil.work at gmail.com >>>>>>>>>> phone: +34669448337 <+34%20669%2044%2083%2037> >>>>>>>>>> >>>>>>>>>> On Tue, May 16, 2017 at 5:26 PM, Ra?l Alexis Betancor Santana < >>>>>>>>>> rbetancor at gmail.com> wrote: >>>>>>>>>> >>>>>>>>>>> Ok, I think the whole point is that I don't undestand how the >>>>>>>>>>> dialplan works. >>>>>>>>>>> >>>>>>>>>>> I started from 0 ... now on the sofia.conf.xml I have this: >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>> value="udp:$${homer_ip}:$${homer_port};hep=3;capture_id=2001"/> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>> value="localnet.auto"/> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>> value="generous"/> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>> value="false"/> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>> value="$${base_dir}/conf/ssl"/> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> Then on the t38_transconde.xml file I have this. >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>> expresion="^(\+1|1)?([2-9]\d\d[2-9]\d{6})$"> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> I could see on the /var/log/freeswitch/freeswitch.xml.fsxml >>>>>>>>>>> file that the t38_transcode.xml file its included ... but still get the >>>>>>>>>>> same error on on the console, so something terrible wrong I'm missing here >>>>>>>>>>> ... :-( >>>>>>>>>>> >>>>>>>>>>> Also ... it's possible to directly send the calls sofia/$1$2@$${external} >>>>>>>>>>> ? ... I mean, sending the call directly to a URI, with username/password >>>>>>>>>>> and so on ... instead of having to define a profile for it. >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> 2017-05-16 16:13 GMT+01:00 David Villasmil < >>>>>>>>>>> david.villasmil.work at gmail.com>: >>>>>>>>>>> >>>>>>>>>>>> No, you may be sending it to the right ports, but the profile >>>>>>>>>>>> attached to that port must have the context set to the correct dialplan. >>>>>>>>>>>> >>>>>>>>>>>> I just did exactly what you're doing and i also got "No Route, >>>>>>>>>>>> Aborting" >>>>>>>>>>>> >>>>>>>>>>>> I've never seen this type of "dialplan" value, tbh >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> Regards, >>>>>>>>>>>> >>>>>>>>>>>> David Villasmil >>>>>>>>>>>> email: david.villasmil.work at gmail.com >>>>>>>>>>>> phone: +34669448337 <+34%20669%2044%2083%2037> >>>>>>>>>>>> >>>>>>>>>>>> On Tue, May 16, 2017 at 3:30 PM, Ra?l Alexis Betancor Santana < >>>>>>>>>>>> rbetancor at gmail.com> wrote: >>>>>>>>>>>> >>>>>>>>>>>>> Only one profile defined on the sofia.conf.xml and I'm sending >>>>>>>>>>>>> the traffict to the wright ports, if not I whould get the logs on the >>>>>>>>>>>>> console, as that are the only ports enabled. >>>>>>>>>>>>> >>>>>>>>>>>>> 2017-05-16 14:22 GMT+01:00 David Villasmil < >>>>>>>>>>>>> david.villasmil.work at gmail.com>: >>>>>>>>>>>>> >>>>>>>>>>>>>> Take a look at your profile, it should be listening on the >>>>>>>>>>>>>> port you're sending to, and must have the context parameter set to your >>>>>>>>>>>>>> dialplan name. >>>>>>>>>>>>>> On Tue, May 16, 2017 at 3:15 PM Ra?l Alexis Betancor Santana < >>>>>>>>>>>>>> rbetancor at gmail.com> wrote: >>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> Hi all, till now I'm been working with pre-made setup files >>>>>>>>>>>>>>> for FS, and now I'm trying to get a deep knowleadge of how the dialplan >>>>>>>>>>>>>>> works. >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> So I modifed my autoload_configs/sofia.conf.xml file and >>>>>>>>>>>>>>> changed my dialplan param to something like this: >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> The Idea is that it loads another .xml file especific task >>>>>>>>>>>>>>> and also use a dialplan throught a socket to a daemon that handle the rest. >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> On my t38_transcode.xml file ... very simple: >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>> expresion="^True"/> >>>>>>>>>>>>>>> >>>>>>>>>>>>>> expresion="^(\+1|1)?([2-9]\d\d[2-9]\d{6})$"> >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>> data="sofia/external/$1$2 at proxy.server.tld"/> >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> But If I fire a call to FS like [number]@[FS_IP] from a sip >>>>>>>>>>>>>>> testing client I get >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> [INFO] switch_core_state_machine.c:311 No Route, Aborting >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> What I'm doing wrong here? >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> My target its just I want to ANY call that came in with a >>>>>>>>>>>>>>> SIP header of X-T39-Transcode=True (market by a sip proxy elsewhere on the >>>>>>>>>>>>>>> net) just do a T38->Ulaw transcoding saving the T38 trace, so I could >>>>>>>>>>>>>>> inspect it later. >>>>>>>>>>>>>>> The rest of calls coming in ... as they don't have the >>>>>>>>>>>>>>> sip-header should end on other app. >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> For doing the testing I disabled the socket_inline part of >>>>>>>>>>>>>>> the dialplan string, so it have only this: >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> Did I miss something? ... or maybe missundestood who the xml >>>>>>>>>>>>>>> dialplan works ? >>>>>>>>>>>>>>> ____________________________________________________________ >>>>>>>>>>>>>>> _____________ >>>>>>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>>>>>> consulting at freeswitch.org >>>>>>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>>>>> http://confluence.freeswitch.org >>>>>>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-user >>>>>>>>>>>>>>> s >>>>>>>>>>>>>>> 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 >>>>>>> >>>>>> >>>>>> >>>>> ____________________________________________________________ >>>>> _____________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170518/ee11a880/attachment-0001.html From gregor at infomedia.si Thu May 18 11:51:15 2017 From: gregor at infomedia.si (Gregor Nanger) Date: Thu, 18 May 2017 09:51:15 +0200 Subject: [Freeswitch-users] Callcenter modul error:: register subclass Message-ID: ?Can anyone help with error that I get when try to load callcenter module: 2017-05-18 09:48:49.828297 [ERR] mod_callcenter.c:3810 Couldn't register subclass callcenter::info! 2017-05-18 09:48:49.828297 [CRIT] switch_loadable_module.c:1522 Error Loading module /usr/lib/freeswitch/mod/mod_callcenter.so **Module load routine returned an error**? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170518/d3404131/attachment.html From rbetancor at gmail.com Thu May 18 12:20:04 2017 From: rbetancor at gmail.com (=?UTF-8?Q?Ra=C3=BAl_Alexis_Betancor_Santana?=) Date: Thu, 18 May 2017 09:20:04 +0100 Subject: [Freeswitch-users] Noob question about xml dialplan In-Reply-To: References: Message-ID: Found it, and I supposed ... a noob fail ... X) ... I just was not adding the correct configuration statement on the freeswithc.xml file, and was loading the dialplan xml files inside the configuration section, instead of the dialplan section. 2017-05-18 8:22 GMT+01:00 Ra?l Alexis Betancor Santana : > That command give me the clue ... if I do xml_locate dialplan it returns > me an empty response ... so I assume that I'm doing something wrong and > it's not loading any dialplan. > > 2017-05-16 19:48 GMT+01:00 David Villasmil >: > >> "xml_locate dialplan" >> >> will give you the whole xml for the dialplan >> >> "xml_locate configuration configuration name sofia.conf" >> >> will give you your sofia.conf >> ? >> >> Regards, >> >> David Villasmil >> email: david.villasmil.work at gmail.com >> phone: +34669448337 <+34%20669%2044%2083%2037> >> >> On Tue, May 16, 2017 at 8:15 PM, David Villasmil < >> david.villasmil.work at gmail.com> wrote: >> >>> Yeah that command doesn't exists. But there is another command, >>> xml_locate or something like that. >>> Remove the context param and try again. Post the latest version of the >>> extension with the section i pointed out before. >>> >>> I can't send you the fsxml as I'm not in my computer right now. I'll >>> send it in a few minutes. >>> >>> On Tue, May 16, 2017 at 8:12 PM Ra?l Alexis Betancor Santana < >>> rbetancor at gmail.com> wrote: >>> >>>> Thats what I'm trying ... to undestand how it works. >>>> >>>> I posted the profile I'm using a couple of messasges earlier, anyway: >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> I changed the the extension pattern to ^(.*) ... still the same result >>>> >>>> Could you please share an example freeswitch.xml.fsxml file ? ... maybe >>>> the problem is how the includes are handled, so something is not on the xml >>>> node it needs to. >>>> >>>> I miss a command like the 'show dialplan' ... where you could check how >>>> an specific call would behalf >>>> >>>> >>>> 2017-05-16 19:01 GMT+01:00 David Villasmil < >>>> david.villasmil.work at gmail.com>: >>>> >>>>> Hello, >>>>> >>>>> You need to start by understanding freeswitch before comparing the >>>>> two, i work with both and there's just no comparison, as far as I'm >>>>> concerned. >>>>> >>>>> When you create a profile, this profile binds to a port. In the >>>>> profile config there's two params, context and diaplan (though tbh until >>>>> today i never ever touched the dialplan param, as i haven't had any need to >>>>> do it. If i need a simple diaplan, i just remove all xmls included in said >>>>> diaplan and have just one file. >>>>> >>>>> Now, the "context" param, along with the includes, is what fs uses to >>>>> hunt for extensions. Any file loaded from the profile which includes the >>>>> context name, will be use to hunt for an extension. >>>>> >>>>> You haven't posted your profile, so it's hard to figure out what could >>>>> be wrong, as when i tried it, it worked perfectly. I should note i >>>>> commented out the context param. >>>>> >>>>> For the sake of simplicity, can you change your extension to answer >>>>> EVERYTHING, (i.e.: "^(.*)") ? >>>>> >>>>> >>>>> >>>>> On Tue, May 16, 2017 at 7:53 PM Ra?l Alexis Betancor Santana < >>>>> rbetancor at gmail.com> wrote: >>>>> >>>>>> Just for the shake of simplicity ... on the Asterisk world would be >>>>>> something like: >>>>>> >>>>>> exten => _1NXXNXXXXXX,1,Noop("T38 Gateway Call") >>>>>> exten => _1NXXNXXXXXX,n,Set(FAXOPT(gateway)=yes) >>>>>> exten => _1NXXNXXXXXX,n,Dial(SIP/${EXTEN}@proxy.server.tld) >>>>>> >>>>>> I just want to make the same with FS, but not allowing T38 >>>>>> passthrought at all ... just T38 (a leg) to Ulaw (b leg) >>>>>> >>>>>> 2017-05-16 17:48 GMT+01:00 Ra?l Alexis Betancor Santana < >>>>>> rbetancor at gmail.com>: >>>>>> >>>>>>> Same result ... No Route, Aborting >>>>>>> >>>>>>> Lets see if I found some step-by-step tutorial that explains who the >>>>>>> dialplan works. The doc on the Wiki it's not clear enought for me. >>>>>>> >>>>>>> 2017-05-16 17:40 GMT+01:00 David Villasmil < >>>>>>> david.villasmil.work at gmail.com>: >>>>>>> >>>>>>>> That was my test extension, you can replace it for whatever you >>>>>>>> need :) >>>>>>>> ? >>>>>>>> >>>>>>>> Regards, >>>>>>>> >>>>>>>> David Villasmil >>>>>>>> email: david.villasmil.work at gmail.com >>>>>>>> phone: +34669448337 <+34%20669%2044%2083%2037> >>>>>>>> >>>>>>>> On Tue, May 16, 2017 at 6:38 PM, Ra?l Alexis Betancor Santana < >>>>>>>> rbetancor at gmail.com> wrote: >>>>>>>> >>>>>>>>> I don't get the point of calling a user/$1 ... as there is no >>>>>>>>> users registered against this FS ... I just want to use it as a T38->Ulaw >>>>>>>>> gateway. >>>>>>>>> >>>>>>>>> 2017-05-16 17:34 GMT+01:00 David Villasmil < >>>>>>>>> david.villasmil.work at gmail.com>: >>>>>>>>> >>>>>>>>>> that context line is not necessary, btw. point is you need the >>>>>>>>>> section: >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>>
>>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>>
>>>>>>>>>> ? >>>>>>>>>> >>>>>>>>>> Regards, >>>>>>>>>> >>>>>>>>>> David Villasmil >>>>>>>>>> email: david.villasmil.work at gmail.com >>>>>>>>>> phone: +34669448337 <+34%20669%2044%2083%2037> >>>>>>>>>> >>>>>>>>>> On Tue, May 16, 2017 at 6:33 PM, David Villasmil < >>>>>>>>>> david.villasmil.work at gmail.com> wrote: >>>>>>>>>> >>>>>>>>>>> Ok i've looked at the source code for mod_dialplan and in your >>>>>>>>>>> xml, you need something like: >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>>
>>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>>
>>>>>>>>>> ? >>>>>>>>>>> >>>>>>>>>>> Regards, >>>>>>>>>>> >>>>>>>>>>> David Villasmil >>>>>>>>>>> email: david.villasmil.work at gmail.com >>>>>>>>>>> phone: +34669448337 <+34%20669%2044%2083%2037> >>>>>>>>>>> >>>>>>>>>>> On Tue, May 16, 2017 at 5:26 PM, Ra?l Alexis Betancor Santana < >>>>>>>>>>> rbetancor at gmail.com> wrote: >>>>>>>>>>> >>>>>>>>>>>> Ok, I think the whole point is that I don't undestand how the >>>>>>>>>>>> dialplan works. >>>>>>>>>>>> >>>>>>>>>>>> I started from 0 ... now on the sofia.conf.xml I have this: >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>> data="../gateways/*.xml"/> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>> value="udp:$${homer_ip}:$${homer_port};hep=3;capture_id=2001"/> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>> value="localnet.auto"/> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>> value="generous"/> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>> value="false"/> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>> value="transport=tls"/> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>> value="$${base_dir}/conf/ssl"/> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> Then on the t38_transconde.xml file I have this. >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>> expresion="^(\+1|1)?([2-9]\d\d[2-9]\d{6})$"> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> I could see on the /var/log/freeswitch/freeswitch.xml.fsxml >>>>>>>>>>>> file that the t38_transcode.xml file its included ... but still get the >>>>>>>>>>>> same error on on the console, so something terrible wrong I'm missing here >>>>>>>>>>>> ... :-( >>>>>>>>>>>> >>>>>>>>>>>> Also ... it's possible to directly send the calls sofia/$1$2@$${external} >>>>>>>>>>>> ? ... I mean, sending the call directly to a URI, with username/password >>>>>>>>>>>> and so on ... instead of having to define a profile for it. >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> 2017-05-16 16:13 GMT+01:00 David Villasmil < >>>>>>>>>>>> david.villasmil.work at gmail.com>: >>>>>>>>>>>> >>>>>>>>>>>>> No, you may be sending it to the right ports, but the profile >>>>>>>>>>>>> attached to that port must have the context set to the correct dialplan. >>>>>>>>>>>>> >>>>>>>>>>>>> I just did exactly what you're doing and i also got "No Route, >>>>>>>>>>>>> Aborting" >>>>>>>>>>>>> >>>>>>>>>>>>> I've never seen this type of "dialplan" value, tbh >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> Regards, >>>>>>>>>>>>> >>>>>>>>>>>>> David Villasmil >>>>>>>>>>>>> email: david.villasmil.work at gmail.com >>>>>>>>>>>>> phone: +34669448337 <+34%20669%2044%2083%2037> >>>>>>>>>>>>> >>>>>>>>>>>>> On Tue, May 16, 2017 at 3:30 PM, Ra?l Alexis Betancor Santana >>>>>>>>>>>>> wrote: >>>>>>>>>>>>> >>>>>>>>>>>>>> Only one profile defined on the sofia.conf.xml and I'm >>>>>>>>>>>>>> sending the traffict to the wright ports, if not I whould get the logs on >>>>>>>>>>>>>> the console, as that are the only ports enabled. >>>>>>>>>>>>>> >>>>>>>>>>>>>> 2017-05-16 14:22 GMT+01:00 David Villasmil < >>>>>>>>>>>>>> david.villasmil.work at gmail.com>: >>>>>>>>>>>>>> >>>>>>>>>>>>>>> Take a look at your profile, it should be listening on the >>>>>>>>>>>>>>> port you're sending to, and must have the context parameter set to your >>>>>>>>>>>>>>> dialplan name. >>>>>>>>>>>>>>> On Tue, May 16, 2017 at 3:15 PM Ra?l Alexis Betancor Santana >>>>>>>>>>>>>>> wrote: >>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> Hi all, till now I'm been working with pre-made setup files >>>>>>>>>>>>>>>> for FS, and now I'm trying to get a deep knowleadge of how the dialplan >>>>>>>>>>>>>>>> works. >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> So I modifed my autoload_configs/sofia.conf.xml file and >>>>>>>>>>>>>>>> changed my dialplan param to something like this: >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> The Idea is that it loads another .xml file especific task >>>>>>>>>>>>>>>> and also use a dialplan throught a socket to a daemon that handle the rest. >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> On my t38_transcode.xml file ... very simple: >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>> expresion="^True"/> >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>> expresion="^(\+1|1)?([2-9]\d\d[2-9]\d{6})$"> >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>> data="sofia/external/$1$2 at proxy.server.tld"/> >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> But If I fire a call to FS like [number]@[FS_IP] from a sip >>>>>>>>>>>>>>>> testing client I get >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> [INFO] switch_core_state_machine.c:311 No Route, Aborting >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> What I'm doing wrong here? >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> My target its just I want to ANY call that came in with a >>>>>>>>>>>>>>>> SIP header of X-T39-Transcode=True (market by a sip proxy elsewhere on the >>>>>>>>>>>>>>>> net) just do a T38->Ulaw transcoding saving the T38 trace, so I could >>>>>>>>>>>>>>>> inspect it later. >>>>>>>>>>>>>>>> The rest of calls coming in ... as they don't have the >>>>>>>>>>>>>>>> sip-header should end on other app. >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> For doing the testing I disabled the socket_inline part of >>>>>>>>>>>>>>>> the dialplan string, so it have only this: >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> Did I miss something? ... or maybe missundestood who the >>>>>>>>>>>>>>>> xml dialplan works ? >>>>>>>>>>>>>>>> ______________________________ >>>>>>>>>>>>>>>> ___________________________________________ >>>>>>>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>>>>>>> consulting at freeswitch.org >>>>>>>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>>>>>> http://confluence.freeswitch.org >>>>>>>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>>>>>>>>>> http://lists.freeswitch.org/ma >>>>>>>>>>>>>>>> ilman/listinfo/freeswitch-users >>>>>>>>>>>>>>>> UNSUBSCRIBE:http://lists.frees >>>>>>>>>>>>>>>> witch.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-user >>>>>>>>>>>>>>> s >>>>>>>>>>>>>>> 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/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/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/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/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 >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170518/7687194b/attachment-0001.html From david.villasmil.work at gmail.com Thu May 18 12:24:02 2017 From: david.villasmil.work at gmail.com (David Villasmil) Date: Thu, 18 May 2017 10:24:02 +0200 Subject: [Freeswitch-users] Noob question about xml dialplan In-Reply-To: References: Message-ID: could you paste it? ? Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 On Thu, May 18, 2017 at 10:20 AM, Ra?l Alexis Betancor Santana < rbetancor at gmail.com> wrote: > Found it, and I supposed ... a noob fail ... X) ... I just was not adding > the correct configuration statement on the freeswithc.xml file, and was > loading the dialplan xml files inside the configuration section, instead of > the dialplan section. > > 2017-05-18 8:22 GMT+01:00 Ra?l Alexis Betancor Santana < > rbetancor at gmail.com>: > >> That command give me the clue ... if I do xml_locate dialplan it returns >> me an empty response ... so I assume that I'm doing something wrong and >> it's not loading any dialplan. >> >> 2017-05-16 19:48 GMT+01:00 David Villasmil > m>: >> >>> "xml_locate dialplan" >>> >>> will give you the whole xml for the dialplan >>> >>> "xml_locate configuration configuration name sofia.conf" >>> >>> will give you your sofia.conf >>> ? >>> >>> Regards, >>> >>> David Villasmil >>> email: david.villasmil.work at gmail.com >>> phone: +34669448337 <+34%20669%2044%2083%2037> >>> >>> On Tue, May 16, 2017 at 8:15 PM, David Villasmil < >>> david.villasmil.work at gmail.com> wrote: >>> >>>> Yeah that command doesn't exists. But there is another command, >>>> xml_locate or something like that. >>>> Remove the context param and try again. Post the latest version of the >>>> extension with the section i pointed out before. >>>> >>>> I can't send you the fsxml as I'm not in my computer right now. I'll >>>> send it in a few minutes. >>>> >>>> On Tue, May 16, 2017 at 8:12 PM Ra?l Alexis Betancor Santana < >>>> rbetancor at gmail.com> wrote: >>>> >>>>> Thats what I'm trying ... to undestand how it works. >>>>> >>>>> I posted the profile I'm using a couple of messasges earlier, anyway: >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> I changed the the extension pattern to ^(.*) ... still the same result >>>>> >>>>> Could you please share an example freeswitch.xml.fsxml file ? ... >>>>> maybe the problem is how the includes are handled, so something is not on >>>>> the xml node it needs to. >>>>> >>>>> I miss a command like the 'show dialplan' ... where you could check >>>>> how an specific call would behalf >>>>> >>>>> >>>>> 2017-05-16 19:01 GMT+01:00 David Villasmil < >>>>> david.villasmil.work at gmail.com>: >>>>> >>>>>> Hello, >>>>>> >>>>>> You need to start by understanding freeswitch before comparing the >>>>>> two, i work with both and there's just no comparison, as far as I'm >>>>>> concerned. >>>>>> >>>>>> When you create a profile, this profile binds to a port. In the >>>>>> profile config there's two params, context and diaplan (though tbh until >>>>>> today i never ever touched the dialplan param, as i haven't had any need to >>>>>> do it. If i need a simple diaplan, i just remove all xmls included in said >>>>>> diaplan and have just one file. >>>>>> >>>>>> Now, the "context" param, along with the includes, is what fs uses to >>>>>> hunt for extensions. Any file loaded from the profile which includes the >>>>>> context name, will be use to hunt for an extension. >>>>>> >>>>>> You haven't posted your profile, so it's hard to figure out what >>>>>> could be wrong, as when i tried it, it worked perfectly. I should note i >>>>>> commented out the context param. >>>>>> >>>>>> For the sake of simplicity, can you change your extension to answer >>>>>> EVERYTHING, (i.e.: "^(.*)") ? >>>>>> >>>>>> >>>>>> >>>>>> On Tue, May 16, 2017 at 7:53 PM Ra?l Alexis Betancor Santana < >>>>>> rbetancor at gmail.com> wrote: >>>>>> >>>>>>> Just for the shake of simplicity ... on the Asterisk world would be >>>>>>> something like: >>>>>>> >>>>>>> exten => _1NXXNXXXXXX,1,Noop("T38 Gateway Call") >>>>>>> exten => _1NXXNXXXXXX,n,Set(FAXOPT(gateway)=yes) >>>>>>> exten => _1NXXNXXXXXX,n,Dial(SIP/${EXTEN}@proxy.server.tld) >>>>>>> >>>>>>> I just want to make the same with FS, but not allowing T38 >>>>>>> passthrought at all ... just T38 (a leg) to Ulaw (b leg) >>>>>>> >>>>>>> 2017-05-16 17:48 GMT+01:00 Ra?l Alexis Betancor Santana < >>>>>>> rbetancor at gmail.com>: >>>>>>> >>>>>>>> Same result ... No Route, Aborting >>>>>>>> >>>>>>>> Lets see if I found some step-by-step tutorial that explains who >>>>>>>> the dialplan works. The doc on the Wiki it's not clear enought for me. >>>>>>>> >>>>>>>> 2017-05-16 17:40 GMT+01:00 David Villasmil < >>>>>>>> david.villasmil.work at gmail.com>: >>>>>>>> >>>>>>>>> That was my test extension, you can replace it for whatever you >>>>>>>>> need :) >>>>>>>>> ? >>>>>>>>> >>>>>>>>> Regards, >>>>>>>>> >>>>>>>>> David Villasmil >>>>>>>>> email: david.villasmil.work at gmail.com >>>>>>>>> phone: +34669448337 <+34%20669%2044%2083%2037> >>>>>>>>> >>>>>>>>> On Tue, May 16, 2017 at 6:38 PM, Ra?l Alexis Betancor Santana < >>>>>>>>> rbetancor at gmail.com> wrote: >>>>>>>>> >>>>>>>>>> I don't get the point of calling a user/$1 ... as there is no >>>>>>>>>> users registered against this FS ... I just want to use it as a T38->Ulaw >>>>>>>>>> gateway. >>>>>>>>>> >>>>>>>>>> 2017-05-16 17:34 GMT+01:00 David Villasmil < >>>>>>>>>> david.villasmil.work at gmail.com>: >>>>>>>>>> >>>>>>>>>>> that context line is not necessary, btw. point is you need the >>>>>>>>>>> section: >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>>
>>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>>
>>>>>>>>>>> ? >>>>>>>>>>> >>>>>>>>>>> Regards, >>>>>>>>>>> >>>>>>>>>>> David Villasmil >>>>>>>>>>> email: david.villasmil.work at gmail.com >>>>>>>>>>> phone: +34669448337 <+34%20669%2044%2083%2037> >>>>>>>>>>> >>>>>>>>>>> On Tue, May 16, 2017 at 6:33 PM, David Villasmil < >>>>>>>>>>> david.villasmil.work at gmail.com> wrote: >>>>>>>>>>> >>>>>>>>>>>> Ok i've looked at the source code for mod_dialplan and in your >>>>>>>>>>>> xml, you need something like: >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>>
>>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>>
>>>>>>>>>>> ? >>>>>>>>>>>> >>>>>>>>>>>> Regards, >>>>>>>>>>>> >>>>>>>>>>>> David Villasmil >>>>>>>>>>>> email: david.villasmil.work at gmail.com >>>>>>>>>>>> phone: +34669448337 <+34%20669%2044%2083%2037> >>>>>>>>>>>> >>>>>>>>>>>> On Tue, May 16, 2017 at 5:26 PM, Ra?l Alexis Betancor Santana < >>>>>>>>>>>> rbetancor at gmail.com> wrote: >>>>>>>>>>>> >>>>>>>>>>>>> Ok, I think the whole point is that I don't undestand how the >>>>>>>>>>>>> dialplan works. >>>>>>>>>>>>> >>>>>>>>>>>>> I started from 0 ... now on the sofia.conf.xml I have this: >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>> data="../gateways/*.xml"/> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>> value="udp:$${homer_ip}:$${homer_port};hep=3;capture_id=2001"/> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>> value="localnet.auto"/> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>> value="generous"/> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>> value="false"/> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>> value="transport=tls"/> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>> value="$${base_dir}/conf/ssl"/> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> Then on the t38_transconde.xml file I have this. >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>> expresion="^(\+1|1)?([2-9]\d\d[2-9]\d{6})$"> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> I could see on the /var/log/freeswitch/freeswitch.xml.fsxml >>>>>>>>>>>>> file that the t38_transcode.xml file its included ... but still get the >>>>>>>>>>>>> same error on on the console, so something terrible wrong I'm missing here >>>>>>>>>>>>> ... :-( >>>>>>>>>>>>> >>>>>>>>>>>>> Also ... it's possible to directly send the calls sofia/$1$2@$${external} >>>>>>>>>>>>> ? ... I mean, sending the call directly to a URI, with username/password >>>>>>>>>>>>> and so on ... instead of having to define a profile for it. >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> 2017-05-16 16:13 GMT+01:00 David Villasmil < >>>>>>>>>>>>> david.villasmil.work at gmail.com>: >>>>>>>>>>>>> >>>>>>>>>>>>>> No, you may be sending it to the right ports, but the profile >>>>>>>>>>>>>> attached to that port must have the context set to the correct dialplan. >>>>>>>>>>>>>> >>>>>>>>>>>>>> I just did exactly what you're doing and i also got "No >>>>>>>>>>>>>> Route, Aborting" >>>>>>>>>>>>>> >>>>>>>>>>>>>> I've never seen this type of "dialplan" value, tbh >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> Regards, >>>>>>>>>>>>>> >>>>>>>>>>>>>> David Villasmil >>>>>>>>>>>>>> email: david.villasmil.work at gmail.com >>>>>>>>>>>>>> phone: +34669448337 <+34%20669%2044%2083%2037> >>>>>>>>>>>>>> >>>>>>>>>>>>>> On Tue, May 16, 2017 at 3:30 PM, Ra?l Alexis Betancor Santana >>>>>>>>>>>>>> wrote: >>>>>>>>>>>>>> >>>>>>>>>>>>>>> Only one profile defined on the sofia.conf.xml and I'm >>>>>>>>>>>>>>> sending the traffict to the wright ports, if not I whould get the logs on >>>>>>>>>>>>>>> the console, as that are the only ports enabled. >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> 2017-05-16 14:22 GMT+01:00 David Villasmil < >>>>>>>>>>>>>>> david.villasmil.work at gmail.com>: >>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> Take a look at your profile, it should be listening on the >>>>>>>>>>>>>>>> port you're sending to, and must have the context parameter set to your >>>>>>>>>>>>>>>> dialplan name. >>>>>>>>>>>>>>>> On Tue, May 16, 2017 at 3:15 PM Ra?l Alexis Betancor >>>>>>>>>>>>>>>> Santana wrote: >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> Hi all, till now I'm been working with pre-made setup >>>>>>>>>>>>>>>>> files for FS, and now I'm trying to get a deep knowleadge of how the >>>>>>>>>>>>>>>>> dialplan works. >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> So I modifed my autoload_configs/sofia.conf.xml file and >>>>>>>>>>>>>>>>> changed my dialplan param to something like this: >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> The Idea is that it loads another .xml file especific task >>>>>>>>>>>>>>>>> and also use a dialplan throught a socket to a daemon that handle the rest. >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> On my t38_transcode.xml file ... very simple: >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> expresion="^True"/> >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> expresion="^(\+1|1)?([2-9]\d\d[2-9]\d{6})$"> >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> data="sofia/external/$1$2 at proxy.server.tld"/> >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> But If I fire a call to FS like [number]@[FS_IP] from a >>>>>>>>>>>>>>>>> sip testing client I get >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> [INFO] switch_core_state_machine.c:311 No Route, Aborting >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> What I'm doing wrong here? >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> My target its just I want to ANY call that came in with a >>>>>>>>>>>>>>>>> SIP header of X-T39-Transcode=True (market by a sip proxy elsewhere on the >>>>>>>>>>>>>>>>> net) just do a T38->Ulaw transcoding saving the T38 trace, so I could >>>>>>>>>>>>>>>>> inspect it later. >>>>>>>>>>>>>>>>> The rest of calls coming in ... as they don't have the >>>>>>>>>>>>>>>>> sip-header should end on other app. >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> For doing the testing I disabled the socket_inline part of >>>>>>>>>>>>>>>>> the dialplan string, so it have only this: >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> Did I miss something? ... or maybe missundestood who the >>>>>>>>>>>>>>>>> xml dialplan works ? >>>>>>>>>>>>>>>>> ______________________________ >>>>>>>>>>>>>>>>> ___________________________________________ >>>>>>>>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>>>>>>>> consulting at freeswitch.org >>>>>>>>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>>>>>>> http://confluence.freeswitch.org >>>>>>>>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>>>>>>>>>>> http://lists.freeswitch.org/ma >>>>>>>>>>>>>>>>> ilman/listinfo/freeswitch-users >>>>>>>>>>>>>>>>> UNSUBSCRIBE:http://lists.frees >>>>>>>>>>>>>>>>> witch.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/ma >>>>>>>>>>>>>>>> ilman/listinfo/freeswitch-users >>>>>>>>>>>>>>>> UNSUBSCRIBE:http://lists.frees >>>>>>>>>>>>>>>> witch.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-user >>>>>>>>>>>>>>> s >>>>>>>>>>>>>>> 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/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/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/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/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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170518/16f9b380/attachment-0001.html From rbetancor at gmail.com Thu May 18 12:46:37 2017 From: rbetancor at gmail.com (=?UTF-8?Q?Ra=C3=BAl_Alexis_Betancor_Santana?=) Date: Thu, 18 May 2017 09:46:37 +0100 Subject: [Freeswitch-users] Noob question about xml dialplan In-Reply-To: References: Message-ID: Sure ... This was what I have before on the freeswitch.xml and was not working:
This is what I have now and works perfectly:
2017-05-18 9:24 GMT+01:00 David Villasmil : > could you paste it? > > ? > > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 <+34%20669%2044%2083%2037> > > On Thu, May 18, 2017 at 10:20 AM, Ra?l Alexis Betancor Santana < > rbetancor at gmail.com> wrote: > >> Found it, and I supposed ... a noob fail ... X) ... I just was not adding >> the correct configuration statement on the freeswithc.xml file, and was >> loading the dialplan xml files inside the configuration section, instead of >> the dialplan section. >> >> 2017-05-18 8:22 GMT+01:00 Ra?l Alexis Betancor Santana < >> rbetancor at gmail.com>: >> >>> That command give me the clue ... if I do xml_locate dialplan it returns >>> me an empty response ... so I assume that I'm doing something wrong and >>> it's not loading any dialplan. >>> >>> 2017-05-16 19:48 GMT+01:00 David Villasmil < >>> david.villasmil.work at gmail.com>: >>> >>>> "xml_locate dialplan" >>>> >>>> will give you the whole xml for the dialplan >>>> >>>> "xml_locate configuration configuration name sofia.conf" >>>> >>>> will give you your sofia.conf >>>> ? >>>> >>>> Regards, >>>> >>>> David Villasmil >>>> email: david.villasmil.work at gmail.com >>>> phone: +34669448337 <+34%20669%2044%2083%2037> >>>> >>>> On Tue, May 16, 2017 at 8:15 PM, David Villasmil < >>>> david.villasmil.work at gmail.com> wrote: >>>> >>>>> Yeah that command doesn't exists. But there is another command, >>>>> xml_locate or something like that. >>>>> Remove the context param and try again. Post the latest version of the >>>>> extension with the section i pointed out before. >>>>> >>>>> I can't send you the fsxml as I'm not in my computer right now. I'll >>>>> send it in a few minutes. >>>>> >>>>> On Tue, May 16, 2017 at 8:12 PM Ra?l Alexis Betancor Santana < >>>>> rbetancor at gmail.com> wrote: >>>>> >>>>>> Thats what I'm trying ... to undestand how it works. >>>>>> >>>>>> I posted the profile I'm using a couple of messasges earlier, anyway: >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> value="generous"/> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> I changed the the extension pattern to ^(.*) ... still the same >>>>>> result >>>>>> >>>>>> Could you please share an example freeswitch.xml.fsxml file ? ... >>>>>> maybe the problem is how the includes are handled, so something is not on >>>>>> the xml node it needs to. >>>>>> >>>>>> I miss a command like the 'show dialplan' ... where you could check >>>>>> how an specific call would behalf >>>>>> >>>>>> >>>>>> 2017-05-16 19:01 GMT+01:00 David Villasmil < >>>>>> david.villasmil.work at gmail.com>: >>>>>> >>>>>>> Hello, >>>>>>> >>>>>>> You need to start by understanding freeswitch before comparing the >>>>>>> two, i work with both and there's just no comparison, as far as I'm >>>>>>> concerned. >>>>>>> >>>>>>> When you create a profile, this profile binds to a port. In the >>>>>>> profile config there's two params, context and diaplan (though tbh until >>>>>>> today i never ever touched the dialplan param, as i haven't had any need to >>>>>>> do it. If i need a simple diaplan, i just remove all xmls included in said >>>>>>> diaplan and have just one file. >>>>>>> >>>>>>> Now, the "context" param, along with the includes, is what fs uses >>>>>>> to hunt for extensions. Any file loaded from the profile which includes the >>>>>>> context name, will be use to hunt for an extension. >>>>>>> >>>>>>> You haven't posted your profile, so it's hard to figure out what >>>>>>> could be wrong, as when i tried it, it worked perfectly. I should note i >>>>>>> commented out the context param. >>>>>>> >>>>>>> For the sake of simplicity, can you change your extension to answer >>>>>>> EVERYTHING, (i.e.: "^(.*)") ? >>>>>>> >>>>>>> >>>>>>> >>>>>>> On Tue, May 16, 2017 at 7:53 PM Ra?l Alexis Betancor Santana < >>>>>>> rbetancor at gmail.com> wrote: >>>>>>> >>>>>>>> Just for the shake of simplicity ... on the Asterisk world would be >>>>>>>> something like: >>>>>>>> >>>>>>>> exten => _1NXXNXXXXXX,1,Noop("T38 Gateway Call") >>>>>>>> exten => _1NXXNXXXXXX,n,Set(FAXOPT(gateway)=yes) >>>>>>>> exten => _1NXXNXXXXXX,n,Dial(SIP/${EXTEN}@proxy.server.tld) >>>>>>>> >>>>>>>> I just want to make the same with FS, but not allowing T38 >>>>>>>> passthrought at all ... just T38 (a leg) to Ulaw (b leg) >>>>>>>> >>>>>>>> 2017-05-16 17:48 GMT+01:00 Ra?l Alexis Betancor Santana < >>>>>>>> rbetancor at gmail.com>: >>>>>>>> >>>>>>>>> Same result ... No Route, Aborting >>>>>>>>> >>>>>>>>> Lets see if I found some step-by-step tutorial that explains who >>>>>>>>> the dialplan works. The doc on the Wiki it's not clear enought for me. >>>>>>>>> >>>>>>>>> 2017-05-16 17:40 GMT+01:00 David Villasmil < >>>>>>>>> david.villasmil.work at gmail.com>: >>>>>>>>> >>>>>>>>>> That was my test extension, you can replace it for whatever you >>>>>>>>>> need :) >>>>>>>>>> ? >>>>>>>>>> >>>>>>>>>> Regards, >>>>>>>>>> >>>>>>>>>> David Villasmil >>>>>>>>>> email: david.villasmil.work at gmail.com >>>>>>>>>> phone: +34669448337 <+34%20669%2044%2083%2037> >>>>>>>>>> >>>>>>>>>> On Tue, May 16, 2017 at 6:38 PM, Ra?l Alexis Betancor Santana < >>>>>>>>>> rbetancor at gmail.com> wrote: >>>>>>>>>> >>>>>>>>>>> I don't get the point of calling a user/$1 ... as there is no >>>>>>>>>>> users registered against this FS ... I just want to use it as a T38->Ulaw >>>>>>>>>>> gateway. >>>>>>>>>>> >>>>>>>>>>> 2017-05-16 17:34 GMT+01:00 David Villasmil < >>>>>>>>>>> david.villasmil.work at gmail.com>: >>>>>>>>>>> >>>>>>>>>>>> that context line is not necessary, btw. point is you need the >>>>>>>>>>>> section: >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>>
>>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>>
>>>>>>>>>>>> ? >>>>>>>>>>>> >>>>>>>>>>>> Regards, >>>>>>>>>>>> >>>>>>>>>>>> David Villasmil >>>>>>>>>>>> email: david.villasmil.work at gmail.com >>>>>>>>>>>> phone: +34669448337 <+34%20669%2044%2083%2037> >>>>>>>>>>>> >>>>>>>>>>>> On Tue, May 16, 2017 at 6:33 PM, David Villasmil < >>>>>>>>>>>> david.villasmil.work at gmail.com> wrote: >>>>>>>>>>>> >>>>>>>>>>>>> Ok i've looked at the source code for mod_dialplan and in your >>>>>>>>>>>>> xml, you need something like: >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>>
>>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>> expression="^(10..)$"> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>>
>>>>>>>>>>>> ? >>>>>>>>>>>>> >>>>>>>>>>>>> Regards, >>>>>>>>>>>>> >>>>>>>>>>>>> David Villasmil >>>>>>>>>>>>> email: david.villasmil.work at gmail.com >>>>>>>>>>>>> phone: +34669448337 <+34%20669%2044%2083%2037> >>>>>>>>>>>>> >>>>>>>>>>>>> On Tue, May 16, 2017 at 5:26 PM, Ra?l Alexis Betancor Santana >>>>>>>>>>>>> wrote: >>>>>>>>>>>>> >>>>>>>>>>>>>> Ok, I think the whole point is that I don't undestand how the >>>>>>>>>>>>>> dialplan works. >>>>>>>>>>>>>> >>>>>>>>>>>>>> I started from 0 ... now on the sofia.conf.xml I have this: >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>> data="../gateways/*.xml"/> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>> value="udp:$${homer_ip}:$${homer_port};hep=3;capture_id=2001"/> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>> value="localnet.auto"/> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>> value="generous"/> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>> value="false"/> >>>>>>>>>>>>>> >>>>>>>>>>>>> value="false"/> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>> value="transport=tls"/> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>> value="$${base_dir}/conf/ssl"/> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> Then on the t38_transconde.xml file I have this. >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>> expresion="^(\+1|1)?([2-9]\d\d[2-9]\d{6})$"> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> I could see on the /var/log/freeswitch/freeswitch.xml.fsxml >>>>>>>>>>>>>> file that the t38_transcode.xml file its included ... but still get the >>>>>>>>>>>>>> same error on on the console, so something terrible wrong I'm missing here >>>>>>>>>>>>>> ... :-( >>>>>>>>>>>>>> >>>>>>>>>>>>>> Also ... it's possible to directly send the calls sofia/$1$2@$${external} >>>>>>>>>>>>>> ? ... I mean, sending the call directly to a URI, with username/password >>>>>>>>>>>>>> and so on ... instead of having to define a profile for it. >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> 2017-05-16 16:13 GMT+01:00 David Villasmil < >>>>>>>>>>>>>> david.villasmil.work at gmail.com>: >>>>>>>>>>>>>> >>>>>>>>>>>>>>> No, you may be sending it to the right ports, but the >>>>>>>>>>>>>>> profile attached to that port must have the context set to the correct >>>>>>>>>>>>>>> dialplan. >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> I just did exactly what you're doing and i also got "No >>>>>>>>>>>>>>> Route, Aborting" >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> I've never seen this type of "dialplan" value, tbh >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> Regards, >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> David Villasmil >>>>>>>>>>>>>>> email: david.villasmil.work at gmail.com >>>>>>>>>>>>>>> phone: +34669448337 <+34%20669%2044%2083%2037> >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> On Tue, May 16, 2017 at 3:30 PM, Ra?l Alexis Betancor >>>>>>>>>>>>>>> Santana wrote: >>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> Only one profile defined on the sofia.conf.xml and I'm >>>>>>>>>>>>>>>> sending the traffict to the wright ports, if not I whould get the logs on >>>>>>>>>>>>>>>> the console, as that are the only ports enabled. >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> 2017-05-16 14:22 GMT+01:00 David Villasmil < >>>>>>>>>>>>>>>> david.villasmil.work at gmail.com>: >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> Take a look at your profile, it should be listening on the >>>>>>>>>>>>>>>>> port you're sending to, and must have the context parameter set to your >>>>>>>>>>>>>>>>> dialplan name. >>>>>>>>>>>>>>>>> On Tue, May 16, 2017 at 3:15 PM Ra?l Alexis Betancor >>>>>>>>>>>>>>>>> Santana wrote: >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>> Hi all, till now I'm been working with pre-made setup >>>>>>>>>>>>>>>>>> files for FS, and now I'm trying to get a deep knowleadge of how the >>>>>>>>>>>>>>>>>> dialplan works. >>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>> So I modifed my autoload_configs/sofia.conf.xml file and >>>>>>>>>>>>>>>>>> changed my dialplan param to something like this: >>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>> The Idea is that it loads another .xml file especific >>>>>>>>>>>>>>>>>> task and also use a dialplan throught a socket to a daemon that handle the >>>>>>>>>>>>>>>>>> rest. >>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>> On my t38_transcode.xml file ... very simple: >>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> expresion="^True"/> >>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> expresion="^(\+1|1)?([2-9]\d\d[2-9]\d{6})$"> >>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> data="sip_execute_on_image=t38_gateway peer nocng"/> >>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> data="sofia/external/$1$2 at proxy.server.tld"/> >>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>> But If I fire a call to FS like [number]@[FS_IP] from a >>>>>>>>>>>>>>>>>> sip testing client I get >>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>> [INFO] switch_core_state_machine.c:311 No Route, Aborting >>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>> What I'm doing wrong here? >>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>> My target its just I want to ANY call that came in with a >>>>>>>>>>>>>>>>>> SIP header of X-T39-Transcode=True (market by a sip proxy elsewhere on the >>>>>>>>>>>>>>>>>> net) just do a T38->Ulaw transcoding saving the T38 trace, so I could >>>>>>>>>>>>>>>>>> inspect it later. >>>>>>>>>>>>>>>>>> The rest of calls coming in ... as they don't have the >>>>>>>>>>>>>>>>>> sip-header should end on other app. >>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>> For doing the testing I disabled the socket_inline part >>>>>>>>>>>>>>>>>> of the dialplan string, so it have only this: >>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>> Did I miss something? ... or maybe missundestood who the >>>>>>>>>>>>>>>>>> xml dialplan works ? >>>>>>>>>>>>>>>>>> ______________________________ >>>>>>>>>>>>>>>>>> ___________________________________________ >>>>>>>>>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>>>>>>>>> consulting at freeswitch.org >>>>>>>>>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>>>>>>>> http://confluence.freeswitch.org >>>>>>>>>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>>>>>>>>>>>> http://lists.freeswitch.org/ma >>>>>>>>>>>>>>>>>> ilman/listinfo/freeswitch-users >>>>>>>>>>>>>>>>>> UNSUBSCRIBE:http://lists.frees >>>>>>>>>>>>>>>>>> witch.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/ma >>>>>>>>>>>>>>>>> ilman/listinfo/freeswitch-users >>>>>>>>>>>>>>>>> UNSUBSCRIBE:http://lists.frees >>>>>>>>>>>>>>>>> witch.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/ma >>>>>>>>>>>>>>>> ilman/listinfo/freeswitch-users >>>>>>>>>>>>>>>> UNSUBSCRIBE:http://lists.frees >>>>>>>>>>>>>>>> witch.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-user >>>>>>>>>>>>>>> s >>>>>>>>>>>>>>> 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/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/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/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/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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170518/ab69a953/attachment-0001.html From ashwinrath at gmail.com Thu May 18 12:46:52 2017 From: ashwinrath at gmail.com (Ashwin Rath) Date: Thu, 18 May 2017 14:16:52 +0530 Subject: [Freeswitch-users] Conditional call forward In-Reply-To: References: Message-ID: Actually this related to fusion PBX. CF is configured on FS as a custom dial-string in the directory . The idea is to not forward if a call comes to a ring group but forward if it comes directly to the number. is there some way the dialplan can be modified to change the dial-string ? On 16 May 2017 at 22:20, Srigo Kana wrote: > Hi, > > Is the callforward configured on the phone? > If you get 302 redirect from a phone, you can jst catch it in a dialplan > and do whatever you want. > > Srigo > > Sent from my iPhone > > > On 13 May 2017, at 19:14, Ashwin Rath wrote: > > > > Hi > > > > I have an extension which has call forward setup BUT i would like the > call forward to work only when dialed from a certain number and not from > another numbers. Can this be achieved ? > > > > -- > > Ashwin Kumar Rath > > ____________________________________________________________ > _____________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 > -- Ashwin Kumar Rath -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170518/aefb3948/attachment.html From rbetancor at gmail.com Thu May 18 14:01:18 2017 From: rbetancor at gmail.com (=?UTF-8?Q?Ra=C3=BAl_Alexis_Betancor_Santana?=) Date: Thu, 18 May 2017 11:01:18 +0100 Subject: [Freeswitch-users] Dialplan conditions Message-ID: Now that I have solved the problem of loading the dialplan from the xml files ... I get to the point that it doesn't work as I expect. This is the only extension defined: I expect not to call the bridge app if there is no T38-Transcode header on the incoming INVITE request and if it is present ... that $1$2 will be filled ... but what I get on the traces is that any call fires the call to the bridge app and also that the dialled number is exactly $1$2 ... instead of the values they should have. I'm loading a minimal setup of modules, did I need to load someother one to get the conditions and the expresions to work on this simple dialplan? My modules.conf.xml file: Best regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170518/4a80d162/attachment.html From david.villasmil.work at gmail.com Thu May 18 16:59:42 2017 From: david.villasmil.work at gmail.com (David Villasmil) Date: Thu, 18 May 2017 14:59:42 +0200 Subject: [Freeswitch-users] Dialplan conditions In-Reply-To: References: Message-ID: http://lists.freeswitch.org/pipermail/freeswitch-users/2013-September/099883.html ? Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 On Thu, May 18, 2017 at 12:01 PM, Ra?l Alexis Betancor Santana < rbetancor at gmail.com> wrote: > > Now that I have solved the problem of loading the dialplan from the xml > files ... I get to the point that it doesn't work as I expect. > > This is the only extension defined: > > > > > > > > > > > > > > > I expect not to call the bridge app if there is no T38-Transcode header on > the incoming INVITE request and if it is present ... that $1$2 will be > filled ... but what I get on the traces is that any call fires the call to > the bridge app and also that the dialled number is exactly $1$2 ... instead > of the values they should have. > I'm loading a minimal setup of modules, did I need to load someother one > to get the conditions and the expresions to work on this simple dialplan? > > My modules.conf.xml file: > > > > > > > > > > > > > > > > > > > > > > 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170518/b04e3d65/attachment.html From rbetancor at gmail.com Thu May 18 17:30:03 2017 From: rbetancor at gmail.com (=?UTF-8?Q?Ra=C3=BAl_Alexis_Betancor_Santana?=) Date: Thu, 18 May 2017 14:30:03 +0100 Subject: [Freeswitch-users] Dialplan conditions In-Reply-To: References: Message-ID: But I used a logical AND, two conditions ... if the header is not there, so been consider emty, the condition should evaluate to false because there is no match ... and the app inside the condition should never been fired. 2017-05-18 13:59 GMT+01:00 David Villasmil : > http://lists.freeswitch.org/pipermail/freeswitch-users/ > 2013-September/099883.html > ? > > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 <+34%20669%2044%2083%2037> > > On Thu, May 18, 2017 at 12:01 PM, Ra?l Alexis Betancor Santana < > rbetancor at gmail.com> wrote: > >> >> Now that I have solved the problem of loading the dialplan from the xml >> files ... I get to the point that it doesn't work as I expect. >> >> This is the only extension defined: >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> I expect not to call the bridge app if there is no T38-Transcode header >> on the incoming INVITE request and if it is present ... that $1$2 will be >> filled ... but what I get on the traces is that any call fires the call to >> the bridge app and also that the dialled number is exactly $1$2 ... instead >> of the values they should have. >> I'm loading a minimal setup of modules, did I need to load someother one >> to get the conditions and the expresions to work on this simple dialplan? >> >> My modules.conf.xml file: >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170518/d2bc8713/attachment-0001.html From david.villasmil.work at gmail.com Thu May 18 18:38:41 2017 From: david.villasmil.work at gmail.com (David Villasmil) Date: Thu, 18 May 2017 16:38:41 +0200 Subject: [Freeswitch-users] Dialplan conditions In-Reply-To: References: Message-ID: "expression" is with 2 "s"... and that works for me ? Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 On Thu, May 18, 2017 at 12:01 PM, Ra?l Alexis Betancor Santana < rbetancor at gmail.com> wrote: > > Now that I have solved the problem of loading the dialplan from the xml > files ... I get to the point that it doesn't work as I expect. > > This is the only extension defined: > > > > > > > > > > > > > > > I expect not to call the bridge app if there is no T38-Transcode header on > the incoming INVITE request and if it is present ... that $1$2 will be > filled ... but what I get on the traces is that any call fires the call to > the bridge app and also that the dialled number is exactly $1$2 ... instead > of the values they should have. > I'm loading a minimal setup of modules, did I need to load someother one > to get the conditions and the expresions to work on this simple dialplan? > > My modules.conf.xml file: > > > > > > > > > > > > > > > > > > > > > > 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170518/2fc3e474/attachment.html From rbetancor at gmail.com Thu May 18 18:41:19 2017 From: rbetancor at gmail.com (=?UTF-8?Q?Ra=C3=BAl_Alexis_Betancor_Santana?=) Date: Thu, 18 May 2017 15:41:19 +0100 Subject: [Freeswitch-users] Dialplan conditions In-Reply-To: References: Message-ID: :-P 2017-05-18 15:38 GMT+01:00 David Villasmil : > "expression" is with 2 "s"... and that works for me > ? > > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 <+34%20669%2044%2083%2037> > > On Thu, May 18, 2017 at 12:01 PM, Ra?l Alexis Betancor Santana < > rbetancor at gmail.com> wrote: > >> >> Now that I have solved the problem of loading the dialplan from the xml >> files ... I get to the point that it doesn't work as I expect. >> >> This is the only extension defined: >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> I expect not to call the bridge app if there is no T38-Transcode header >> on the incoming INVITE request and if it is present ... that $1$2 will be >> filled ... but what I get on the traces is that any call fires the call to >> the bridge app and also that the dialled number is exactly $1$2 ... instead >> of the values they should have. >> I'm loading a minimal setup of modules, did I need to load someother one >> to get the conditions and the expresions to work on this simple dialplan? >> >> My modules.conf.xml file: >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170518/c9385fb1/attachment.html From rbetancor at gmail.com Thu May 18 18:41:49 2017 From: rbetancor at gmail.com (=?UTF-8?Q?Ra=C3=BAl_Alexis_Betancor_Santana?=) Date: Thu, 18 May 2017 15:41:49 +0100 Subject: [Freeswitch-users] Dialplan conditions In-Reply-To: References: Message-ID: I'll need to add the syntax check to my sublime editor 2017-05-18 15:41 GMT+01:00 Ra?l Alexis Betancor Santana : > :-P > > 2017-05-18 15:38 GMT+01:00 David Villasmil >: > >> "expression" is with 2 "s"... and that works for me >> ? >> >> Regards, >> >> David Villasmil >> email: david.villasmil.work at gmail.com >> phone: +34669448337 <+34%20669%2044%2083%2037> >> >> On Thu, May 18, 2017 at 12:01 PM, Ra?l Alexis Betancor Santana < >> rbetancor at gmail.com> wrote: >> >>> >>> Now that I have solved the problem of loading the dialplan from the xml >>> files ... I get to the point that it doesn't work as I expect. >>> >>> This is the only extension defined: >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> I expect not to call the bridge app if there is no T38-Transcode header >>> on the incoming INVITE request and if it is present ... that $1$2 will be >>> filled ... but what I get on the traces is that any call fires the call to >>> the bridge app and also that the dialled number is exactly $1$2 ... instead >>> of the values they should have. >>> I'm loading a minimal setup of modules, did I need to load someother one >>> to get the conditions and the expresions to work on this simple dialplan? >>> >>> My modules.conf.xml file: >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170518/871d0301/attachment-0001.html From joel at gogii.net Thu May 18 18:45:03 2017 From: joel at gogii.net (Joel Serrano) Date: Thu, 18 May 2017 07:45:03 -0700 Subject: [Freeswitch-users] Dialplan conditions In-Reply-To: References: Message-ID: Arrrrg, I see a typo more than a syntax problem... Does sublime have typo checkers? :P On Thu, May 18, 2017 at 7:41 AM, Ra?l Alexis Betancor Santana < rbetancor at gmail.com> wrote: > I'll need to add the syntax check to my sublime editor > > 2017-05-18 15:41 GMT+01:00 Ra?l Alexis Betancor Santana < > rbetancor at gmail.com>: > >> :-P >> >> 2017-05-18 15:38 GMT+01:00 David Villasmil > m>: >> >>> "expression" is with 2 "s"... and that works for me >>> ? >>> >>> Regards, >>> >>> David Villasmil >>> email: david.villasmil.work at gmail.com >>> phone: +34669448337 <+34%20669%2044%2083%2037> >>> >>> On Thu, May 18, 2017 at 12:01 PM, Ra?l Alexis Betancor Santana < >>> rbetancor at gmail.com> wrote: >>> >>>> >>>> Now that I have solved the problem of loading the dialplan from the xml >>>> files ... I get to the point that it doesn't work as I expect. >>>> >>>> This is the only extension defined: >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> I expect not to call the bridge app if there is no T38-Transcode header >>>> on the incoming INVITE request and if it is present ... that $1$2 will be >>>> filled ... but what I get on the traces is that any call fires the call to >>>> the bridge app and also that the dialled number is exactly $1$2 ... instead >>>> of the values they should have. >>>> I'm loading a minimal setup of modules, did I need to load someother >>>> one to get the conditions and the expresions to work on this simple >>>> dialplan? >>>> >>>> My modules.conf.xml file: >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> 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/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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170518/51c50da8/attachment.html From david.villasmil.work at gmail.com Thu May 18 18:49:08 2017 From: david.villasmil.work at gmail.com (David Villasmil) Date: Thu, 18 May 2017 14:49:08 +0000 Subject: [Freeswitch-users] Dialplan conditions In-Reply-To: References: Message-ID: You just need an english checker :) On Thu, May 18, 2017 at 4:46 PM Joel Serrano wrote: > Arrrrg, I see a typo more than a syntax problem... Does sublime have typo > checkers? :P > > On Thu, May 18, 2017 at 7:41 AM, Ra?l Alexis Betancor Santana < > rbetancor at gmail.com> wrote: > >> I'll need to add the syntax check to my sublime editor >> >> 2017-05-18 15:41 GMT+01:00 Ra?l Alexis Betancor Santana < >> rbetancor at gmail.com>: >> >>> :-P >>> >>> 2017-05-18 15:38 GMT+01:00 David Villasmil < >>> david.villasmil.work at gmail.com>: >>> >>>> "expression" is with 2 "s"... and that works for me >>>> ? >>>> >>>> Regards, >>>> >>>> David Villasmil >>>> email: david.villasmil.work at gmail.com >>>> phone: +34669448337 <+34%20669%2044%2083%2037> >>>> >>>> On Thu, May 18, 2017 at 12:01 PM, Ra?l Alexis Betancor Santana < >>>> rbetancor at gmail.com> wrote: >>>> >>>>> >>>>> Now that I have solved the problem of loading the dialplan from the >>>>> xml files ... I get to the point that it doesn't work as I expect. >>>>> >>>>> This is the only extension defined: >>>>> >>>>> >>>>> >>>>> >>>>> >>>> expresion="^(\+1|1)([2-9]\d\d[2-9]\d{6})$"> >>>>> >>>>> >>>>> >>>> data="sofia/external/$1$2 at proxy.server.tld"/> >>>>> >>>>> >>>>> >>>>> >>>>> I expect not to call the bridge app if there is no T38-Transcode >>>>> header on the incoming INVITE request and if it is present ... that $1$2 >>>>> will be filled ... but what I get on the traces is that any call fires the >>>>> call to the bridge app and also that the dialled number is exactly $1$2 ... >>>>> instead of the values they should have. >>>>> I'm loading a minimal setup of modules, did I need to load someother >>>>> one to get the conditions and the expresions to work on this simple >>>>> dialplan? >>>>> >>>>> My modules.conf.xml file: >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> 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 >>>> >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170518/c88461ac/attachment-0001.html From rbetancor at gmail.com Thu May 18 19:16:02 2017 From: rbetancor at gmail.com (=?UTF-8?Q?Ra=C3=BAl_Alexis_Betancor_Santana?=) Date: Thu, 18 May 2017 16:16:02 +0100 Subject: [Freeswitch-users] Dialplan conditions In-Reply-To: References: Message-ID: Could do it, if you have a completion plugging for the syntax file ... 2017-05-18 15:45 GMT+01:00 Joel Serrano : > Arrrrg, I see a typo more than a syntax problem... Does sublime have typo > checkers? :P > > On Thu, May 18, 2017 at 7:41 AM, Ra?l Alexis Betancor Santana < > rbetancor at gmail.com> wrote: > >> I'll need to add the syntax check to my sublime editor >> >> 2017-05-18 15:41 GMT+01:00 Ra?l Alexis Betancor Santana < >> rbetancor at gmail.com>: >> >>> :-P >>> >>> 2017-05-18 15:38 GMT+01:00 David Villasmil < >>> david.villasmil.work at gmail.com>: >>> >>>> "expression" is with 2 "s"... and that works for me >>>> ? >>>> >>>> Regards, >>>> >>>> David Villasmil >>>> email: david.villasmil.work at gmail.com >>>> phone: +34669448337 <+34%20669%2044%2083%2037> >>>> >>>> On Thu, May 18, 2017 at 12:01 PM, Ra?l Alexis Betancor Santana < >>>> rbetancor at gmail.com> wrote: >>>> >>>>> >>>>> Now that I have solved the problem of loading the dialplan from the >>>>> xml files ... I get to the point that it doesn't work as I expect. >>>>> >>>>> This is the only extension defined: >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> I expect not to call the bridge app if there is no T38-Transcode >>>>> header on the incoming INVITE request and if it is present ... that $1$2 >>>>> will be filled ... but what I get on the traces is that any call fires the >>>>> call to the bridge app and also that the dialled number is exactly $1$2 ... >>>>> instead of the values they should have. >>>>> I'm loading a minimal setup of modules, did I need to load someother >>>>> one to get the conditions and the expresions to work on this simple >>>>> dialplan? >>>>> >>>>> My modules.conf.xml file: >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> 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/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 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170518/787a7188/attachment.html From virtualguard2015 at gmail.com Thu May 18 03:02:52 2017 From: virtualguard2015 at gmail.com (Ravi sanyal) Date: Thu, 18 May 2017 11:02:52 +1200 Subject: [Freeswitch-users] Uninstalling freeswitch after compiling and installing from source Message-ID: Hi, I made a noob mistake and ran the script for installing freeswitch from binaries without doing logs on the process and fs_cli is complaining that commands are not found. specifically, sofia status, or sofia status profile but im sure that it extends further than that. I'm dont know how to get a list of the commands available. typing in help (not /help) at the prompt says it can't find a command. So i've installed it and it doesn't seem to be working, and don't know how to uninstall it. Where can i find a list of acceptable commands for the configure command? Im trying to enable sctp, but when looking through the logs for it you have a line: checking whether SCTP is supported... yes is this the same as checking with the --enable-sctp flag? (when looking at the logs for the added command there's no extra lines pertaining to enable-sctp as if you didn't specify it). I'm running the latest build from the github repository. When running this command: ./configure --enable-sctp > /home/fleetchat/config.log it keeps on displaying errors on output, but don't know if they're serious or not. ./configure: line 31222: php-config: command not found ./configure: line 31223: php-config: command not found ./configure: line 31224: php-config: command not found ./configure: line 31225: php: command not found ./configure: line 31226: php-config: command not found ./configure: line 31227: php-config: command not found ./configure: line 5306: AX_COMPILER_VENDOR: command not found rm: cannot remove 'conftest*': No such file or directory ./configure: line 19537: test: 3.16.0-4-amd64: integer expression expected ./configure: line 19819: test: 3.16.0-4-amd64: integer expression expected config.status: WARNING: 'apr-config.in' seems to ignore the --datarootdir setting rm: cannot remove 'libtoolT': No such file or directory config.status: WARNING: 'Makefile.in' seems to ignore the --datarootdir setting config.status: WARNING: 'lib/Makefile.in' seems to ignore the --datarootdir setting configure: WARNING: ** STUN support disabled ** Im sure it's somewhere but i also don't know where the configuration files should be located when building from source. i placed them in /etc/freeswitch. From memory that was where the installation instructions said they should be located. The configuration files were downloaded from the ubuntu repository command freeswitch-meta-vanilla The reason they come from the ubuntu repository is because i had freeswitch set up on ubuntu 16.04 but found out debian was the recommended OS. I'm running freeswitch on debian 8.8 now. Nathan -- *Virtual Guard Ltd* *info at virtualguard.co.nz * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170518/17faa2ba/attachment-0001.html From snehsach at rediffmail.com Wed May 17 20:56:07 2017 From: snehsach at rediffmail.com (sachin ) Date: 17 May 2017 16:56:07 -0000 Subject: [Freeswitch-users] =?utf-8?q?Changing_audio_DTLS_state_from_HANDS?= =?utf-8?q?HAKE_to_FAIL_issue?= Message-ID: <1495039928.S.72889.autosave.drafts.1495040167.30061@webmail.rediffmail.com> Hello Michael, Thanks for the information. Can you please let me know what was the client used? Also I am using self signed certificates on my local LAN. Thanks and RegardsSD From: Michael Jerris <mike at jerris.com> Sent: Fri, 12 May 2017 03:58:08 GMT+0530 To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> Subject: Re: [Freeswitch-users] Changing audio DTLS state from HANDSHAKE to FAIL issue we just had confirmed on Jira that 1.6.17 windows is working fine with latest chrome.  If you are getting errors still with something that is other than sipml5, please confirm details of what exactly the issue is.On May 8, 2017, at 8:08 AM, sachin <snehsach at rediffmail.com> wrote:Hello Michael,I tried using Sip.js instead of sipml5 and with Linux as the server. I have installed the FreeSWITCH Version 1.9.0+git~20170501T171230Z~e3ef041517~64bit (git e3ef041 2017-05-01 17:12:30Z 64bit)I am getting INCOMPATIBLE  DESTINATION error. Also tried sipml5.. with bot the client I am getting the same error.?Please let me know what I am missing.?Thanks and Regards,?SDFrom: Michael Jerris <mike at jerris.com>Sent: Fri, 05 May 2017 21:22:20To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>Subject: Re: [Freeswitch-users] Changing audio DTLS state from HANDSHAKE to FAIL issuecan you try this using sip.js or something thats confirmed to work when using linux as a server please.  This may just be an issue with sipml5. On May 5, 2017, at 9:53 AM, Brian West <brian at freeswitch.org> wrote: Clearly that isn't the case, I suspect a define is missing to actually enable ECDSA at this point. On Fri, May 5, 2017 at 8:47 AM, Gregor Nanger <gregor at infomedia.si> wrote:Brian, isn't this solved in 1.6.16? FS-10037 [core] Update OpenSSL to version 1.0.2k for Windows 2017-05-05 15:26 GMT+02:00 Brian West <brian at freeswitch.org>:OpenSSL on the windows build needs to be updated. https://freeswitch.org/jira/browse/FS-9510 On Thu, May 4, 2017 at 2:31 PM, Gregor Nanger <gregor at infomedia.si> wrote:This error is familiar to me, I think so, if I remembered correctly.  I am using windows. I had this error when client was Chrome. I just updated openssl dlls. I think that openssl library is updated on windows build now with version 1.6.17. And this version is also compiled on FS FTP to just download and install it. 2017-05-04 19:15 GMT+02:00 Anthony Minessale <anthony.minessale at gmail.com>:FS 1.5 sounds like a bad plan.Try latest FS 1.6 or master.  On Thu, May 4, 2017 at 12:04 PM, ?talo Rossi <italo at freeswitch.org> wrote:Old openssl version maybe?Em qui, 4 de mai de 2017 ?s 11:19, sachin <snehsach at rediffmail.com> escreveu:Hello All,?I am trying to setup the Webrtc call between 2 SIPML5 client. I am using FreeSWITCH Version 1.6.14 on Windows. I am using internal LAN.The clients are getting registered over wss. I have created self signed certificates.  In var.xml I have set the codecs setting as follows  <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMU,PCMA">  <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA">?I am able to establish the call and there is 2 way voice when I call from Sipmpl5 to Linphone and it works. But when I call from Linphone to Firefox (Simpl5) then the call is not getting established and I am getting following error2017-05-04 11:29:04.543434 [ERR] switch_rtp.c:2907 audio Handshake failure 12017-05-04 11:29:04.543434 [INFO] switch_rtp.c:2908 Changing audio DTLS state from HANDSHAKE to FAIL2017-05-04 11:29:04.543434 [NOTICE] switch_rtp.c:2889 Hangup sofia/internal/sip:1001 at df7jal23ls0d.invalid [CS_EXCHANGE_MEDIA] [DESTINATION_OUT_OF_ORDER]My setup is as followsSIPML5 (FireFox browser 53.0)<-----> FS (v1.5)(Windows) <----> LinphoneI am attaching the logs for the reference.fs-logleve9.txt : Debug trace with loglevel =9?fs-sip-trace.txt : Sip tracePlease let me know what could the issue and pointers to resolve the same.?Thanks and RegardsSD _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170517/80cfee72/attachment.html From michael at mailworks.org Tue May 16 23:03:35 2017 From: michael at mailworks.org (Michael Avers) Date: Tue, 16 May 2017 12:03:35 -0700 Subject: [Freeswitch-users] mod_xml_curl performance questions Message-ID: <1494961415.1484944.978717552.3EE447BC@webmail.messagingengine.com> Hello, I'm planning on moving all our local lua-generated XML to use mod_xml_curl over HTTPS. However, the primary http server is not in the same data center as our FS instances. Ping time is about 50ms. Is this going to be an issue? Is it typical to run xml_curl hosted elsewhere? The web app itself is written in Go and is very fast. Just wondering about network latency aspect of it all. Thanks -Mike From kk-mailinglist at ednt.de Thu May 18 17:42:25 2017 From: kk-mailinglist at ednt.de (kk Mailinglist) Date: Thu, 18 May 2017 15:42:25 +0200 Subject: [Freeswitch-users] Busy Message-ID: <6075eed2-9e19-080a-87a3-f4f98fca9f11@ednt.de> When a phone have a running call and a second call comes in i get no busy and i have redirect to voicemail by busy and the new incoming call will not redirect to VM. I am aware that my phone can have multiple Lines, but how i can signal that i am busy ? regards From agubbe at gmail.com Thu May 18 11:15:46 2017 From: agubbe at gmail.com (=?UTF-8?Q?Agust=C3=AD_Ubalde_Bellot?=) Date: Thu, 18 May 2017 09:15:46 +0200 Subject: [Freeswitch-users] Verto calls Message-ID: Hi all, I am trying to call from Verto extension to another Verto extension. *Both are successfully registered* (Verto status show the successfully register) but the call between is not established. The call remains in ring state. This is the last dialplan function executed (calling 1000 to 1001): *EXECUTE verto.rtc/1001 bridge()* Regards, Tineli -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170518/9b3d5445/attachment.html From alfonso.pinto at gmail.com Wed May 17 00:11:52 2017 From: alfonso.pinto at gmail.com (Alfonso Pinto) Date: Tue, 16 May 2017 22:11:52 +0200 Subject: [Freeswitch-users] Noob question about xml dialplan In-Reply-To: References: Message-ID: It's difficult to give an answer without seeing more information. Do you have other profiles? do you have other dialplans? I see a couple of strange things in your mails, first you try to to specify port 8022 in local ip using this line: Later you modify it to remove the port from there and put it in your profile, but in your profile you are using 5060 on $${sofia_ip}. Can you confirm where FS is listening using netstat -putan? Are you sending the traffic to the IP:PORT indicated by the previous command? If you put in your profile: It means FS will expect a dialplan that has inside If you didn't change the default config, your file t38_transcode.xml should be loaded automatically and the context incoming interpreted correctly As David mentioned, with xml_locate dialplan you can see if its loaded. Regards, Alfonso On Tue, May 16, 2017 at 8:48 PM, David Villasmil < david.villasmil.work at gmail.com> wrote: > "xml_locate dialplan" > > will give you the whole xml for the dialplan > > "xml_locate configuration configuration name sofia.conf" > > will give you your sofia.conf > ? > > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 <+34%20669%2044%2083%2037> > > On Tue, May 16, 2017 at 8:15 PM, David Villasmil < > david.villasmil.work at gmail.com> wrote: > >> Yeah that command doesn't exists. But there is another command, >> xml_locate or something like that. >> Remove the context param and try again. Post the latest version of the >> extension with the section i pointed out before. >> >> I can't send you the fsxml as I'm not in my computer right now. I'll send >> it in a few minutes. >> >> On Tue, May 16, 2017 at 8:12 PM Ra?l Alexis Betancor Santana < >> rbetancor at gmail.com> wrote: >> >>> Thats what I'm trying ... to undestand how it works. >>> >>> I posted the profile I'm using a couple of messasges earlier, anyway: >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> I changed the the extension pattern to ^(.*) ... still the same result >>> >>> Could you please share an example freeswitch.xml.fsxml file ? ... maybe >>> the problem is how the includes are handled, so something is not on the xml >>> node it needs to. >>> >>> I miss a command like the 'show dialplan' ... where you could check how >>> an specific call would behalf >>> >>> >>> 2017-05-16 19:01 GMT+01:00 David Villasmil < >>> david.villasmil.work at gmail.com>: >>> >>>> Hello, >>>> >>>> You need to start by understanding freeswitch before comparing the two, >>>> i work with both and there's just no comparison, as far as I'm concerned. >>>> >>>> When you create a profile, this profile binds to a port. In the profile >>>> config there's two params, context and diaplan (though tbh until today i >>>> never ever touched the dialplan param, as i haven't had any need to do it. >>>> If i need a simple diaplan, i just remove all xmls included in said diaplan >>>> and have just one file. >>>> >>>> Now, the "context" param, along with the includes, is what fs uses to >>>> hunt for extensions. Any file loaded from the profile which includes the >>>> context name, will be use to hunt for an extension. >>>> >>>> You haven't posted your profile, so it's hard to figure out what could >>>> be wrong, as when i tried it, it worked perfectly. I should note i >>>> commented out the context param. >>>> >>>> For the sake of simplicity, can you change your extension to answer >>>> EVERYTHING, (i.e.: "^(.*)") ? >>>> >>>> >>>> >>>> On Tue, May 16, 2017 at 7:53 PM Ra?l Alexis Betancor Santana < >>>> rbetancor at gmail.com> wrote: >>>> >>>>> Just for the shake of simplicity ... on the Asterisk world would be >>>>> something like: >>>>> >>>>> exten => _1NXXNXXXXXX,1,Noop("T38 Gateway Call") >>>>> exten => _1NXXNXXXXXX,n,Set(FAXOPT(gateway)=yes) >>>>> exten => _1NXXNXXXXXX,n,Dial(SIP/${EXTEN}@proxy.server.tld) >>>>> >>>>> I just want to make the same with FS, but not allowing T38 >>>>> passthrought at all ... just T38 (a leg) to Ulaw (b leg) >>>>> >>>>> 2017-05-16 17:48 GMT+01:00 Ra?l Alexis Betancor Santana < >>>>> rbetancor at gmail.com>: >>>>> >>>>>> Same result ... No Route, Aborting >>>>>> >>>>>> Lets see if I found some step-by-step tutorial that explains who the >>>>>> dialplan works. The doc on the Wiki it's not clear enought for me. >>>>>> >>>>>> 2017-05-16 17:40 GMT+01:00 David Villasmil < >>>>>> david.villasmil.work at gmail.com>: >>>>>> >>>>>>> That was my test extension, you can replace it for whatever you need >>>>>>> :) >>>>>>> ? >>>>>>> >>>>>>> Regards, >>>>>>> >>>>>>> David Villasmil >>>>>>> email: david.villasmil.work at gmail.com >>>>>>> phone: +34669448337 <+34%20669%2044%2083%2037> >>>>>>> >>>>>>> On Tue, May 16, 2017 at 6:38 PM, Ra?l Alexis Betancor Santana < >>>>>>> rbetancor at gmail.com> wrote: >>>>>>> >>>>>>>> I don't get the point of calling a user/$1 ... as there is no users >>>>>>>> registered against this FS ... I just want to use it as a T38->Ulaw gateway. >>>>>>>> >>>>>>>> 2017-05-16 17:34 GMT+01:00 David Villasmil < >>>>>>>> david.villasmil.work at gmail.com>: >>>>>>>> >>>>>>>>> that context line is not necessary, btw. point is you need the >>>>>>>>> section: >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>>
>>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>>
>>>>>>>>> ? >>>>>>>>> >>>>>>>>> Regards, >>>>>>>>> >>>>>>>>> David Villasmil >>>>>>>>> email: david.villasmil.work at gmail.com >>>>>>>>> phone: +34669448337 <+34%20669%2044%2083%2037> >>>>>>>>> >>>>>>>>> On Tue, May 16, 2017 at 6:33 PM, David Villasmil < >>>>>>>>> david.villasmil.work at gmail.com> wrote: >>>>>>>>> >>>>>>>>>> Ok i've looked at the source code for mod_dialplan and in your >>>>>>>>>> xml, you need something like: >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>>
>>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>>
>>>>>>>>> ? >>>>>>>>>> >>>>>>>>>> Regards, >>>>>>>>>> >>>>>>>>>> David Villasmil >>>>>>>>>> email: david.villasmil.work at gmail.com >>>>>>>>>> phone: +34669448337 <+34%20669%2044%2083%2037> >>>>>>>>>> >>>>>>>>>> On Tue, May 16, 2017 at 5:26 PM, Ra?l Alexis Betancor Santana < >>>>>>>>>> rbetancor at gmail.com> wrote: >>>>>>>>>> >>>>>>>>>>> Ok, I think the whole point is that I don't undestand how the >>>>>>>>>>> dialplan works. >>>>>>>>>>> >>>>>>>>>>> I started from 0 ... now on the sofia.conf.xml I have this: >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>> value="udp:$${homer_ip}:$${homer_port};hep=3;capture_id=2001"/> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>> value="localnet.auto"/> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>> value="generous"/> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>> value="false"/> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>> value="$${base_dir}/conf/ssl"/> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> Then on the t38_transconde.xml file I have this. >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>> expresion="^(\+1|1)?([2-9]\d\d[2-9]\d{6})$"> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> I could see on the /var/log/freeswitch/freeswitch.xml.fsxml >>>>>>>>>>> file that the t38_transcode.xml file its included ... but still get the >>>>>>>>>>> same error on on the console, so something terrible wrong I'm missing here >>>>>>>>>>> ... :-( >>>>>>>>>>> >>>>>>>>>>> Also ... it's possible to directly send the calls sofia/$1$2@$${external} >>>>>>>>>>> ? ... I mean, sending the call directly to a URI, with username/password >>>>>>>>>>> and so on ... instead of having to define a profile for it. >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> 2017-05-16 16:13 GMT+01:00 David Villasmil < >>>>>>>>>>> david.villasmil.work at gmail.com>: >>>>>>>>>>> >>>>>>>>>>>> No, you may be sending it to the right ports, but the profile >>>>>>>>>>>> attached to that port must have the context set to the correct dialplan. >>>>>>>>>>>> >>>>>>>>>>>> I just did exactly what you're doing and i also got "No Route, >>>>>>>>>>>> Aborting" >>>>>>>>>>>> >>>>>>>>>>>> I've never seen this type of "dialplan" value, tbh >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> Regards, >>>>>>>>>>>> >>>>>>>>>>>> David Villasmil >>>>>>>>>>>> email: david.villasmil.work at gmail.com >>>>>>>>>>>> phone: +34669448337 <+34%20669%2044%2083%2037> >>>>>>>>>>>> >>>>>>>>>>>> On Tue, May 16, 2017 at 3:30 PM, Ra?l Alexis Betancor Santana < >>>>>>>>>>>> rbetancor at gmail.com> wrote: >>>>>>>>>>>> >>>>>>>>>>>>> Only one profile defined on the sofia.conf.xml and I'm sending >>>>>>>>>>>>> the traffict to the wright ports, if not I whould get the logs on the >>>>>>>>>>>>> console, as that are the only ports enabled. >>>>>>>>>>>>> >>>>>>>>>>>>> 2017-05-16 14:22 GMT+01:00 David Villasmil < >>>>>>>>>>>>> david.villasmil.work at gmail.com>: >>>>>>>>>>>>> >>>>>>>>>>>>>> Take a look at your profile, it should be listening on the >>>>>>>>>>>>>> port you're sending to, and must have the context parameter set to your >>>>>>>>>>>>>> dialplan name. >>>>>>>>>>>>>> On Tue, May 16, 2017 at 3:15 PM Ra?l Alexis Betancor Santana < >>>>>>>>>>>>>> rbetancor at gmail.com> wrote: >>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> Hi all, till now I'm been working with pre-made setup files >>>>>>>>>>>>>>> for FS, and now I'm trying to get a deep knowleadge of how the dialplan >>>>>>>>>>>>>>> works. >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> So I modifed my autoload_configs/sofia.conf.xml file and >>>>>>>>>>>>>>> changed my dialplan param to something like this: >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> The Idea is that it loads another .xml file especific task >>>>>>>>>>>>>>> and also use a dialplan throught a socket to a daemon that handle the rest. >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> On my t38_transcode.xml file ... very simple: >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>> expresion="^True"/> >>>>>>>>>>>>>>> >>>>>>>>>>>>>> expresion="^(\+1|1)?([2-9]\d\d[2-9]\d{6})$"> >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>> data="sofia/external/$1$2 at proxy.server.tld"/> >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> But If I fire a call to FS like [number]@[FS_IP] from a sip >>>>>>>>>>>>>>> testing client I get >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> [INFO] switch_core_state_machine.c:311 No Route, Aborting >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> What I'm doing wrong here? >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> My target its just I want to ANY call that came in with a >>>>>>>>>>>>>>> SIP header of X-T39-Transcode=True (market by a sip proxy elsewhere on the >>>>>>>>>>>>>>> net) just do a T38->Ulaw transcoding saving the T38 trace, so I could >>>>>>>>>>>>>>> inspect it later. >>>>>>>>>>>>>>> The rest of calls coming in ... as they don't have the >>>>>>>>>>>>>>> sip-header should end on other app. >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> For doing the testing I disabled the socket_inline part of >>>>>>>>>>>>>>> the dialplan string, so it have only this: >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> Did I miss something? ... or maybe missundestood who the xml >>>>>>>>>>>>>>> dialplan works ? >>>>>>>>>>>>>>> ____________________________________________________________ >>>>>>>>>>>>>>> _____________ >>>>>>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>>>>>> consulting at freeswitch.org >>>>>>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>>>>> http://confluence.freeswitch.org >>>>>>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-user >>>>>>>>>>>>>>> s >>>>>>>>>>>>>>> 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 >>>>>>> >>>>>> >>>>>> >>>>> ____________________________________________________________ >>>>> _____________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170516/65a96630/attachment-0001.html From alfonso.pinto at gmail.com Thu May 18 11:44:09 2017 From: alfonso.pinto at gmail.com (Alfonso Pinto) Date: Thu, 18 May 2017 00:44:09 -0700 Subject: [Freeswitch-users] Noob question about xml dialplan In-Reply-To: References: Message-ID: Do you have this line in autoload_configs/modules.conf.xml: And these in freeswitch.xml:
With that it should load all your dialplans On Thu, May 18, 2017 at 12:22 AM, Ra?l Alexis Betancor Santana < rbetancor at gmail.com> wrote: > That command give me the clue ... if I do xml_locate dialplan it returns > me an empty response ... so I assume that I'm doing something wrong and > it's not loading any dialplan. > > 2017-05-16 19:48 GMT+01:00 David Villasmil >: > >> "xml_locate dialplan" >> >> will give you the whole xml for the dialplan >> >> "xml_locate configuration configuration name sofia.conf" >> >> will give you your sofia.conf >> ? >> >> Regards, >> >> David Villasmil >> email: david.villasmil.work at gmail.com >> phone: +34669448337 <+34%20669%2044%2083%2037> >> >> On Tue, May 16, 2017 at 8:15 PM, David Villasmil < >> david.villasmil.work at gmail.com> wrote: >> >>> Yeah that command doesn't exists. But there is another command, >>> xml_locate or something like that. >>> Remove the context param and try again. Post the latest version of the >>> extension with the section i pointed out before. >>> >>> I can't send you the fsxml as I'm not in my computer right now. I'll >>> send it in a few minutes. >>> >>> On Tue, May 16, 2017 at 8:12 PM Ra?l Alexis Betancor Santana < >>> rbetancor at gmail.com> wrote: >>> >>>> Thats what I'm trying ... to undestand how it works. >>>> >>>> I posted the profile I'm using a couple of messasges earlier, anyway: >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> I changed the the extension pattern to ^(.*) ... still the same result >>>> >>>> Could you please share an example freeswitch.xml.fsxml file ? ... maybe >>>> the problem is how the includes are handled, so something is not on the xml >>>> node it needs to. >>>> >>>> I miss a command like the 'show dialplan' ... where you could check how >>>> an specific call would behalf >>>> >>>> >>>> 2017-05-16 19:01 GMT+01:00 David Villasmil < >>>> david.villasmil.work at gmail.com>: >>>> >>>>> Hello, >>>>> >>>>> You need to start by understanding freeswitch before comparing the >>>>> two, i work with both and there's just no comparison, as far as I'm >>>>> concerned. >>>>> >>>>> When you create a profile, this profile binds to a port. In the >>>>> profile config there's two params, context and diaplan (though tbh until >>>>> today i never ever touched the dialplan param, as i haven't had any need to >>>>> do it. If i need a simple diaplan, i just remove all xmls included in said >>>>> diaplan and have just one file. >>>>> >>>>> Now, the "context" param, along with the includes, is what fs uses to >>>>> hunt for extensions. Any file loaded from the profile which includes the >>>>> context name, will be use to hunt for an extension. >>>>> >>>>> You haven't posted your profile, so it's hard to figure out what could >>>>> be wrong, as when i tried it, it worked perfectly. I should note i >>>>> commented out the context param. >>>>> >>>>> For the sake of simplicity, can you change your extension to answer >>>>> EVERYTHING, (i.e.: "^(.*)") ? >>>>> >>>>> >>>>> >>>>> On Tue, May 16, 2017 at 7:53 PM Ra?l Alexis Betancor Santana < >>>>> rbetancor at gmail.com> wrote: >>>>> >>>>>> Just for the shake of simplicity ... on the Asterisk world would be >>>>>> something like: >>>>>> >>>>>> exten => _1NXXNXXXXXX,1,Noop("T38 Gateway Call") >>>>>> exten => _1NXXNXXXXXX,n,Set(FAXOPT(gateway)=yes) >>>>>> exten => _1NXXNXXXXXX,n,Dial(SIP/${EXTEN}@proxy.server.tld) >>>>>> >>>>>> I just want to make the same with FS, but not allowing T38 >>>>>> passthrought at all ... just T38 (a leg) to Ulaw (b leg) >>>>>> >>>>>> 2017-05-16 17:48 GMT+01:00 Ra?l Alexis Betancor Santana < >>>>>> rbetancor at gmail.com>: >>>>>> >>>>>>> Same result ... No Route, Aborting >>>>>>> >>>>>>> Lets see if I found some step-by-step tutorial that explains who the >>>>>>> dialplan works. The doc on the Wiki it's not clear enought for me. >>>>>>> >>>>>>> 2017-05-16 17:40 GMT+01:00 David Villasmil < >>>>>>> david.villasmil.work at gmail.com>: >>>>>>> >>>>>>>> That was my test extension, you can replace it for whatever you >>>>>>>> need :) >>>>>>>> ? >>>>>>>> >>>>>>>> Regards, >>>>>>>> >>>>>>>> David Villasmil >>>>>>>> email: david.villasmil.work at gmail.com >>>>>>>> phone: +34669448337 <+34%20669%2044%2083%2037> >>>>>>>> >>>>>>>> On Tue, May 16, 2017 at 6:38 PM, Ra?l Alexis Betancor Santana < >>>>>>>> rbetancor at gmail.com> wrote: >>>>>>>> >>>>>>>>> I don't get the point of calling a user/$1 ... as there is no >>>>>>>>> users registered against this FS ... I just want to use it as a T38->Ulaw >>>>>>>>> gateway. >>>>>>>>> >>>>>>>>> 2017-05-16 17:34 GMT+01:00 David Villasmil < >>>>>>>>> david.villasmil.work at gmail.com>: >>>>>>>>> >>>>>>>>>> that context line is not necessary, btw. point is you need the >>>>>>>>>> section: >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>>
>>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>>
>>>>>>>>>> ? >>>>>>>>>> >>>>>>>>>> Regards, >>>>>>>>>> >>>>>>>>>> David Villasmil >>>>>>>>>> email: david.villasmil.work at gmail.com >>>>>>>>>> phone: +34669448337 <+34%20669%2044%2083%2037> >>>>>>>>>> >>>>>>>>>> On Tue, May 16, 2017 at 6:33 PM, David Villasmil < >>>>>>>>>> david.villasmil.work at gmail.com> wrote: >>>>>>>>>> >>>>>>>>>>> Ok i've looked at the source code for mod_dialplan and in your >>>>>>>>>>> xml, you need something like: >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>>
>>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>>
>>>>>>>>>> ? >>>>>>>>>>> >>>>>>>>>>> Regards, >>>>>>>>>>> >>>>>>>>>>> David Villasmil >>>>>>>>>>> email: david.villasmil.work at gmail.com >>>>>>>>>>> phone: +34669448337 <+34%20669%2044%2083%2037> >>>>>>>>>>> >>>>>>>>>>> On Tue, May 16, 2017 at 5:26 PM, Ra?l Alexis Betancor Santana < >>>>>>>>>>> rbetancor at gmail.com> wrote: >>>>>>>>>>> >>>>>>>>>>>> Ok, I think the whole point is that I don't undestand how the >>>>>>>>>>>> dialplan works. >>>>>>>>>>>> >>>>>>>>>>>> I started from 0 ... now on the sofia.conf.xml I have this: >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>> data="../gateways/*.xml"/> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>> value="udp:$${homer_ip}:$${homer_port};hep=3;capture_id=2001"/> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>> value="localnet.auto"/> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>> value="generous"/> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>> value="false"/> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>> value="transport=tls"/> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>> value="$${base_dir}/conf/ssl"/> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> Then on the t38_transconde.xml file I have this. >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>> expresion="^(\+1|1)?([2-9]\d\d[2-9]\d{6})$"> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> I could see on the /var/log/freeswitch/freeswitch.xml.fsxml >>>>>>>>>>>> file that the t38_transcode.xml file its included ... but still get the >>>>>>>>>>>> same error on on the console, so something terrible wrong I'm missing here >>>>>>>>>>>> ... :-( >>>>>>>>>>>> >>>>>>>>>>>> Also ... it's possible to directly send the calls sofia/$1$2@$${external} >>>>>>>>>>>> ? ... I mean, sending the call directly to a URI, with username/password >>>>>>>>>>>> and so on ... instead of having to define a profile for it. >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> 2017-05-16 16:13 GMT+01:00 David Villasmil < >>>>>>>>>>>> david.villasmil.work at gmail.com>: >>>>>>>>>>>> >>>>>>>>>>>>> No, you may be sending it to the right ports, but the profile >>>>>>>>>>>>> attached to that port must have the context set to the correct dialplan. >>>>>>>>>>>>> >>>>>>>>>>>>> I just did exactly what you're doing and i also got "No Route, >>>>>>>>>>>>> Aborting" >>>>>>>>>>>>> >>>>>>>>>>>>> I've never seen this type of "dialplan" value, tbh >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> Regards, >>>>>>>>>>>>> >>>>>>>>>>>>> David Villasmil >>>>>>>>>>>>> email: david.villasmil.work at gmail.com >>>>>>>>>>>>> phone: +34669448337 <+34%20669%2044%2083%2037> >>>>>>>>>>>>> >>>>>>>>>>>>> On Tue, May 16, 2017 at 3:30 PM, Ra?l Alexis Betancor Santana >>>>>>>>>>>>> wrote: >>>>>>>>>>>>> >>>>>>>>>>>>>> Only one profile defined on the sofia.conf.xml and I'm >>>>>>>>>>>>>> sending the traffict to the wright ports, if not I whould get the logs on >>>>>>>>>>>>>> the console, as that are the only ports enabled. >>>>>>>>>>>>>> >>>>>>>>>>>>>> 2017-05-16 14:22 GMT+01:00 David Villasmil < >>>>>>>>>>>>>> david.villasmil.work at gmail.com>: >>>>>>>>>>>>>> >>>>>>>>>>>>>>> Take a look at your profile, it should be listening on the >>>>>>>>>>>>>>> port you're sending to, and must have the context parameter set to your >>>>>>>>>>>>>>> dialplan name. >>>>>>>>>>>>>>> On Tue, May 16, 2017 at 3:15 PM Ra?l Alexis Betancor Santana >>>>>>>>>>>>>>> wrote: >>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> Hi all, till now I'm been working with pre-made setup files >>>>>>>>>>>>>>>> for FS, and now I'm trying to get a deep knowleadge of how the dialplan >>>>>>>>>>>>>>>> works. >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> So I modifed my autoload_configs/sofia.conf.xml file and >>>>>>>>>>>>>>>> changed my dialplan param to something like this: >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> The Idea is that it loads another .xml file especific task >>>>>>>>>>>>>>>> and also use a dialplan throught a socket to a daemon that handle the rest. >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> On my t38_transcode.xml file ... very simple: >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>> expresion="^True"/> >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>> expresion="^(\+1|1)?([2-9]\d\d[2-9]\d{6})$"> >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>> data="sofia/external/$1$2 at proxy.server.tld"/> >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> But If I fire a call to FS like [number]@[FS_IP] from a sip >>>>>>>>>>>>>>>> testing client I get >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> [INFO] switch_core_state_machine.c:311 No Route, Aborting >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> What I'm doing wrong here? >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> My target its just I want to ANY call that came in with a >>>>>>>>>>>>>>>> SIP header of X-T39-Transcode=True (market by a sip proxy elsewhere on the >>>>>>>>>>>>>>>> net) just do a T38->Ulaw transcoding saving the T38 trace, so I could >>>>>>>>>>>>>>>> inspect it later. >>>>>>>>>>>>>>>> The rest of calls coming in ... as they don't have the >>>>>>>>>>>>>>>> sip-header should end on other app. >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> For doing the testing I disabled the socket_inline part of >>>>>>>>>>>>>>>> the dialplan string, so it have only this: >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> Did I miss something? ... or maybe missundestood who the >>>>>>>>>>>>>>>> xml dialplan works ? >>>>>>>>>>>>>>>> ______________________________ >>>>>>>>>>>>>>>> ___________________________________________ >>>>>>>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>>>>>>> consulting at freeswitch.org >>>>>>>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>>>>>> http://confluence.freeswitch.org >>>>>>>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>>>>>>>>>> http://lists.freeswitch.org/ma >>>>>>>>>>>>>>>> ilman/listinfo/freeswitch-users >>>>>>>>>>>>>>>> UNSUBSCRIBE:http://lists.frees >>>>>>>>>>>>>>>> witch.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-user >>>>>>>>>>>>>>> s >>>>>>>>>>>>>>> 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/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/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/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/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 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170518/d2a28fc5/attachment-0001.html From alfonso.pinto at gmail.com Thu May 18 16:07:57 2017 From: alfonso.pinto at gmail.com (Alfonso Pinto) Date: Thu, 18 May 2017 14:07:57 +0200 Subject: [Freeswitch-users] Dialplan conditions In-Reply-To: References: Message-ID: add and check if the channel variable is setup always. Regards, Alfonso. On Thu, May 18, 2017 at 12:01 PM, Ra?l Alexis Betancor Santana wrote: > > Now that I have solved the problem of loading the dialplan from the xml > files ... I get to the point that it doesn't work as I expect. > > This is the only extension defined: > > > > > expresion="^(\+1|1)([2-9]\d\d[2-9]\d{6})$"> > > > data="sofia/external/$1$2 at proxy.server.tld"/> > > > > > I expect not to call the bridge app if there is no T38-Transcode header on > the incoming INVITE request and if it is present ... that $1$2 will be > filled ... but what I get on the traces is that any call fires the call to > the bridge app and also that the dialled number is exactly $1$2 ... instead > of the values they should have. > I'm loading a minimal setup of modules, did I need to load someother one to > get the conditions and the expresions to work on this simple dialplan? > > My modules.conf.xml file: > > > > > > > > > > > > > > > > > > > > > > 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 From mike at jerris.com Thu May 18 20:17:30 2017 From: mike at jerris.com (Michael Jerris) Date: Thu, 18 May 2017 12:17:30 -0400 Subject: [Freeswitch-users] Callcenter modul error:: register subclass In-Reply-To: References: Message-ID: <13C9308F-F688-4A24-8175-76E761CAB9AB@jerris.com> Thats strange, that means some other module already has that loaded? did you do something strange like copy the module and try to load it again? > On May 18, 2017, at 3:51 AM, Gregor Nanger wrote: > > ?Can anyone help with error that I get when try to load callcenter module: > > 2017-05-18 09:48:49.828297 [ERR] mod_callcenter.c:3810 Couldn't register subclass callcenter::info! > 2017-05-18 09:48:49.828297 [CRIT] switch_loadable_module.c:1522 Error Loading module /usr/lib/freeswitch/mod/mod_callcenter.so > **Module load routine returned an error**? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170518/a3cb5daf/attachment.html From mike at jerris.com Thu May 18 20:19:03 2017 From: mike at jerris.com (Michael Jerris) Date: Thu, 18 May 2017 12:19:03 -0400 Subject: [Freeswitch-users] Changing audio DTLS state from HANDSHAKE to FAIL issue In-Reply-To: <1495039928.S.72889.autosave.drafts.1495040167.30061@webmail.rediffmail.com> References: <1495039928.S.72889.autosave.drafts.1495040167.30061@webmail.rediffmail.com> Message-ID: <84A60AB0-C470-48A3-8552-56550785B93D@jerris.com> I can?t specifically? I know i?ve had luck with sip.js in the past. > On May 17, 2017, at 12:56 PM, sachin wrote: > > Hello Michael, > > Thanks for the information. Can you please let me know what was the client used? Also I am using self signed certificates on my local LAN. > > > Thanks and Regards > SD > > > > > > From: Michael Jerris > Sent: Fri, 12 May 2017 03:58:08 GMT+0530 > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Changing audio DTLS state from HANDSHAKE to FAIL issue > > we just had confirmed on Jira that 1.6.17 windows is working fine with latest chrome. If you are getting errors still with something that is other than sipml5, please confirm details of what exactly the issue is. > >> On May 8, 2017, at 8:08 AM, sachin wrote: >> >> Hello Michael, >> >> I tried using Sip.js instead of sipml5 and with Linux as the server. I have installed the FreeSWITCH Version 1.9.0+git~20170501T171230Z~e3ef041517~64bit (git e3ef041 2017-05-01 17:12:30Z 64bit) >> >> I am getting INCOMPATIBLE DESTINATION error. Also tried sipml5.. with bot the client I am getting the same error. >> ? >> Please let me know what I am missing. >> ? >> Thanks and Regards, >> ?SD >> >> >> From: Michael Jerris >> Sent: Fri, 05 May 2017 21:22:20 >> To: FreeSWITCH Users Help >> Subject: Re: [Freeswitch-users] Changing audio DTLS state from HANDSHAKE to FAIL issue >> >> can you try this using sip.js or something thats confirmed to work when using linux as a server please. This may just be an issue with sipml5. >> >>> On May 5, 2017, at 9:53 AM, Brian West wrote: >>> >>> Clearly that isn't the case, I suspect a define is missing to actually enable ECDSA at this point. >>> >>> On Fri, May 5, 2017 at 8:47 AM, Gregor Nanger wrote: >>> Brian, isn't this solved in 1.6.16? >>> >>> FS-10037 [core] Update OpenSSL to version 1.0.2k for Windows >>> >>> 2017-05-05 15:26 GMT+02:00 Brian West : >>> OpenSSL on the windows build needs to be updated. >>> >>> https://freeswitch.org/jira/browse/FS-9510 >>> >>> On Thu, May 4, 2017 at 2:31 PM, Gregor Nanger wrote: >>> This error is familiar to me, I think so, if I remembered correctly. >>> >>> I am using windows. I had this error when client was Chrome. I just updated openssl dlls. I think that openssl library is updated on windows build now with version 1.6.17. And this version is also compiled on FS FTP to just download and install it. >>> >>> 2017-05-04 19:15 GMT+02:00 Anthony Minessale : >>> FS 1.5 sounds like a bad plan. >>> Try latest FS 1.6 or master. >>> >>> >>> On Thu, May 4, 2017 at 12:04 PM, ?talo Rossi wrote: >>> Old openssl version maybe? >>> Em qui, 4 de mai de 2017 ?s 11:19, sachin escreveu: >>> Hello All, >>> >>> ?I am trying to setup the Webrtc call between 2 SIPML5 client. I am using FreeSWITCH Version 1.6.14 on Windows. I am using internal LAN. >>> >>> The clients are getting registered over wss. I have created self signed certificates. In var.xml I have set the codecs setting as follows >>> >>> >>> >>> >>> ?I am able to establish the call and there is 2 way voice when I call from Sipmpl5 to Linphone and it works. But when I call from Linphone to Firefox (Simpl5) then the call is not getting established and I am getting following error >>> >>> 2017-05-04 11:29:04.543434 [ERR] switch_rtp.c:2907 audio Handshake failure 1 >>> 2017-05-04 11:29:04.543434 [INFO] switch_rtp.c:2908 Changing audio DTLS state from HANDSHAKE to FAIL >>> 2017-05-04 11:29:04.543434 [NOTICE] switch_rtp.c:2889 Hangup sofia/internal/sip:1001 at df7jal 23ls0d.invalid [CS_EXCHANGE_MEDIA] [DESTINATION_OUT_OF_ORDER] >>> >>> My setup is as follows >>> >>> SIPML5 (FireFox browser 53.0)<-----> FS (v1.5)(Windows) <----> Linphone >>> >>> I am attaching the logs for the reference. >>> fs-logleve9.txt : Debug trace with loglevel =9 >>> ?fs-sip-trace.txt : Sip trace >>> Please let me know what could the issue and pointers to resolve the same. >>> >>> ?Thanks and Regards >>> SD > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170518/c03b00a5/attachment-0001.html From mike at jerris.com Thu May 18 20:20:24 2017 From: mike at jerris.com (Michael Jerris) Date: Thu, 18 May 2017 12:20:24 -0400 Subject: [Freeswitch-users] Uninstalling freeswitch after compiling and installing from source In-Reply-To: References: Message-ID: Depends how you configured, when compiling from source configs default to /usr/local/freeswitch/conf > On May 17, 2017, at 7:02 PM, Ravi sanyal wrote: > > Hi, > > I made a noob mistake and ran the script for installing freeswitch from binaries without doing logs on the process and fs_cli is complaining that commands are not found. > > specifically, sofia status, or sofia status profile but im sure that it extends further than that. I'm dont know how to get a list of the commands available. typing in help (not /help) at the prompt says it can't find a command. > > So i've installed it and it doesn't seem to be working, and don't know how to uninstall it. > > Where can i find a list of acceptable commands for the configure command? > > Im trying to enable sctp, but when looking through the logs for it you have a line: > > checking whether SCTP is supported... yes > > is this the same as checking with the --enable-sctp flag? (when looking at the logs for the added command there's no extra lines pertaining to enable-sctp as if you didn't specify it). > > I'm running the latest build from the github repository. > > When running this command: > > ./configure --enable-sctp > /home/fleetchat/config.log > > it keeps on displaying errors on output, but don't know if they're serious or not. > > ./configure: line 31222: php-config: command not found > ./configure: line 31223: php-config: command not found > ./configure: line 31224: php-config: command not found > ./configure: line 31225: php: command not found > ./configure: line 31226: php-config: command not found > ./configure: line 31227: php-config: command not found > ./configure: line 5306: AX_COMPILER_VENDOR: command not found > rm: cannot remove 'conftest*': No such file or directory > ./configure: line 19537: test: 3.16.0-4-amd64: integer expression expected > ./configure: line 19819: test: 3.16.0-4-amd64: integer expression expected > config.status: WARNING: 'apr-config.in ' seems to ignore the --datarootdir setting > rm: cannot remove 'libtoolT': No such file or directory > config.status: WARNING: 'Makefile.in' seems to ignore the --datarootdir setting > config.status: WARNING: 'lib/Makefile.in' seems to ignore the --datarootdir setting > configure: WARNING: ** STUN support disabled ** > > Im sure it's somewhere but i also don't know where the configuration files should be located when building from source. i placed them in /etc/freeswitch. From memory that was where the installation instructions said they should be located. > > The configuration files were downloaded from the ubuntu repository command freeswitch-meta-vanilla > > The reason they come from the ubuntu repository is because i had freeswitch set up on ubuntu 16.04 but found out debian was the recommended OS. I'm running freeswitch on debian 8.8 now. > > Nathan > > -- > Virtual Guard Ltd > info at virtualguard.co.nz _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170518/0177ba15/attachment.html From gmaruzz at gmail.com Thu May 18 20:21:44 2017 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Thu, 18 May 2017 18:21:44 +0200 Subject: [Freeswitch-users] Verto calls In-Reply-To: References: Message-ID: from FreeSWITCH command line: fsctl loglevel 7 console loglevel 7 then repeat the call, and post the *entire*, since beginning, not edited console output in https://pastebin.freeswitch.org/ (***not here in mail***), then write here again and tell us the pastebin address On 18 May 2017 at 09:15, Agust? Ubalde Bellot wrote: > Hi all, > > > > I am trying to call from Verto extension to another Verto extension. *Both > are successfully registered* (Verto status show the successfully > register) but the call between is not established. > > The call remains in ring state. > > > > This is the last dialplan function executed (calling 1000 to 1001): > > *EXECUTE verto.rtc/1001 bridge()* > > > > Regards, > > Tineli > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170518/130f4484/attachment.html From gmaruzz at gmail.com Thu May 18 20:27:49 2017 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Thu, 18 May 2017 18:27:49 +0200 Subject: [Freeswitch-users] mod_xml_curl performance questions In-Reply-To: <1494961415.1484944.978717552.3EE447BC@webmail.messagingengine.com> References: <1494961415.1484944.978717552.3EE447BC@webmail.messagingengine.com> Message-ID: On 16 May 2017 at 21:03, Michael Avers wrote: > > I'm planning on moving all our local lua-generated XML to use mod_xml_curl > over HTTPS. However, the primary http server is not in the same data center > as our FS instances. Ping time is about 50ms. Is this going to be an issue? > Is it typical to run xml_curl hosted elsewhere? The web app itself is > written in Go and is very fast. Just wondering about network latency aspect > of it all. > usually you run the webserver in same datacenter. That said, maybe in your case this is not a problem, and your application can bear the round trips. Take into account this do not affect call quality or call latency, it only affect how fast FreeSWITCH can get data (eg, how fast it takes to get auth info for a caller, how fast get the dialplan snippet to be executed, etc.) Once the call is established, rtp packets runs independently from database. > > 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 > -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170518/4258eb09/attachment-0001.html From david.villasmil.work at gmail.com Thu May 18 20:46:11 2017 From: david.villasmil.work at gmail.com (David Villasmil) Date: Thu, 18 May 2017 16:46:11 +0000 Subject: [Freeswitch-users] Uninstalling freeswitch after compiling and installing from source In-Reply-To: References: Message-ID: I'm assuming you're starting off with FS, If you're on debian and this is a fresh install, you're probably better off reinstalling debian and installing from apt following: https://freeswitch.org/confluence/plugins/servlet/mobile#content/view/9633817 On the "easy way" part. On Thu, May 18, 2017 at 6:21 PM Michael Jerris wrote: > Depends how you configured, when compiling from source configs default to > /usr/local/freeswitch/conf > > On May 17, 2017, at 7:02 PM, Ravi sanyal > wrote: > > Hi, > > I made a noob mistake and ran the script for installing freeswitch from > binaries without doing logs on the process and fs_cli is complaining that > commands are not found. > > specifically, sofia status, or sofia status profile but im sure > that it extends further than that. I'm dont know how to get a list of the > commands available. typing in help (not /help) at the prompt says it can't > find a command. > > So i've installed it and it doesn't seem to be working, and don't know how > to uninstall it. > > Where can i find a list of acceptable commands for the configure command? > > Im trying to enable sctp, but when looking through the logs for it you > have a line: > > checking whether SCTP is supported... yes > > is this the same as checking with the --enable-sctp flag? (when looking at > the logs for the added command there's no extra lines pertaining to > enable-sctp as if you didn't specify it). > > I'm running the latest build from the github repository. > > When running this command: > > ./configure --enable-sctp > /home/fleetchat/config.log > > it keeps on displaying errors on output, but don't know if they're serious > or not. > > ./configure: line 31222: php-config: command not found > ./configure: line 31223: php-config: command not found > ./configure: line 31224: php-config: command not found > ./configure: line 31225: php: command not found > ./configure: line 31226: php-config: command not found > ./configure: line 31227: php-config: command not found > ./configure: line 5306: AX_COMPILER_VENDOR: command not found > rm: cannot remove 'conftest*': No such file or directory > ./configure: line 19537: test: 3.16.0-4-amd64: integer expression expected > ./configure: line 19819: test: 3.16.0-4-amd64: integer expression expected > config.status: WARNING: 'apr-config.in' seems to ignore the > --datarootdir setting > rm: cannot remove 'libtoolT': No such file or directory > config.status: WARNING: 'Makefile.in' seems to ignore the --datarootdir > setting > config.status: WARNING: 'lib/Makefile.in' seems to ignore the > --datarootdir setting > configure: WARNING: ** STUN support disabled ** > > Im sure it's somewhere but i also don't know where the configuration files > should be located when building from source. i placed them in > /etc/freeswitch. From memory that was where the installation instructions > said they should be located. > > The configuration files were downloaded from the ubuntu repository command > freeswitch-meta-vanilla > > The reason they come from the ubuntu repository is because i had > freeswitch set up on ubuntu 16.04 but found out debian was the recommended > OS. I'm running freeswitch on debian 8.8 now. > > Nathan > > -- > > *Virtual Guard Ltd* > *info at virtualguard.co.nz * > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170518/56482385/attachment.html From gregor at infomedia.si Thu May 18 21:12:38 2017 From: gregor at infomedia.si (Gregor Nanger) Date: Thu, 18 May 2017 17:12:38 +0000 Subject: [Freeswitch-users] Callcenter modul error:: register subclass In-Reply-To: <13C9308F-F688-4A24-8175-76E761CAB9AB@jerris.com> References: <13C9308F-F688-4A24-8175-76E761CAB9AB@jerris.com> Message-ID: Well, fs restart solved issue. Will keep an eye if problem repeats. On Thu, May 18, 2017, 18:18 Michael Jerris wrote: > Thats strange, that means some other module already has that loaded? did > you do something strange like copy the module and try to load it again? > > On May 18, 2017, at 3:51 AM, Gregor Nanger wrote: > > ?Can anyone help with error that I get when try to load callcenter module: > > 2017-05-18 09:48:49.828297 [ERR] mod_callcenter.c:3810 Couldn't register > subclass callcenter::info! > 2017-05-18 09:48:49.828297 [CRIT] switch_loadable_module.c:1522 Error > Loading module /usr/lib/freeswitch/mod/mod_callcenter.so > **Module load routine returned an error**? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170518/35f22d76/attachment.html From ovoshlook at gmail.com Thu May 18 21:13:28 2017 From: ovoshlook at gmail.com (Yuriy Gorlichenko) Date: Thu, 18 May 2017 20:13:28 +0300 Subject: [Freeswitch-users] REFER for transer call handling Message-ID: Hi. I found some examples about features.xml but it is just a part of dialplan and will be executed on the DTMF sended in call processing (am i right?), so I trying to understand how Freeswitch handles REFER. Presume i have phone with button "transfer" that sends REFER message to server. How can I associate dialplan whith REFER message arrived to server (rewrite function for it or give additional functionality). Also can i bind some codes (like in conference mode) for every call? Because in coference mode it is very cool to use: session:execute("bind_digit_action","moderator,*1,exec:execute_extension,add") session:execute("bind_digit_action","moderator,*2,exec:execute_extension,remove") for rewrite feature codes for it. Thx you for response -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170518/d7c0121f/attachment.html From lwahlmeier at gmail.com Thu May 18 21:34:52 2017 From: lwahlmeier at gmail.com (Luke Wahlmeier) Date: Thu, 18 May 2017 11:34:52 -0600 Subject: [Freeswitch-users] WSS SSL errors "decryption failed or bad record mac" under load In-Reply-To: References: <358E7E05-A60A-463A-911F-A921F6CDEAA2@jerris.com> Message-ID: I have been able to verify this is definitely an issue with the sofia wss transport. If I put my own ssl proxy (stunnel) in front and use sofia ws transport the problem goes away. I have tried quite a bit of debugging and cant figure out when/how this is happening, my best guess is something around the ssl locking on connection close, but I have not had the time to dig into that yet. Thanks Luke On Fri, May 12, 2017 at 12:03 PM, Luke Wahlmeier wrote: > Here is the debug logs from this happening: > > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [NOTICE] > switch_channel.c:1104 New Channel sofia/websocket/nobody at 52LX8LP8BBWG6990 > [f9a97c98-373b-11e7-9136-499bb33ea37e] > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] > switch_core_state_machine.c:584 (sofia/websocket/nobody at 52LX8LP8BBWG6990) > Running State Change CS_NEW (Cur 1 Tot 1) > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] > sofia.c:10028 sofia/websocket/nobody at 52LX8LP8BBWG6990 receiving invite > from 192.168.56.151:53442 version: 1.9.0 git db24869 2017-05-11 18:22:45Z > 64bit > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] > sofia.c:11325 Setting NAT mode based on websockets > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] > sofia.c:7247 Channel sofia/websocket/nobody at 52LX8LP8BBWG6990 entering > state [received][100] > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] > sofia.c:7257 Remote SDP: > f9a97c98-373b-11e7-9136-499bb33ea37e v=0 > f9a97c98-373b-11e7-9136-499bb33ea37e o=- 196478633 2 IN IP4 192.168.56.151 > f9a97c98-373b-11e7-9136-499bb33ea37e s=- > f9a97c98-373b-11e7-9136-499bb33ea37e c=IN IP4 192.168.56.151 > f9a97c98-373b-11e7-9136-499bb33ea37e t=0 0 > f9a97c98-373b-11e7-9136-499bb33ea37e m=audio 42504 RTP/AVP 0 100 > f9a97c98-373b-11e7-9136-499bb33ea37e a=rtpmap:0 PCMU/8000 > f9a97c98-373b-11e7-9136-499bb33ea37e a=maxptime:20 > f9a97c98-373b-11e7-9136-499bb33ea37e > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] > switch_core_media.c:5110 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[ > PCMU:0:8000:20:64000:1] > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] > switch_core_media.c:5165 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ > is saved as a match > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] > switch_core_media.c:5110 Audio Codec Compare [:100:0:20:0:1]/[PCMU:0:8000: > 20:64000:1] > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] > switch_core_media.c:3425 Set Codec sofia/websocket/nobody at 52LX8LP8BBWG6990 > PCMU/8000 20 ms 160 samples 64000 bits 1 channels > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] > switch_core_codec.c:111 sofia/websocket/nobody at 52LX8LP8BBWG6990 Original > read codec set to PCMU:0 > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] > switch_core_media.c:5398 No 2833 in SDP. Liberal DTMF mode adding 101 as > telephone-event. > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] > switch_core_media.c:5427 sofia/websocket/nobody at 52LX8LP8BBWG6990 Set 2833 > dtmf send payload to 101 recv payload to 101 > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] > sofia.c:7670 (sofia/websocket/nobody at 52LX8LP8BBWG6990) State Change > CS_NEW -> CS_INIT > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] > switch_core_state_machine.c:603 (sofia/websocket/nobody at 52LX8LP8BBWG6990) > State NEW > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] > switch_core_state_machine.c:584 (sofia/websocket/nobody at 52LX8LP8BBWG6990) > Running State Change CS_INIT (Cur 1 Tot 1) > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] > switch_core_state_machine.c:627 (sofia/websocket/nobody at 52LX8LP8BBWG6990) > State INIT > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] > mod_sofia.c:93 sofia/websocket/nobody at 52LX8LP8BBWG6990 SOFIA INIT > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] > switch_core_state_machine.c:40 sofia/websocket/nobody at 52LX8LP8BBWG6990 > Standard INIT > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] > switch_core_state_machine.c:48 (sofia/websocket/nobody at 52LX8LP8BBWG6990) > State Change CS_INIT -> CS_ROUTING > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] > switch_core_state_machine.c:627 (sofia/websocket/nobody at 52LX8LP8BBWG6990) > State INIT going to sleep > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] > switch_core_state_machine.c:584 (sofia/websocket/nobody at 52LX8LP8BBWG6990) > Running State Change CS_ROUTING (Cur 1 Tot 1) > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] > switch_channel.c:2249 (sofia/websocket/nobody at 52LX8LP8BBWG6990) Callstate > Change DOWN -> RINGING > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] > switch_core_state_machine.c:643 (sofia/websocket/nobody at 52LX8LP8BBWG6990) > State ROUTING > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] > mod_sofia.c:154 sofia/websocket/nobody at 52LX8LP8BBWG6990 SOFIA ROUTING > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] > switch_core_state_machine.c:236 sofia/websocket/nobody at 52LX8LP8BBWG6990 > Standard ROUTING > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [INFO] > mod_dialplan_xml.c:637 Processing unknown <>->52LX8LP8BBWG6990 in context > wss-dialplan > f9a97c98-373b-11e7-9136-499bb33ea37e Dialplan: sofia/websocket/nobody at 52LX8LP8BBWG6990 > parsing [wss-dialplan->wss-dialplan] continue=false > f9a97c98-373b-11e7-9136-499bb33ea37e Dialplan: sofia/websocket/nobody at 52LX8LP8BBWG6990 > Regex (PASS) [wss-dialplan] destination_number(52LX8LP8BBWG6990) =~ /.+/ > break=on-false > f9a97c98-373b-11e7-9136-499bb33ea37e Dialplan: sofia/websocket/nobody at 52LX8LP8BBWG6990 > Action conference(test123) > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] > switch_core_state_machine.c:286 (sofia/websocket/nobody at 52LX8LP8BBWG6990) > State Change CS_ROUTING -> CS_EXECUTE > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] > switch_core_state_machine.c:643 (sofia/websocket/nobody at 52LX8LP8BBWG6990) > State ROUTING going to sleep > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] > switch_core_state_machine.c:584 (sofia/websocket/nobody at 52LX8LP8BBWG6990) > Running State Change CS_EXECUTE (Cur 1 Tot 1) > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] > switch_core_state_machine.c:650 (sofia/websocket/nobody at 52LX8LP8BBWG6990) > State EXECUTE > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] > mod_sofia.c:209 sofia/websocket/nobody at 52LX8LP8BBWG6990 SOFIA EXECUTE > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] > switch_core_state_machine.c:328 sofia/websocket/nobody at 52LX8LP8BBWG6990 > Standard EXECUTE > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] > switch_core_session.c:2707 Application conference Requires media! > pre_answering channel sofia/websocket/nobody at 52LX8LP8BBWG6990 > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [INFO] > switch_core_session.c:2709 Sending early media > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] > switch_core_media.c:8144 AUDIO RTP [sofia/websocket/nobody at 52LX8LP8BBWG6990] > 172.16.19.215 port 27662 -> 192.168.56.151 port 42504 codec: 0 ms: 20 > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] > switch_rtp.c:4096 Starting timer [soft] 160 bytes per 20ms > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] > switch_core_media.c:8447 sofia/websocket/nobody at 52LX8LP8BBWG6990 Set 2833 > dtmf send payload to 101 > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] > switch_core_media.c:8454 sofia/websocket/nobody at 52LX8LP8BBWG6990 Set 2833 > dtmf receive payload to 101 > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] > switch_core_media.c:8477 sofia/websocket/nobody at 52LX8LP8BBWG6990 Set rtp > dtmf delay to 40 > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] > mod_sofia.c:2512 Ring SDP: > f9a97c98-373b-11e7-9136-499bb33ea37e v=0 > f9a97c98-373b-11e7-9136-499bb33ea37e o=FreeSWITCH 1494583976 1494583977 > IN IP4 172.16.19.215 > f9a97c98-373b-11e7-9136-499bb33ea37e s=FreeSWITCH > f9a97c98-373b-11e7-9136-499bb33ea37e c=IN IP4 172.16.19.215 > f9a97c98-373b-11e7-9136-499bb33ea37e t=0 0 > f9a97c98-373b-11e7-9136-499bb33ea37e m=audio 27662 RTP/AVP 0 101 > f9a97c98-373b-11e7-9136-499bb33ea37e a=rtpmap:0 PCMU/8000 > f9a97c98-373b-11e7-9136-499bb33ea37e a=rtpmap:101 telephone-event/8000 > f9a97c98-373b-11e7-9136-499bb33ea37e a=fmtp:101 0-16 > f9a97c98-373b-11e7-9136-499bb33ea37e a=ptime:20 > f9a97c98-373b-11e7-9136-499bb33ea37e a=sendrecv > f9a97c98-373b-11e7-9136-499bb33ea37e > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [NOTICE] > mod_sofia.c:2515 Pre-Answer sofia/websocket/nobody at 52LX8LP8BBWG6990! > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] > switch_channel.c:3481 (sofia/websocket/nobody at 52LX8LP8BBWG6990) Callstate > Change RINGING -> EARLY > f9a97c98-373b-11e7-9136-499bb33ea37e EXECUTE sofia/websocket/nobody at 52LX8LP8BBWG6990 > conference(test123) > f9a97c98-373b-11e7-9136-499bb33ea37e v=0 > f9a97c98-373b-11e7-9136-499bb33ea37e o=FreeSWITCH 1494583976 1494583978 > IN IP4 172.16.19.215 > f9a97c98-373b-11e7-9136-499bb33ea37e s=FreeSWITCH > f9a97c98-373b-11e7-9136-499bb33ea37e c=IN IP4 172.16.19.215 > f9a97c98-373b-11e7-9136-499bb33ea37e t=0 0 > f9a97c98-373b-11e7-9136-499bb33ea37e m=audio 27662 RTP/AVP 0 101 > f9a97c98-373b-11e7-9136-499bb33ea37e a=rtpmap:0 PCMU/8000 > f9a97c98-373b-11e7-9136-499bb33ea37e a=rtpmap:101 telephone-event/8000 > f9a97c98-373b-11e7-9136-499bb33ea37e a=fmtp:101 0-16 > f9a97c98-373b-11e7-9136-499bb33ea37e a=ptime:20 > f9a97c98-373b-11e7-9136-499bb33ea37e a=sendrecv > f9a97c98-373b-11e7-9136-499bb33ea37e > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] > sofia.c:7247 Channel sofia/websocket/nobody at 52LX8LP8BBWG6990 entering > state [early][183] > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [NOTICE] > mod_conference.c:1829 Channel [sofia/websocket/nobody at 52LX8LP8BBWG6990] > has been answered > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] > switch_channel.c:3780 (sofia/websocket/nobody at 52LX8LP8BBWG6990) Callstate > Change EARLY -> ACTIVE > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] > sofia.c:7247 Channel sofia/websocket/nobody at 52LX8LP8BBWG6990 entering > state [completed][200] > 2017-05-12 17:53:58.932851 [DEBUG] mod_conference.c:3289 using channel > sound prefix: /tmp/fs1.8/share/freeswitch/sounds/en/us/callie > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] > conference_member.c:1715 Raw Codec Activation Success L16 at 8000hz 1 > channel 20ms > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] > conference_member.c:1762 Raw Codec Activation Success L16 at 8000hz 1 > channel 20ms > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] > switch_core_codec.c:223 sofia/websocket/nobody at 52LX8LP8BBWG6990 Push > codec L16:100 > 2017-05-12 17:53:58.932851 [DEBUG] mod_conference.c:227 Setup timer > success interval: 20 samples: 160 > 2017-05-12 17:53:58.932851 [ERR] switch_core_video.c:2868 This function is > not available, libpng not installed > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] > conference_loop.c:1306 Setup timer soft success interval: 20 samples: 160 > from codec PCMU > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [NOTICE] > switch_channel.c:1104 New Channel sofia/websocket/nobody at 3PXT9NIJPMV6KC8W > [02b30be2-373c-11e7-913a-499bb33ea37e] > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] > switch_core_state_machine.c:584 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) > Running State Change CS_NEW (Cur 2 Tot 2) > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] > sofia.c:10028 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W receiving invite > from 192.168.56.151:53568 version: 1.9.0 git db24869 2017-05-11 18:22:45Z > 64bit > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] > sofia.c:11325 Setting NAT mode based on websockets > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] > sofia.c:7247 Channel sofia/websocket/nobody at 3PXT9NIJPMV6KC8W entering > state [received][100] > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] > sofia.c:7257 Remote SDP: > 02b30be2-373c-11e7-913a-499bb33ea37e v=0 > 02b30be2-373c-11e7-913a-499bb33ea37e o=- 716582477 2 IN IP4 192.168.56.151 > 02b30be2-373c-11e7-913a-499bb33ea37e s=- > 02b30be2-373c-11e7-913a-499bb33ea37e c=IN IP4 192.168.56.151 > 02b30be2-373c-11e7-913a-499bb33ea37e t=0 0 > 02b30be2-373c-11e7-913a-499bb33ea37e m=audio 11866 RTP/AVP 0 100 > 02b30be2-373c-11e7-913a-499bb33ea37e a=rtpmap:0 PCMU/8000 > 02b30be2-373c-11e7-913a-499bb33ea37e a=maxptime:20 > 02b30be2-373c-11e7-913a-499bb33ea37e > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] > switch_core_media.c:5110 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[ > PCMU:0:8000:20:64000:1] > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] > switch_core_media.c:5165 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ > is saved as a match > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] > switch_core_media.c:5110 Audio Codec Compare [:100:0:20:0:1]/[PCMU:0:8000: > 20:64000:1] > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] > switch_core_media.c:3425 Set Codec sofia/websocket/nobody at 3PXT9NIJPMV6KC8W > PCMU/8000 20 ms 160 samples 64000 bits 1 channels > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] > switch_core_codec.c:111 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W Original > read codec set to PCMU:0 > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] > switch_core_media.c:5398 No 2833 in SDP. Liberal DTMF mode adding 101 as > telephone-event. > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] > switch_core_media.c:5427 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W Set 2833 > dtmf send payload to 101 recv payload to 101 > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] > sofia.c:7670 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) State Change > CS_NEW -> CS_INIT > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] > switch_core_state_machine.c:603 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) > State NEW > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] > switch_core_state_machine.c:584 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) > Running State Change CS_INIT (Cur 2 Tot 2) > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] > switch_core_state_machine.c:627 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) > State INIT > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] > mod_sofia.c:93 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W SOFIA INIT > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] > switch_core_state_machine.c:40 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W > Standard INIT > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] > switch_core_state_machine.c:48 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) > State Change CS_INIT -> CS_ROUTING > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] > switch_core_state_machine.c:627 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) > State INIT going to sleep > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] > switch_core_state_machine.c:584 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) > Running State Change CS_ROUTING (Cur 2 Tot 2) > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] > switch_channel.c:2249 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) Callstate > Change DOWN -> RINGING > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] > switch_core_state_machine.c:643 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) > State ROUTING > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] > mod_sofia.c:154 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W SOFIA ROUTING > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] > switch_core_state_machine.c:236 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W > Standard ROUTING > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [INFO] > mod_dialplan_xml.c:637 Processing unknown <>->3PXT9NIJPMV6KC8W in context > wss-dialplan > 02b30be2-373c-11e7-913a-499bb33ea37e Dialplan: sofia/websocket/nobody at 3PXT9NIJPMV6KC8W > parsing [wss-dialplan->wss-dialplan] continue=false > 02b30be2-373c-11e7-913a-499bb33ea37e Dialplan: sofia/websocket/nobody at 3PXT9NIJPMV6KC8W > Regex (PASS) [wss-dialplan] destination_number(3PXT9NIJPMV6KC8W) =~ /.+/ > break=on-false > 02b30be2-373c-11e7-913a-499bb33ea37e Dialplan: sofia/websocket/nobody at 3PXT9NIJPMV6KC8W > Action conference(test123) > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] > switch_core_state_machine.c:286 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) > State Change CS_ROUTING -> CS_EXECUTE > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] > switch_core_state_machine.c:643 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) > State ROUTING going to sleep > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] > switch_core_state_machine.c:584 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) > Running State Change CS_EXECUTE (Cur 2 Tot 2) > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] > switch_core_state_machine.c:650 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) > State EXECUTE > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] > mod_sofia.c:209 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W SOFIA EXECUTE > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] > switch_core_state_machine.c:328 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W > Standard EXECUTE > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] > switch_core_session.c:2707 Application conference Requires media! > pre_answering channel sofia/websocket/nobody at 3PXT9NIJPMV6KC8W > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [INFO] > switch_core_session.c:2709 Sending early media > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] > switch_core_media.c:8144 AUDIO RTP [sofia/websocket/nobody at 3PXT9NIJPMV6KC8W] > 172.16.19.215 port 28858 -> 192.168.56.151 port 11866 codec: 0 ms: 20 > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] > switch_rtp.c:4096 Starting timer [soft] 160 bytes per 20ms > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] > switch_core_media.c:8447 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W Set 2833 > dtmf send payload to 101 > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] > switch_core_media.c:8454 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W Set 2833 > dtmf receive payload to 101 > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] > switch_core_media.c:8477 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W Set rtp > dtmf delay to 40 > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] > mod_sofia.c:2512 Ring SDP: > 02b30be2-373c-11e7-913a-499bb33ea37e v=0 > 02b30be2-373c-11e7-913a-499bb33ea37e o=FreeSWITCH 1494582796 1494582797 > IN IP4 172.16.19.215 > 02b30be2-373c-11e7-913a-499bb33ea37e s=FreeSWITCH > 02b30be2-373c-11e7-913a-499bb33ea37e c=IN IP4 172.16.19.215 > 02b30be2-373c-11e7-913a-499bb33ea37e t=0 0 > 02b30be2-373c-11e7-913a-499bb33ea37e m=audio 28858 RTP/AVP 0 101 > 02b30be2-373c-11e7-913a-499bb33ea37e a=rtpmap:0 PCMU/8000 > 02b30be2-373c-11e7-913a-499bb33ea37e a=rtpmap:101 telephone-event/8000 > 02b30be2-373c-11e7-913a-499bb33ea37e a=fmtp:101 0-16 > 02b30be2-373c-11e7-913a-499bb33ea37e a=ptime:20 > 02b30be2-373c-11e7-913a-499bb33ea37e a=sendrecv > 02b30be2-373c-11e7-913a-499bb33ea37e > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [NOTICE] > mod_sofia.c:2515 Pre-Answer sofia/websocket/nobody at 3PXT9NIJPMV6KC8W! > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] > switch_channel.c:3481 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) Callstate > Change RINGING -> EARLY > 02b30be2-373c-11e7-913a-499bb33ea37e EXECUTE sofia/websocket/nobody at 3PXT9NIJPMV6KC8W > conference(test123) > 02b30be2-373c-11e7-913a-499bb33ea37e v=0 > 02b30be2-373c-11e7-913a-499bb33ea37e o=FreeSWITCH 1494582796 1494582798 > IN IP4 172.16.19.215 > 02b30be2-373c-11e7-913a-499bb33ea37e s=FreeSWITCH > 02b30be2-373c-11e7-913a-499bb33ea37e c=IN IP4 172.16.19.215 > 02b30be2-373c-11e7-913a-499bb33ea37e t=0 0 > 02b30be2-373c-11e7-913a-499bb33ea37e m=audio 28858 RTP/AVP 0 101 > 02b30be2-373c-11e7-913a-499bb33ea37e a=rtpmap:0 PCMU/8000 > 02b30be2-373c-11e7-913a-499bb33ea37e a=rtpmap:101 telephone-event/8000 > 02b30be2-373c-11e7-913a-499bb33ea37e a=fmtp:101 0-16 > 02b30be2-373c-11e7-913a-499bb33ea37e a=ptime:20 > 02b30be2-373c-11e7-913a-499bb33ea37e a=sendrecv > 02b30be2-373c-11e7-913a-499bb33ea37e > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] > sofia.c:7247 Channel sofia/websocket/nobody at 3PXT9NIJPMV6KC8W entering > state [early][183] > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [NOTICE] > mod_conference.c:1829 Channel [sofia/websocket/nobody at 3PXT9NIJPMV6KC8W] > has been answered > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] > switch_channel.c:3780 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) Callstate > Change EARLY -> ACTIVE > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] > sofia.c:7247 Channel sofia/websocket/nobody at 3PXT9NIJPMV6KC8W entering > state [completed][200] > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] > conference_member.c:1715 Raw Codec Activation Success L16 at 8000hz 1 > channel 20ms > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] > conference_member.c:1762 Raw Codec Activation Success L16 at 8000hz 1 > channel 20ms > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] > switch_core_codec.c:223 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W Push > codec L16:100 > 2017-05-12 17:54:14.092839 [ERR] switch_core_video.c:2868 This function is > not available, libpng not installed > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] > conference_loop.c:1306 Setup timer soft success interval: 20 samples: 160 > from codec PCMU > > This is where session 02b30be2-373c-11e7-913a-499bb33ea37e wss socket > gets an ssl errror: > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.952843 [DEBUG] > sofia.c:7247 Channel sofia/websocket/nobody at 52LX8LP8BBWG6990 entering > state [terminating][0] > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.952843 [NOTICE] > sofia.c:8438 Hangup sofia/websocket/nobody at 52LX8LP8BBWG6990 [CS_EXECUTE] > [NORMAL_UNSPECIFIED] > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [INFO] > conference_loop.c:1621 Channel leaving conference, cause: NORMAL_UNSPECIFIED > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] > mod_conference.c:2404 sofia/websocket/nobody at 52LX8LP8BBWG6990 skip > receive message [TRANSFER] (channel is hungup already) > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] > switch_core_media.c:11838 sofia/websocket/nobody at 52LX8LP8BBWG6990 skip > receive message [BITRATE_REQ] (channel is hungup already) > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] > switch_core_codec.c:248 sofia/websocket/nobody at 52LX8LP8BBWG6990 Restore > previous codec PCMU:0. > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] > switch_core_session.c:2884 sofia/websocket/nobody at 52LX8LP8BBWG6990 skip > receive message [PHONE_EVENT] (channel is hungup already) > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] > switch_core_state_machine.c:650 (sofia/websocket/nobody at 52LX8LP8BBWG6990) > State EXECUTE going to sleep > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] > switch_core_state_machine.c:584 (sofia/websocket/nobody at 52LX8LP8BBWG6990) > Running State Change CS_HANGUP (Cur 2 Tot 2) > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] > switch_core_state_machine.c:850 (sofia/websocket/nobody at 52LX8LP8BBWG6990) > Callstate Change ACTIVE -> HANGUP > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] > switch_core_state_machine.c:852 (sofia/websocket/nobody at 52LX8LP8BBWG6990) > State HANGUP > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] > mod_sofia.c:449 Channel sofia/websocket/nobody at 52LX8LP8BBWG6990 hanging > up, cause: NORMAL_UNSPECIFIED > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] > switch_core_state_machine.c:60 sofia/websocket/nobody at 52LX8LP8BBWG6990 > Standard HANGUP, cause: NORMAL_UNSPECIFIED > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] > switch_core_state_machine.c:852 (sofia/websocket/nobody at 52LX8LP8BBWG6990) > State HANGUP going to sleep > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] > switch_core_state_machine.c:619 (sofia/websocket/nobody at 52LX8LP8BBWG6990) > State Change CS_HANGUP -> CS_REPORTING > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] > switch_core_state_machine.c:584 (sofia/websocket/nobody at 52LX8LP8BBWG6990) > Running State Change CS_REPORTING (Cur 2 Tot 2) > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] > switch_core_state_machine.c:938 (sofia/websocket/nobody at 52LX8LP8BBWG6990) > State REPORTING > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] > switch_core_state_machine.c:174 sofia/websocket/nobody at 52LX8LP8BBWG6990 > Standard REPORTING, cause: NORMAL_UNSPECIFIED > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] > switch_core_state_machine.c:938 (sofia/websocket/nobody at 52LX8LP8BBWG6990) > State REPORTING going to sleep > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] > switch_core_state_machine.c:610 (sofia/websocket/nobody at 52LX8LP8BBWG6990) > State Change CS_REPORTING -> CS_DESTROY > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] > switch_core_session.c:1712 Session 1 (sofia/websocket/nobody at 52LX8LP8BBWG6990) > Locked, Waiting on external entities > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [NOTICE] > switch_core_session.c:1730 Session 1 (sofia/websocket/nobody at 52LX8LP8BBWG6990) > Ended > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [NOTICE] > switch_core_session.c:1734 Close Channel sofia/websocket/nobody at 52LX8LP8BBWG6990 > [CS_DESTROY] > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] > switch_core_state_machine.c:741 (sofia/websocket/nobody at 52LX8LP8BBWG6990) > Running State Change CS_DESTROY (Cur 1 Tot 2) > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] > switch_core_state_machine.c:751 (sofia/websocket/nobody at 52LX8LP8BBWG6990) > State DESTROY > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] > mod_sofia.c:354 sofia/websocket/nobody at 52LX8LP8BBWG6990 SOFIA DESTROY > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] > switch_core_state_machine.c:181 sofia/websocket/nobody at 52LX8LP8BBWG6990 > Standard DESTROY > f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] > switch_core_state_machine.c:751 (sofia/websocket/nobody at 52LX8LP8BBWG6990) > State DESTROY going to sleep > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.112842 [DEBUG] > sofia.c:7247 Channel sofia/websocket/nobody at 3PXT9NIJPMV6KC8W entering > state [terminating][0] > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.112842 [NOTICE] > sofia.c:8438 Hangup sofia/websocket/nobody at 3PXT9NIJPMV6KC8W [CS_EXECUTE] > [NORMAL_UNSPECIFIED] > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [INFO] > conference_loop.c:1621 Channel leaving conference, cause: NORMAL_UNSPECIFIED > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] > mod_conference.c:2404 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W skip > receive message [TRANSFER] (channel is hungup already) > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] > switch_core_media.c:11838 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W skip > receive message [BITRATE_REQ] (channel is hungup already) > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] > switch_core_codec.c:248 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W Restore > previous codec PCMU:0. > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] > switch_core_session.c:2884 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W skip > receive message [PHONE_EVENT] (channel is hungup already) > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] > switch_core_state_machine.c:650 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) > State EXECUTE going to sleep > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] > switch_core_state_machine.c:584 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) > Running State Change CS_HANGUP (Cur 1 Tot 2) > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] > switch_core_state_machine.c:850 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) > Callstate Change ACTIVE -> HANGUP > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] > switch_core_state_machine.c:852 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) > State HANGUP > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] > mod_sofia.c:449 Channel sofia/websocket/nobody at 3PXT9NIJPMV6KC8W hanging > up, cause: NORMAL_UNSPECIFIED > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] > switch_core_state_machine.c:60 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W > Standard HANGUP, cause: NORMAL_UNSPECIFIED > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] > switch_core_state_machine.c:852 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) > State HANGUP going to sleep > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] > switch_core_state_machine.c:619 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) > State Change CS_HANGUP -> CS_REPORTING > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] > switch_core_state_machine.c:584 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) > Running State Change CS_REPORTING (Cur 1 Tot 2) > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] > switch_core_state_machine.c:938 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) > State REPORTING > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] > switch_core_state_machine.c:174 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W > Standard REPORTING, cause: NORMAL_UNSPECIFIED > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] > switch_core_state_machine.c:938 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) > State REPORTING going to sleep > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] > switch_core_state_machine.c:610 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) > State Change CS_REPORTING -> CS_DESTROY > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] > switch_core_session.c:1712 Session 2 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) > Locked, Waiting on external entities > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [NOTICE] > switch_core_session.c:1730 Session 2 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) > Ended > 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [NOTICE] > switch_core_session.c:1734 Close Channel sofia/websocket/nobody at 3PXT9NIJPMV6KC8W > [CS_DESTROY] > > On Fri, May 12, 2017 at 11:51 AM, Luke Wahlmeier > wrote: > >> Thanks Michael, >> >> I am more then happy to setup something with libks if needed. >> >> I have figured out some more however. It appears that this only happens >> when a wss connections session has not fully established and is cleaning up >> because of timing out. The problem is that it causes another wss >> connection it to get this ssl error, even if that other wss connection has >> a fully established and running audio session. It is important to note it >> does not seem to interrupt audio just the wss sip channel, which I am >> fairly sure can be reestablished for that audio session w/o an issue. >> >> The sessions that is being cleaned up sends the logs messages as its >> doing it: >> 2017-05-12 17:32:46.607768 [NOTICE] sofia.c:8438 Hangup >> sofia/websocket/nobody at 1LF3F6I924P9WH6U [CS_EXECUTE] [NORMAL_UNSPECIFIED] >> 2017-05-12 17:32:46.627768 [INFO] conference_loop.c:1621 Channel leaving >> conference, cause: NORMAL_UNSPECIFIED >> 2017-05-12 17:32:46.627768 [NOTICE] switch_core_session.c:1730 Session 46 >> (sofia/websocket/nobody at 1LF3F6I924P9WH6U) Ended >> 2017-05-12 17:32:46.627768 [NOTICE] switch_core_session.c:1734 Close >> Channel sofia/websocket/nobody at 1LF3F6I924P9WH6U [CS_DESTROY] >> >> I have attached the updated python script, it can duplicate this every >> time now with only 2 connections. I verified with a webRTC client that if >> I initiate this first connection in the script, let it close, then connect >> the webRTC client and get full audio, once the first session from the >> script times out it causes the webRTC wss connection to get an error and >> close. >> >> The webRTC connection is in chrome with sip.js. >> >> Sorry the python script is so nasty, was working through any possible >> duplicated sip session stuff in it to make sure that was not why it was >> hitting the second connection. >> >> >> On Fri, May 12, 2017 at 10:20 AM, Michael Jerris wrote: >> >>> test on master.. work a similar test for verto maybe, this might have to >>> do with sip specifically trying to keep state. Might make sense to build >>> something out of libks as it has basically the same web socket code, and >>> has both client and server web socket support in it, to do a ?real? test?, >>> instead of this fake sip without any state over web sockets. >>> >>> >>> On May 12, 2017, at 11:42 AM, Luke Wahlmeier >>> wrote: >>> >>> Just got done testing this on v1.6 head and master, both seem to still >>> have this issue. This box is using libssl version 1.0.1t-1+deb8u6. I am >>> gonna start digging more into the ws/wss/sofia code to see if I can figure >>> it out. Any suggestions on debugging this would be appreciated. >>> >>> Thanks >>> Luke >>> >>> On Thu, May 11, 2017 at 5:12 PM, Luke Wahlmeier >>> wrote: >>> >>>> Its just in our isolated lab, pretty normal dell xeon server running >>>> Jessie 8.6. I just want to get it building on the same box I am testing >>>> with so setting that all up. >>>> >>>> I was able to reproduce it w/o DTLS/Srtp. here is a much simpler and >>>> cleaned up version of the python script. >>>> >>>> >>>> >>>> On Thu, May 11, 2017 at 4:34 PM, Michael Jerris >>>> wrote: >>>> >>>>> what is ?this environment? ? >>>>> >>>>> On May 11, 2017, at 6:31 PM, Luke Wahlmeier >>>>> wrote: >>>>> >>>>> Yeah I can usually get it to happen within about 5 minutes or so of >>>>> testing. Still getting all setup to build freeswitch in this environment, >>>>> but I should have it working by tomorrow. I will try more w/o dtls/srtp as >>>>> well and make sure it does not need to be on. >>>>> >>>>> Thanks >>>>> Luke >>>>> >>>>> On Thu, May 11, 2017 at 4:20 PM, Michael Jerris >>>>> wrote: >>>>> >>>>>> if you can reproduce this reliably, i?d try master as well. Unless >>>>>> this is a bug in openssl, i can?t imagine how dtls would come into play in >>>>>> something like this. >>>>>> >>>>>> > On May 11, 2017, at 5:48 PM, Luke Wahlmeier >>>>>> wrote: >>>>>> > >>>>>> > I keep semi-regularly running into issues using the wss transport >>>>>> when using dtls/strp/ice. This is on the latest 1.6.17~34~0fc0946 on >>>>>> Debian jessie, but I am pretty sure it was happening on the last couple >>>>>> releases as well. >>>>>> > >>>>>> > It seems like something bad/wrong happens to the encrypted data >>>>>> going over the websocket coming from freeswitch when more then 1 websocket >>>>>> connection are going and so far ice/srtp/dtls also seem to be needed in the >>>>>> invite to duplicate it. >>>>>> > >>>>>> > I have tried many different languages and network/ssl stacks and >>>>>> keep running into this. It is always on data coming in from freeswitch on >>>>>> the websocket connection, and its very very random. Sometimes I will get >>>>>> it 20 times in a row, other times it takes thousands of >>>>>> connections/sessions before it happen. It also, obviously, completely goes >>>>>> away if I use plain ws instead wss. >>>>>> > >>>>>> > Here are the errors: >>>>>> > python: >>>>>> > SSLError: [SSL: DECRYPTION_FAILED_OR_BAD_RECORD_MAC] decryption >>>>>> failed or bad record mac (_ssl.c:1750) >>>>>> > c/c++ (stunnel4): >>>>>> > SSL_read: 1408F119: error:1408F119:SSL >>>>>> routines:SSL3_GET_RECORD:decryption failed or bad record mac >>>>>> > Java: >>>>>> > java.lang.IllegalArgumentException: Bad arguments >>>>>> > at javax.crypto.Mac.update(Mac.java:509) >>>>>> > at sun.security.ssl.MAC.compute(MAC.java:135) >>>>>> > at sun.security.ssl.InputRecord.checkMacTags(InputRecord.java:2 >>>>>> 65) >>>>>> > at sun.security.ssl.InputRecord.decrypt(InputRecord.java:216) >>>>>> > at sun.security.ssl.EngineInputRecord.decrypt(EngineInputRecord >>>>>> .java:177) >>>>>> > at sun.security.ssl.SSLEngineImpl.readRecord(SSLEngineImpl.java >>>>>> :974) >>>>>> > at sun.security.ssl.SSLEngineImpl.readNetRecord(SSLEngineImpl.j >>>>>> ava:907) >>>>>> > at sun.security.ssl.SSLEngineImpl.unwrap(SSLEngineImpl.java:781 >>>>>> ) >>>>>> > at javax.net.ssl.SSLEngine.unwrap(SSLEngine.java:624) >>>>>> > >>>>>> > Attached are a simple python script to do the load, my dialplan and >>>>>> sip_profile. The python script can take a few runs before it see the >>>>>> error, and I know its not completing the sip or rtp, but even if it does >>>>>> this still happens. >>>>>> > >>>>>> > I have also looked at libsofia-sip-ua/tport/ws.c and I dont see >>>>>> anything obvious. I am getting setup to build v1.6 head and test this any >>>>>> guidance on ways I can trouble shoot this better or requests for more info >>>>>> are very welcome. >>>>>> > >>>>>> >>>>> >>>>> >>>>> ____________________________________________________________ >>>>> _____________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-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 >>> >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170518/346b254a/attachment-0001.html From anthony.minessale at gmail.com Thu May 18 22:08:25 2017 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 18 May 2017 13:08:25 -0500 Subject: [Freeswitch-users] WSS SSL errors "decryption failed or bad record mac" under load In-Reply-To: References: <358E7E05-A60A-463A-911F-A921F6CDEAA2@jerris.com> Message-ID: Please do not post logs into email bodies, people use mobile devices to browse this list. Please do no report issues on this list, its a discussion list. We have a Bug tracker specifically for issues. I forgot completely about this thread because its impossible to mentally remember every one of the email threads in a busy mailing list. https://jira.freeswitch.org This has all you need to attach your data and file a proper report. Otherwise its pure Chaos..... On Thu, May 18, 2017 at 12:34 PM, Luke Wahlmeier wrote: > I have been able to verify this is definitely an issue with the sofia wss > transport. If I put my own ssl proxy (stunnel) in front and use sofia ws > transport the problem goes away. > > I have tried quite a bit of debugging and cant figure out when/how this is > happening, my best guess is something around the ssl locking on connection > close, but I have not had the time to dig into that yet. > > Thanks > Luke > > On Fri, May 12, 2017 at 12:03 PM, Luke Wahlmeier > wrote: > >> Here is the debug logs from this happening: >> >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [NOTICE] >> switch_channel.c:1104 New Channel sofia/websocket/nobody at 52LX8LP8BBWG6990 >> [f9a97c98-373b-11e7-9136-499bb33ea37e] >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] >> switch_core_state_machine.c:584 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >> Running State Change CS_NEW (Cur 1 Tot 1) >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] >> sofia.c:10028 sofia/websocket/nobody at 52LX8LP8BBWG6990 receiving invite >> from 192.168.56.151:53442 version: 1.9.0 git db24869 2017-05-11 >> 18:22:45Z 64bit >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] >> sofia.c:11325 Setting NAT mode based on websockets >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] >> sofia.c:7247 Channel sofia/websocket/nobody at 52LX8LP8BBWG6990 entering >> state [received][100] >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] >> sofia.c:7257 Remote SDP: >> f9a97c98-373b-11e7-9136-499bb33ea37e v=0 >> f9a97c98-373b-11e7-9136-499bb33ea37e o=- 196478633 2 IN IP4 >> 192.168.56.151 >> f9a97c98-373b-11e7-9136-499bb33ea37e s=- >> f9a97c98-373b-11e7-9136-499bb33ea37e c=IN IP4 192.168.56.151 >> f9a97c98-373b-11e7-9136-499bb33ea37e t=0 0 >> f9a97c98-373b-11e7-9136-499bb33ea37e m=audio 42504 RTP/AVP 0 100 >> f9a97c98-373b-11e7-9136-499bb33ea37e a=rtpmap:0 PCMU/8000 >> f9a97c98-373b-11e7-9136-499bb33ea37e a=maxptime:20 >> f9a97c98-373b-11e7-9136-499bb33ea37e >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] >> switch_core_media.c:5110 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU >> :0:8000:20:64000:1] >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] >> switch_core_media.c:5165 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ >> is saved as a match >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] >> switch_core_media.c:5110 Audio Codec Compare [:100:0:20:0:1]/[PCMU:0:8000:2 >> 0:64000:1] >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] >> switch_core_media.c:3425 Set Codec sofia/websocket/nobody at 52LX8LP8BBWG6990 >> PCMU/8000 20 ms 160 samples 64000 bits 1 channels >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] >> switch_core_codec.c:111 sofia/websocket/nobody at 52LX8LP8BBWG6990 Original >> read codec set to PCMU:0 >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] >> switch_core_media.c:5398 No 2833 in SDP. Liberal DTMF mode adding 101 as >> telephone-event. >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] >> switch_core_media.c:5427 sofia/websocket/nobody at 52LX8LP8BBWG6990 Set >> 2833 dtmf send payload to 101 recv payload to 101 >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] >> sofia.c:7670 (sofia/websocket/nobody at 52LX8LP8BBWG6990) State Change >> CS_NEW -> CS_INIT >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] >> switch_core_state_machine.c:603 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >> State NEW >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] >> switch_core_state_machine.c:584 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >> Running State Change CS_INIT (Cur 1 Tot 1) >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] >> switch_core_state_machine.c:627 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >> State INIT >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] >> mod_sofia.c:93 sofia/websocket/nobody at 52LX8LP8BBWG6990 SOFIA INIT >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] >> switch_core_state_machine.c:40 sofia/websocket/nobody at 52LX8LP8BBWG6990 >> Standard INIT >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] >> switch_core_state_machine.c:48 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >> State Change CS_INIT -> CS_ROUTING >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] >> switch_core_state_machine.c:627 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >> State INIT going to sleep >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] >> switch_core_state_machine.c:584 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >> Running State Change CS_ROUTING (Cur 1 Tot 1) >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] >> switch_channel.c:2249 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >> Callstate Change DOWN -> RINGING >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] >> switch_core_state_machine.c:643 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >> State ROUTING >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] >> mod_sofia.c:154 sofia/websocket/nobody at 52LX8LP8BBWG6990 SOFIA ROUTING >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] >> switch_core_state_machine.c:236 sofia/websocket/nobody at 52LX8LP8BBWG6990 >> Standard ROUTING >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [INFO] >> mod_dialplan_xml.c:637 Processing unknown <>->52LX8LP8BBWG6990 in context >> wss-dialplan >> f9a97c98-373b-11e7-9136-499bb33ea37e Dialplan: >> sofia/websocket/nobody at 52LX8LP8BBWG6990 parsing >> [wss-dialplan->wss-dialplan] continue=false >> f9a97c98-373b-11e7-9136-499bb33ea37e Dialplan: >> sofia/websocket/nobody at 52LX8LP8BBWG6990 Regex (PASS) [wss-dialplan] >> destination_number(52LX8LP8BBWG6990) =~ /.+/ break=on-false >> f9a97c98-373b-11e7-9136-499bb33ea37e Dialplan: >> sofia/websocket/nobody at 52LX8LP8BBWG6990 Action conference(test123) >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] >> switch_core_state_machine.c:286 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >> State Change CS_ROUTING -> CS_EXECUTE >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] >> switch_core_state_machine.c:643 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >> State ROUTING going to sleep >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] >> switch_core_state_machine.c:584 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >> Running State Change CS_EXECUTE (Cur 1 Tot 1) >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] >> switch_core_state_machine.c:650 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >> State EXECUTE >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] >> mod_sofia.c:209 sofia/websocket/nobody at 52LX8LP8BBWG6990 SOFIA EXECUTE >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] >> switch_core_state_machine.c:328 sofia/websocket/nobody at 52LX8LP8BBWG6990 >> Standard EXECUTE >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] >> switch_core_session.c:2707 Application conference Requires media! >> pre_answering channel sofia/websocket/nobody at 52LX8LP8BBWG6990 >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [INFO] >> switch_core_session.c:2709 Sending early media >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] >> switch_core_media.c:8144 AUDIO RTP [sofia/websocket/nobody at 52LX8LP8BBWG6990] >> 172.16.19.215 port 27662 -> 192.168.56.151 port 42504 codec: 0 ms: 20 >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] >> switch_rtp.c:4096 Starting timer [soft] 160 bytes per 20ms >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] >> switch_core_media.c:8447 sofia/websocket/nobody at 52LX8LP8BBWG6990 Set >> 2833 dtmf send payload to 101 >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] >> switch_core_media.c:8454 sofia/websocket/nobody at 52LX8LP8BBWG6990 Set >> 2833 dtmf receive payload to 101 >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] >> switch_core_media.c:8477 sofia/websocket/nobody at 52LX8LP8BBWG6990 Set rtp >> dtmf delay to 40 >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] >> mod_sofia.c:2512 Ring SDP: >> f9a97c98-373b-11e7-9136-499bb33ea37e v=0 >> f9a97c98-373b-11e7-9136-499bb33ea37e o=FreeSWITCH 1494583976 1494583977 >> IN IP4 172.16.19.215 >> f9a97c98-373b-11e7-9136-499bb33ea37e s=FreeSWITCH >> f9a97c98-373b-11e7-9136-499bb33ea37e c=IN IP4 172.16.19.215 >> f9a97c98-373b-11e7-9136-499bb33ea37e t=0 0 >> f9a97c98-373b-11e7-9136-499bb33ea37e m=audio 27662 RTP/AVP 0 101 >> f9a97c98-373b-11e7-9136-499bb33ea37e a=rtpmap:0 PCMU/8000 >> f9a97c98-373b-11e7-9136-499bb33ea37e a=rtpmap:101 telephone-event/8000 >> f9a97c98-373b-11e7-9136-499bb33ea37e a=fmtp:101 0-16 >> f9a97c98-373b-11e7-9136-499bb33ea37e a=ptime:20 >> f9a97c98-373b-11e7-9136-499bb33ea37e a=sendrecv >> f9a97c98-373b-11e7-9136-499bb33ea37e >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [NOTICE] >> mod_sofia.c:2515 Pre-Answer sofia/websocket/nobody at 52LX8LP8BBWG6990! >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] >> switch_channel.c:3481 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >> Callstate Change RINGING -> EARLY >> f9a97c98-373b-11e7-9136-499bb33ea37e EXECUTE >> sofia/websocket/nobody at 52LX8LP8BBWG6990 conference(test123) >> f9a97c98-373b-11e7-9136-499bb33ea37e v=0 >> f9a97c98-373b-11e7-9136-499bb33ea37e o=FreeSWITCH 1494583976 1494583978 >> IN IP4 172.16.19.215 >> f9a97c98-373b-11e7-9136-499bb33ea37e s=FreeSWITCH >> f9a97c98-373b-11e7-9136-499bb33ea37e c=IN IP4 172.16.19.215 >> f9a97c98-373b-11e7-9136-499bb33ea37e t=0 0 >> f9a97c98-373b-11e7-9136-499bb33ea37e m=audio 27662 RTP/AVP 0 101 >> f9a97c98-373b-11e7-9136-499bb33ea37e a=rtpmap:0 PCMU/8000 >> f9a97c98-373b-11e7-9136-499bb33ea37e a=rtpmap:101 telephone-event/8000 >> f9a97c98-373b-11e7-9136-499bb33ea37e a=fmtp:101 0-16 >> f9a97c98-373b-11e7-9136-499bb33ea37e a=ptime:20 >> f9a97c98-373b-11e7-9136-499bb33ea37e a=sendrecv >> f9a97c98-373b-11e7-9136-499bb33ea37e >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] >> sofia.c:7247 Channel sofia/websocket/nobody at 52LX8LP8BBWG6990 entering >> state [early][183] >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [NOTICE] >> mod_conference.c:1829 Channel [sofia/websocket/nobody at 52LX8LP8BBWG6990] >> has been answered >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] >> switch_channel.c:3780 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >> Callstate Change EARLY -> ACTIVE >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] >> sofia.c:7247 Channel sofia/websocket/nobody at 52LX8LP8BBWG6990 entering >> state [completed][200] >> 2017-05-12 17:53:58.932851 [DEBUG] mod_conference.c:3289 using channel >> sound prefix: /tmp/fs1.8/share/freeswitch/sounds/en/us/callie >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] >> conference_member.c:1715 Raw Codec Activation Success L16 at 8000hz 1 >> channel 20ms >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] >> conference_member.c:1762 Raw Codec Activation Success L16 at 8000hz 1 >> channel 20ms >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] >> switch_core_codec.c:223 sofia/websocket/nobody at 52LX8LP8BBWG6990 Push >> codec L16:100 >> 2017-05-12 17:53:58.932851 [DEBUG] mod_conference.c:227 Setup timer >> success interval: 20 samples: 160 >> 2017-05-12 17:53:58.932851 [ERR] switch_core_video.c:2868 This function >> is not available, libpng not installed >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] >> conference_loop.c:1306 Setup timer soft success interval: 20 samples: 160 >> from codec PCMU >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [NOTICE] >> switch_channel.c:1104 New Channel sofia/websocket/nobody at 3PXT9NIJPMV6KC8W >> [02b30be2-373c-11e7-913a-499bb33ea37e] >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >> switch_core_state_machine.c:584 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) >> Running State Change CS_NEW (Cur 2 Tot 2) >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >> sofia.c:10028 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W receiving invite >> from 192.168.56.151:53568 version: 1.9.0 git db24869 2017-05-11 >> 18:22:45Z 64bit >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >> sofia.c:11325 Setting NAT mode based on websockets >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >> sofia.c:7247 Channel sofia/websocket/nobody at 3PXT9NIJPMV6KC8W entering >> state [received][100] >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >> sofia.c:7257 Remote SDP: >> 02b30be2-373c-11e7-913a-499bb33ea37e v=0 >> 02b30be2-373c-11e7-913a-499bb33ea37e o=- 716582477 2 IN IP4 >> 192.168.56.151 >> 02b30be2-373c-11e7-913a-499bb33ea37e s=- >> 02b30be2-373c-11e7-913a-499bb33ea37e c=IN IP4 192.168.56.151 >> 02b30be2-373c-11e7-913a-499bb33ea37e t=0 0 >> 02b30be2-373c-11e7-913a-499bb33ea37e m=audio 11866 RTP/AVP 0 100 >> 02b30be2-373c-11e7-913a-499bb33ea37e a=rtpmap:0 PCMU/8000 >> 02b30be2-373c-11e7-913a-499bb33ea37e a=maxptime:20 >> 02b30be2-373c-11e7-913a-499bb33ea37e >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >> switch_core_media.c:5110 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU >> :0:8000:20:64000:1] >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >> switch_core_media.c:5165 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ >> is saved as a match >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >> switch_core_media.c:5110 Audio Codec Compare [:100:0:20:0:1]/[PCMU:0:8000:2 >> 0:64000:1] >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >> switch_core_media.c:3425 Set Codec sofia/websocket/nobody at 3PXT9NIJPMV6KC8W >> PCMU/8000 20 ms 160 samples 64000 bits 1 channels >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >> switch_core_codec.c:111 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W Original >> read codec set to PCMU:0 >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >> switch_core_media.c:5398 No 2833 in SDP. Liberal DTMF mode adding 101 as >> telephone-event. >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >> switch_core_media.c:5427 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W Set >> 2833 dtmf send payload to 101 recv payload to 101 >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >> sofia.c:7670 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) State Change >> CS_NEW -> CS_INIT >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >> switch_core_state_machine.c:603 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) >> State NEW >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >> switch_core_state_machine.c:584 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) >> Running State Change CS_INIT (Cur 2 Tot 2) >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >> switch_core_state_machine.c:627 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) >> State INIT >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >> mod_sofia.c:93 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W SOFIA INIT >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >> switch_core_state_machine.c:40 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W >> Standard INIT >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >> switch_core_state_machine.c:48 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) >> State Change CS_INIT -> CS_ROUTING >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >> switch_core_state_machine.c:627 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) >> State INIT going to sleep >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >> switch_core_state_machine.c:584 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) >> Running State Change CS_ROUTING (Cur 2 Tot 2) >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >> switch_channel.c:2249 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) >> Callstate Change DOWN -> RINGING >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >> switch_core_state_machine.c:643 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) >> State ROUTING >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >> mod_sofia.c:154 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W SOFIA ROUTING >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >> switch_core_state_machine.c:236 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W >> Standard ROUTING >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [INFO] >> mod_dialplan_xml.c:637 Processing unknown <>->3PXT9NIJPMV6KC8W in context >> wss-dialplan >> 02b30be2-373c-11e7-913a-499bb33ea37e Dialplan: >> sofia/websocket/nobody at 3PXT9NIJPMV6KC8W parsing >> [wss-dialplan->wss-dialplan] continue=false >> 02b30be2-373c-11e7-913a-499bb33ea37e Dialplan: >> sofia/websocket/nobody at 3PXT9NIJPMV6KC8W Regex (PASS) [wss-dialplan] >> destination_number(3PXT9NIJPMV6KC8W) =~ /.+/ break=on-false >> 02b30be2-373c-11e7-913a-499bb33ea37e Dialplan: >> sofia/websocket/nobody at 3PXT9NIJPMV6KC8W Action conference(test123) >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >> switch_core_state_machine.c:286 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) >> State Change CS_ROUTING -> CS_EXECUTE >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >> switch_core_state_machine.c:643 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) >> State ROUTING going to sleep >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >> switch_core_state_machine.c:584 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) >> Running State Change CS_EXECUTE (Cur 2 Tot 2) >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >> switch_core_state_machine.c:650 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) >> State EXECUTE >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >> mod_sofia.c:209 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W SOFIA EXECUTE >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >> switch_core_state_machine.c:328 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W >> Standard EXECUTE >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >> switch_core_session.c:2707 Application conference Requires media! >> pre_answering channel sofia/websocket/nobody at 3PXT9NIJPMV6KC8W >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [INFO] >> switch_core_session.c:2709 Sending early media >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >> switch_core_media.c:8144 AUDIO RTP [sofia/websocket/nobody at 3PXT9NIJPMV6KC8W] >> 172.16.19.215 port 28858 -> 192.168.56.151 port 11866 codec: 0 ms: 20 >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >> switch_rtp.c:4096 Starting timer [soft] 160 bytes per 20ms >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >> switch_core_media.c:8447 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W Set >> 2833 dtmf send payload to 101 >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >> switch_core_media.c:8454 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W Set >> 2833 dtmf receive payload to 101 >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >> switch_core_media.c:8477 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W Set rtp >> dtmf delay to 40 >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >> mod_sofia.c:2512 Ring SDP: >> 02b30be2-373c-11e7-913a-499bb33ea37e v=0 >> 02b30be2-373c-11e7-913a-499bb33ea37e o=FreeSWITCH 1494582796 1494582797 >> IN IP4 172.16.19.215 >> 02b30be2-373c-11e7-913a-499bb33ea37e s=FreeSWITCH >> 02b30be2-373c-11e7-913a-499bb33ea37e c=IN IP4 172.16.19.215 >> 02b30be2-373c-11e7-913a-499bb33ea37e t=0 0 >> 02b30be2-373c-11e7-913a-499bb33ea37e m=audio 28858 RTP/AVP 0 101 >> 02b30be2-373c-11e7-913a-499bb33ea37e a=rtpmap:0 PCMU/8000 >> 02b30be2-373c-11e7-913a-499bb33ea37e a=rtpmap:101 telephone-event/8000 >> 02b30be2-373c-11e7-913a-499bb33ea37e a=fmtp:101 0-16 >> 02b30be2-373c-11e7-913a-499bb33ea37e a=ptime:20 >> 02b30be2-373c-11e7-913a-499bb33ea37e a=sendrecv >> 02b30be2-373c-11e7-913a-499bb33ea37e >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [NOTICE] >> mod_sofia.c:2515 Pre-Answer sofia/websocket/nobody at 3PXT9NIJPMV6KC8W! >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >> switch_channel.c:3481 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) >> Callstate Change RINGING -> EARLY >> 02b30be2-373c-11e7-913a-499bb33ea37e EXECUTE >> sofia/websocket/nobody at 3PXT9NIJPMV6KC8W conference(test123) >> 02b30be2-373c-11e7-913a-499bb33ea37e v=0 >> 02b30be2-373c-11e7-913a-499bb33ea37e o=FreeSWITCH 1494582796 1494582798 >> IN IP4 172.16.19.215 >> 02b30be2-373c-11e7-913a-499bb33ea37e s=FreeSWITCH >> 02b30be2-373c-11e7-913a-499bb33ea37e c=IN IP4 172.16.19.215 >> 02b30be2-373c-11e7-913a-499bb33ea37e t=0 0 >> 02b30be2-373c-11e7-913a-499bb33ea37e m=audio 28858 RTP/AVP 0 101 >> 02b30be2-373c-11e7-913a-499bb33ea37e a=rtpmap:0 PCMU/8000 >> 02b30be2-373c-11e7-913a-499bb33ea37e a=rtpmap:101 telephone-event/8000 >> 02b30be2-373c-11e7-913a-499bb33ea37e a=fmtp:101 0-16 >> 02b30be2-373c-11e7-913a-499bb33ea37e a=ptime:20 >> 02b30be2-373c-11e7-913a-499bb33ea37e a=sendrecv >> 02b30be2-373c-11e7-913a-499bb33ea37e >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >> sofia.c:7247 Channel sofia/websocket/nobody at 3PXT9NIJPMV6KC8W entering >> state [early][183] >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [NOTICE] >> mod_conference.c:1829 Channel [sofia/websocket/nobody at 3PXT9NIJPMV6KC8W] >> has been answered >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >> switch_channel.c:3780 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) >> Callstate Change EARLY -> ACTIVE >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >> sofia.c:7247 Channel sofia/websocket/nobody at 3PXT9NIJPMV6KC8W entering >> state [completed][200] >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >> conference_member.c:1715 Raw Codec Activation Success L16 at 8000hz 1 >> channel 20ms >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >> conference_member.c:1762 Raw Codec Activation Success L16 at 8000hz 1 >> channel 20ms >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >> switch_core_codec.c:223 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W Push >> codec L16:100 >> 2017-05-12 17:54:14.092839 [ERR] switch_core_video.c:2868 This function >> is not available, libpng not installed >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >> conference_loop.c:1306 Setup timer soft success interval: 20 samples: 160 >> from codec PCMU >> >> This is where session 02b30be2-373c-11e7-913a-499bb33ea37e wss socket >> gets an ssl errror: >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.952843 [DEBUG] >> sofia.c:7247 Channel sofia/websocket/nobody at 52LX8LP8BBWG6990 entering >> state [terminating][0] >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.952843 [NOTICE] >> sofia.c:8438 Hangup sofia/websocket/nobody at 52LX8LP8BBWG6990 [CS_EXECUTE] >> [NORMAL_UNSPECIFIED] >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [INFO] >> conference_loop.c:1621 Channel leaving conference, cause: NORMAL_UNSPECIFIED >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] >> mod_conference.c:2404 sofia/websocket/nobody at 52LX8LP8BBWG6990 skip >> receive message [TRANSFER] (channel is hungup already) >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] >> switch_core_media.c:11838 sofia/websocket/nobody at 52LX8LP8BBWG6990 skip >> receive message [BITRATE_REQ] (channel is hungup already) >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] >> switch_core_codec.c:248 sofia/websocket/nobody at 52LX8LP8BBWG6990 Restore >> previous codec PCMU:0. >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] >> switch_core_session.c:2884 sofia/websocket/nobody at 52LX8LP8BBWG6990 skip >> receive message [PHONE_EVENT] (channel is hungup already) >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] >> switch_core_state_machine.c:650 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >> State EXECUTE going to sleep >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] >> switch_core_state_machine.c:584 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >> Running State Change CS_HANGUP (Cur 2 Tot 2) >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] >> switch_core_state_machine.c:850 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >> Callstate Change ACTIVE -> HANGUP >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] >> switch_core_state_machine.c:852 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >> State HANGUP >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] >> mod_sofia.c:449 Channel sofia/websocket/nobody at 52LX8LP8BBWG6990 hanging >> up, cause: NORMAL_UNSPECIFIED >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] >> switch_core_state_machine.c:60 sofia/websocket/nobody at 52LX8LP8BBWG6990 >> Standard HANGUP, cause: NORMAL_UNSPECIFIED >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] >> switch_core_state_machine.c:852 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >> State HANGUP going to sleep >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] >> switch_core_state_machine.c:619 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >> State Change CS_HANGUP -> CS_REPORTING >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] >> switch_core_state_machine.c:584 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >> Running State Change CS_REPORTING (Cur 2 Tot 2) >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] >> switch_core_state_machine.c:938 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >> State REPORTING >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] >> switch_core_state_machine.c:174 sofia/websocket/nobody at 52LX8LP8BBWG6990 >> Standard REPORTING, cause: NORMAL_UNSPECIFIED >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] >> switch_core_state_machine.c:938 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >> State REPORTING going to sleep >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] >> switch_core_state_machine.c:610 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >> State Change CS_REPORTING -> CS_DESTROY >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] >> switch_core_session.c:1712 Session 1 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >> Locked, Waiting on external entities >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [NOTICE] >> switch_core_session.c:1730 Session 1 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >> Ended >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [NOTICE] >> switch_core_session.c:1734 Close Channel sofia/websocket/nobody at 52LX8LP8BBWG6990 >> [CS_DESTROY] >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] >> switch_core_state_machine.c:741 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >> Running State Change CS_DESTROY (Cur 1 Tot 2) >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] >> switch_core_state_machine.c:751 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >> State DESTROY >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] >> mod_sofia.c:354 sofia/websocket/nobody at 52LX8LP8BBWG6990 SOFIA DESTROY >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] >> switch_core_state_machine.c:181 sofia/websocket/nobody at 52LX8LP8BBWG6990 >> Standard DESTROY >> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] >> switch_core_state_machine.c:751 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >> State DESTROY going to sleep >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.112842 [DEBUG] >> sofia.c:7247 Channel sofia/websocket/nobody at 3PXT9NIJPMV6KC8W entering >> state [terminating][0] >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.112842 [NOTICE] >> sofia.c:8438 Hangup sofia/websocket/nobody at 3PXT9NIJPMV6KC8W [CS_EXECUTE] >> [NORMAL_UNSPECIFIED] >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [INFO] >> conference_loop.c:1621 Channel leaving conference, cause: NORMAL_UNSPECIFIED >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] >> mod_conference.c:2404 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W skip >> receive message [TRANSFER] (channel is hungup already) >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] >> switch_core_media.c:11838 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W skip >> receive message [BITRATE_REQ] (channel is hungup already) >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] >> switch_core_codec.c:248 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W Restore >> previous codec PCMU:0. >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] >> switch_core_session.c:2884 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W skip >> receive message [PHONE_EVENT] (channel is hungup already) >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] >> switch_core_state_machine.c:650 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) >> State EXECUTE going to sleep >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] >> switch_core_state_machine.c:584 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) >> Running State Change CS_HANGUP (Cur 1 Tot 2) >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] >> switch_core_state_machine.c:850 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) >> Callstate Change ACTIVE -> HANGUP >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] >> switch_core_state_machine.c:852 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) >> State HANGUP >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] >> mod_sofia.c:449 Channel sofia/websocket/nobody at 3PXT9NIJPMV6KC8W hanging >> up, cause: NORMAL_UNSPECIFIED >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] >> switch_core_state_machine.c:60 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W >> Standard HANGUP, cause: NORMAL_UNSPECIFIED >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] >> switch_core_state_machine.c:852 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) >> State HANGUP going to sleep >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] >> switch_core_state_machine.c:619 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) >> State Change CS_HANGUP -> CS_REPORTING >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] >> switch_core_state_machine.c:584 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) >> Running State Change CS_REPORTING (Cur 1 Tot 2) >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] >> switch_core_state_machine.c:938 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) >> State REPORTING >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] >> switch_core_state_machine.c:174 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W >> Standard REPORTING, cause: NORMAL_UNSPECIFIED >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] >> switch_core_state_machine.c:938 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) >> State REPORTING going to sleep >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] >> switch_core_state_machine.c:610 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) >> State Change CS_REPORTING -> CS_DESTROY >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] >> switch_core_session.c:1712 Session 2 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) >> Locked, Waiting on external entities >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [NOTICE] >> switch_core_session.c:1730 Session 2 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) >> Ended >> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [NOTICE] >> switch_core_session.c:1734 Close Channel sofia/websocket/nobody at 3PXT9NIJPMV6KC8W >> [CS_DESTROY] >> >> On Fri, May 12, 2017 at 11:51 AM, Luke Wahlmeier >> wrote: >> >>> Thanks Michael, >>> >>> I am more then happy to setup something with libks if needed. >>> >>> I have figured out some more however. It appears that this only happens >>> when a wss connections session has not fully established and is cleaning up >>> because of timing out. The problem is that it causes another wss >>> connection it to get this ssl error, even if that other wss connection has >>> a fully established and running audio session. It is important to note it >>> does not seem to interrupt audio just the wss sip channel, which I am >>> fairly sure can be reestablished for that audio session w/o an issue. >>> >>> The sessions that is being cleaned up sends the logs messages as its >>> doing it: >>> 2017-05-12 17:32:46.607768 [NOTICE] sofia.c:8438 Hangup >>> sofia/websocket/nobody at 1LF3F6I924P9WH6U [CS_EXECUTE] >>> [NORMAL_UNSPECIFIED] >>> 2017-05-12 17:32:46.627768 [INFO] conference_loop.c:1621 Channel leaving >>> conference, cause: NORMAL_UNSPECIFIED >>> 2017-05-12 17:32:46.627768 [NOTICE] switch_core_session.c:1730 Session >>> 46 (sofia/websocket/nobody at 1LF3F6I924P9WH6U) Ended >>> 2017-05-12 17:32:46.627768 [NOTICE] switch_core_session.c:1734 Close >>> Channel sofia/websocket/nobody at 1LF3F6I924P9WH6U [CS_DESTROY] >>> >>> I have attached the updated python script, it can duplicate this every >>> time now with only 2 connections. I verified with a webRTC client that if >>> I initiate this first connection in the script, let it close, then connect >>> the webRTC client and get full audio, once the first session from the >>> script times out it causes the webRTC wss connection to get an error and >>> close. >>> >>> The webRTC connection is in chrome with sip.js. >>> >>> Sorry the python script is so nasty, was working through any possible >>> duplicated sip session stuff in it to make sure that was not why it was >>> hitting the second connection. >>> >>> >>> On Fri, May 12, 2017 at 10:20 AM, Michael Jerris >>> wrote: >>> >>>> test on master.. work a similar test for verto maybe, this might have >>>> to do with sip specifically trying to keep state. Might make sense to >>>> build something out of libks as it has basically the same web socket code, >>>> and has both client and server web socket support in it, to do a ?real? >>>> test?, instead of this fake sip without any state over web sockets. >>>> >>>> >>>> On May 12, 2017, at 11:42 AM, Luke Wahlmeier >>>> wrote: >>>> >>>> Just got done testing this on v1.6 head and master, both seem to still >>>> have this issue. This box is using libssl version 1.0.1t-1+deb8u6. I am >>>> gonna start digging more into the ws/wss/sofia code to see if I can figure >>>> it out. Any suggestions on debugging this would be appreciated. >>>> >>>> Thanks >>>> Luke >>>> >>>> On Thu, May 11, 2017 at 5:12 PM, Luke Wahlmeier >>>> wrote: >>>> >>>>> Its just in our isolated lab, pretty normal dell xeon server running >>>>> Jessie 8.6. I just want to get it building on the same box I am testing >>>>> with so setting that all up. >>>>> >>>>> I was able to reproduce it w/o DTLS/Srtp. here is a much simpler and >>>>> cleaned up version of the python script. >>>>> >>>>> >>>>> >>>>> On Thu, May 11, 2017 at 4:34 PM, Michael Jerris >>>>> wrote: >>>>> >>>>>> what is ?this environment? ? >>>>>> >>>>>> On May 11, 2017, at 6:31 PM, Luke Wahlmeier >>>>>> wrote: >>>>>> >>>>>> Yeah I can usually get it to happen within about 5 minutes or so of >>>>>> testing. Still getting all setup to build freeswitch in this environment, >>>>>> but I should have it working by tomorrow. I will try more w/o dtls/srtp as >>>>>> well and make sure it does not need to be on. >>>>>> >>>>>> Thanks >>>>>> Luke >>>>>> >>>>>> On Thu, May 11, 2017 at 4:20 PM, Michael Jerris >>>>>> wrote: >>>>>> >>>>>>> if you can reproduce this reliably, i?d try master as well. Unless >>>>>>> this is a bug in openssl, i can?t imagine how dtls would come into play in >>>>>>> something like this. >>>>>>> >>>>>>> > On May 11, 2017, at 5:48 PM, Luke Wahlmeier >>>>>>> wrote: >>>>>>> > >>>>>>> > I keep semi-regularly running into issues using the wss transport >>>>>>> when using dtls/strp/ice. This is on the latest 1.6.17~34~0fc0946 on >>>>>>> Debian jessie, but I am pretty sure it was happening on the last couple >>>>>>> releases as well. >>>>>>> > >>>>>>> > It seems like something bad/wrong happens to the encrypted data >>>>>>> going over the websocket coming from freeswitch when more then 1 websocket >>>>>>> connection are going and so far ice/srtp/dtls also seem to be needed in the >>>>>>> invite to duplicate it. >>>>>>> > >>>>>>> > I have tried many different languages and network/ssl stacks and >>>>>>> keep running into this. It is always on data coming in from freeswitch on >>>>>>> the websocket connection, and its very very random. Sometimes I will get >>>>>>> it 20 times in a row, other times it takes thousands of >>>>>>> connections/sessions before it happen. It also, obviously, completely goes >>>>>>> away if I use plain ws instead wss. >>>>>>> > >>>>>>> > Here are the errors: >>>>>>> > python: >>>>>>> > SSLError: [SSL: DECRYPTION_FAILED_OR_BAD_RECORD_MAC] decryption >>>>>>> failed or bad record mac (_ssl.c:1750) >>>>>>> > c/c++ (stunnel4): >>>>>>> > SSL_read: 1408F119: error:1408F119:SSL >>>>>>> routines:SSL3_GET_RECORD:decryption failed or bad record mac >>>>>>> > Java: >>>>>>> > java.lang.IllegalArgumentException: Bad arguments >>>>>>> > at javax.crypto.Mac.update(Mac.java:509) >>>>>>> > at sun.security.ssl.MAC.compute(MAC.java:135) >>>>>>> > at sun.security.ssl.InputRecord.c >>>>>>> heckMacTags(InputRecord.java:265) >>>>>>> > at sun.security.ssl.InputRecord.decrypt(InputRecord.java:216) >>>>>>> > at sun.security.ssl.EngineInputRe >>>>>>> cord.decrypt(EngineInputRecord.java:177) >>>>>>> > at sun.security.ssl.SSLEngineImpl >>>>>>> .readRecord(SSLEngineImpl.java:974) >>>>>>> > at sun.security.ssl.SSLEngineImpl >>>>>>> .readNetRecord(SSLEngineImpl.java:907) >>>>>>> > at sun.security.ssl.SSLEngineImpl >>>>>>> .unwrap(SSLEngineImpl.java:781) >>>>>>> > at javax.net.ssl.SSLEngine.unwrap(SSLEngine.java:624) >>>>>>> > >>>>>>> > Attached are a simple python script to do the load, my dialplan >>>>>>> and sip_profile. The python script can take a few runs before it see the >>>>>>> error, and I know its not completing the sip or rtp, but even if it does >>>>>>> this still happens. >>>>>>> > >>>>>>> > I have also looked at libsofia-sip-ua/tport/ws.c and I dont see >>>>>>> anything obvious. I am getting setup to build v1.6 head and test this any >>>>>>> guidance on ways I can trouble shoot this better or requests for more info >>>>>>> are very welcome. >>>>>>> > >>>>>>> >>>>>> >>>>>> >>>>>> ____________________________________________________________ >>>>>> _____________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-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/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 > -- 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=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170518/169f606d/attachment-0001.html From bipin at xbipin.com Fri May 19 00:37:14 2017 From: bipin at xbipin.com (Bipin Patel) Date: Fri, 19 May 2017 00:37:14 +0400 Subject: [Freeswitch-users] Busy In-Reply-To: <6075eed2-9e19-080a-87a3-f4f98fca9f11@ednt.de> References: <6075eed2-9e19-080a-87a3-f4f98fca9f11@ednt.de> Message-ID: <15c1d475890.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com> Hi, I had a similar situation, what I did is when some one places a call I set a counter for the calling and called party, then when someone tries to call any one of them I check the counter and if it's one I send them busy or change ringback tone. I'll check my setup tomorrow and let you know the actual module I use On May 18, 2017 8:16:39 PM kk Mailinglist wrote: > When a phone have a running call and a second call comes in i get no > busy and i have redirect to voicemail by busy and the new incoming call > will not redirect to VM. > > I am aware that my phone can have multiple Lines, but how i can signal > that i am busy ? > > 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 From lwahlmeier at gmail.com Fri May 19 00:43:33 2017 From: lwahlmeier at gmail.com (Luke Wahlmeier) Date: Thu, 18 May 2017 14:43:33 -0600 Subject: [Freeswitch-users] WSS SSL errors "decryption failed or bad record mac" under load In-Reply-To: References: <358E7E05-A60A-463A-911F-A921F6CDEAA2@jerris.com> Message-ID: Will do, thanks On Thu, May 18, 2017 at 12:08 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Please do not post logs into email bodies, people use mobile devices to > browse this list. > > Please do no report issues on this list, its a discussion list. > > We have a Bug tracker specifically for issues. > I forgot completely about this thread because its impossible to mentally > remember every one of the email threads in a busy mailing list. > > https://jira.freeswitch.org This has all you need to attach your data > and file a proper report. Otherwise its pure Chaos..... > > > > > On Thu, May 18, 2017 at 12:34 PM, Luke Wahlmeier > wrote: > >> I have been able to verify this is definitely an issue with the sofia wss >> transport. If I put my own ssl proxy (stunnel) in front and use sofia ws >> transport the problem goes away. >> >> I have tried quite a bit of debugging and cant figure out when/how this >> is happening, my best guess is something around the ssl locking on >> connection close, but I have not had the time to dig into that yet. >> >> Thanks >> Luke >> >> On Fri, May 12, 2017 at 12:03 PM, Luke Wahlmeier >> wrote: >> >>> Here is the debug logs from this happening: >>> >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 >>> [NOTICE] switch_channel.c:1104 New Channel sofia/websocket/nobody at 52LX8LP8BBWG6990 >>> [f9a97c98-373b-11e7-9136-499bb33ea37e] >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] >>> switch_core_state_machine.c:584 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >>> Running State Change CS_NEW (Cur 1 Tot 1) >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] >>> sofia.c:10028 sofia/websocket/nobody at 52LX8LP8BBWG6990 receiving invite >>> from 192.168.56.151:53442 version: 1.9.0 git db24869 2017-05-11 >>> 18:22:45Z 64bit >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] >>> sofia.c:11325 Setting NAT mode based on websockets >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] >>> sofia.c:7247 Channel sofia/websocket/nobody at 52LX8LP8BBWG6990 entering >>> state [received][100] >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] >>> sofia.c:7257 Remote SDP: >>> f9a97c98-373b-11e7-9136-499bb33ea37e v=0 >>> f9a97c98-373b-11e7-9136-499bb33ea37e o=- 196478633 2 IN IP4 >>> 192.168.56.151 >>> f9a97c98-373b-11e7-9136-499bb33ea37e s=- >>> f9a97c98-373b-11e7-9136-499bb33ea37e c=IN IP4 192.168.56.151 >>> f9a97c98-373b-11e7-9136-499bb33ea37e t=0 0 >>> f9a97c98-373b-11e7-9136-499bb33ea37e m=audio 42504 RTP/AVP 0 100 >>> f9a97c98-373b-11e7-9136-499bb33ea37e a=rtpmap:0 PCMU/8000 >>> f9a97c98-373b-11e7-9136-499bb33ea37e a=maxptime:20 >>> f9a97c98-373b-11e7-9136-499bb33ea37e >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] >>> switch_core_media.c:5110 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU >>> :0:8000:20:64000:1] >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] >>> switch_core_media.c:5165 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ >>> is saved as a match >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] >>> switch_core_media.c:5110 Audio Codec Compare [:100:0:20:0:1]/[PCMU:0:8000:2 >>> 0:64000:1] >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] >>> switch_core_media.c:3425 Set Codec sofia/websocket/nobody at 52LX8LP8BBWG6990 >>> PCMU/8000 20 ms 160 samples 64000 bits 1 channels >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] >>> switch_core_codec.c:111 sofia/websocket/nobody at 52LX8LP8BBWG6990 >>> Original read codec set to PCMU:0 >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] >>> switch_core_media.c:5398 No 2833 in SDP. Liberal DTMF mode adding 101 as >>> telephone-event. >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] >>> switch_core_media.c:5427 sofia/websocket/nobody at 52LX8LP8BBWG6990 Set >>> 2833 dtmf send payload to 101 recv payload to 101 >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] >>> sofia.c:7670 (sofia/websocket/nobody at 52LX8LP8BBWG6990) State Change >>> CS_NEW -> CS_INIT >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] >>> switch_core_state_machine.c:603 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >>> State NEW >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] >>> switch_core_state_machine.c:584 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >>> Running State Change CS_INIT (Cur 1 Tot 1) >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] >>> switch_core_state_machine.c:627 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >>> State INIT >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] >>> mod_sofia.c:93 sofia/websocket/nobody at 52LX8LP8BBWG6990 SOFIA INIT >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.912851 [DEBUG] >>> switch_core_state_machine.c:40 sofia/websocket/nobody at 52LX8LP8BBWG6990 >>> Standard INIT >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] >>> switch_core_state_machine.c:48 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >>> State Change CS_INIT -> CS_ROUTING >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] >>> switch_core_state_machine.c:627 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >>> State INIT going to sleep >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] >>> switch_core_state_machine.c:584 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >>> Running State Change CS_ROUTING (Cur 1 Tot 1) >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] >>> switch_channel.c:2249 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >>> Callstate Change DOWN -> RINGING >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] >>> switch_core_state_machine.c:643 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >>> State ROUTING >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] >>> mod_sofia.c:154 sofia/websocket/nobody at 52LX8LP8BBWG6990 SOFIA ROUTING >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] >>> switch_core_state_machine.c:236 sofia/websocket/nobody at 52LX8LP8BBWG6990 >>> Standard ROUTING >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [INFO] >>> mod_dialplan_xml.c:637 Processing unknown <>->52LX8LP8BBWG6990 in context >>> wss-dialplan >>> f9a97c98-373b-11e7-9136-499bb33ea37e Dialplan: >>> sofia/websocket/nobody at 52LX8LP8BBWG6990 parsing >>> [wss-dialplan->wss-dialplan] continue=false >>> f9a97c98-373b-11e7-9136-499bb33ea37e Dialplan: >>> sofia/websocket/nobody at 52LX8LP8BBWG6990 Regex (PASS) [wss-dialplan] >>> destination_number(52LX8LP8BBWG6990) =~ /.+/ break=on-false >>> f9a97c98-373b-11e7-9136-499bb33ea37e Dialplan: >>> sofia/websocket/nobody at 52LX8LP8BBWG6990 Action conference(test123) >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] >>> switch_core_state_machine.c:286 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >>> State Change CS_ROUTING -> CS_EXECUTE >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] >>> switch_core_state_machine.c:643 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >>> State ROUTING going to sleep >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] >>> switch_core_state_machine.c:584 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >>> Running State Change CS_EXECUTE (Cur 1 Tot 1) >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] >>> switch_core_state_machine.c:650 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >>> State EXECUTE >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] >>> mod_sofia.c:209 sofia/websocket/nobody at 52LX8LP8BBWG6990 SOFIA EXECUTE >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] >>> switch_core_state_machine.c:328 sofia/websocket/nobody at 52LX8LP8BBWG6990 >>> Standard EXECUTE >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] >>> switch_core_session.c:2707 Application conference Requires media! >>> pre_answering channel sofia/websocket/nobody at 52LX8LP8BBWG6990 >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [INFO] >>> switch_core_session.c:2709 Sending early media >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] >>> switch_core_media.c:8144 AUDIO RTP [sofia/websocket/nobody at 52LX8LP8BBWG6990] >>> 172.16.19.215 port 27662 -> 192.168.56.151 port 42504 codec: 0 ms: 20 >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] >>> switch_rtp.c:4096 Starting timer [soft] 160 bytes per 20ms >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] >>> switch_core_media.c:8447 sofia/websocket/nobody at 52LX8LP8BBWG6990 Set >>> 2833 dtmf send payload to 101 >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] >>> switch_core_media.c:8454 sofia/websocket/nobody at 52LX8LP8BBWG6990 Set >>> 2833 dtmf receive payload to 101 >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] >>> switch_core_media.c:8477 sofia/websocket/nobody at 52LX8LP8BBWG6990 Set >>> rtp dtmf delay to 40 >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] >>> mod_sofia.c:2512 Ring SDP: >>> f9a97c98-373b-11e7-9136-499bb33ea37e v=0 >>> f9a97c98-373b-11e7-9136-499bb33ea37e o=FreeSWITCH 1494583976 1494583977 >>> IN IP4 172.16.19.215 >>> f9a97c98-373b-11e7-9136-499bb33ea37e s=FreeSWITCH >>> f9a97c98-373b-11e7-9136-499bb33ea37e c=IN IP4 172.16.19.215 >>> f9a97c98-373b-11e7-9136-499bb33ea37e t=0 0 >>> f9a97c98-373b-11e7-9136-499bb33ea37e m=audio 27662 RTP/AVP 0 101 >>> f9a97c98-373b-11e7-9136-499bb33ea37e a=rtpmap:0 PCMU/8000 >>> f9a97c98-373b-11e7-9136-499bb33ea37e a=rtpmap:101 telephone-event/8000 >>> f9a97c98-373b-11e7-9136-499bb33ea37e a=fmtp:101 0-16 >>> f9a97c98-373b-11e7-9136-499bb33ea37e a=ptime:20 >>> f9a97c98-373b-11e7-9136-499bb33ea37e a=sendrecv >>> f9a97c98-373b-11e7-9136-499bb33ea37e >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 >>> [NOTICE] mod_sofia.c:2515 Pre-Answer sofia/websocket/nobody at 52LX8LP >>> 8BBWG6990! >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] >>> switch_channel.c:3481 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >>> Callstate Change RINGING -> EARLY >>> f9a97c98-373b-11e7-9136-499bb33ea37e EXECUTE >>> sofia/websocket/nobody at 52LX8LP8BBWG6990 conference(test123) >>> f9a97c98-373b-11e7-9136-499bb33ea37e v=0 >>> f9a97c98-373b-11e7-9136-499bb33ea37e o=FreeSWITCH 1494583976 1494583978 >>> IN IP4 172.16.19.215 >>> f9a97c98-373b-11e7-9136-499bb33ea37e s=FreeSWITCH >>> f9a97c98-373b-11e7-9136-499bb33ea37e c=IN IP4 172.16.19.215 >>> f9a97c98-373b-11e7-9136-499bb33ea37e t=0 0 >>> f9a97c98-373b-11e7-9136-499bb33ea37e m=audio 27662 RTP/AVP 0 101 >>> f9a97c98-373b-11e7-9136-499bb33ea37e a=rtpmap:0 PCMU/8000 >>> f9a97c98-373b-11e7-9136-499bb33ea37e a=rtpmap:101 telephone-event/8000 >>> f9a97c98-373b-11e7-9136-499bb33ea37e a=fmtp:101 0-16 >>> f9a97c98-373b-11e7-9136-499bb33ea37e a=ptime:20 >>> f9a97c98-373b-11e7-9136-499bb33ea37e a=sendrecv >>> f9a97c98-373b-11e7-9136-499bb33ea37e >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] >>> sofia.c:7247 Channel sofia/websocket/nobody at 52LX8LP8BBWG6990 entering >>> state [early][183] >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 >>> [NOTICE] mod_conference.c:1829 Channel [sofia/websocket/nobody at 52LX8LP8BBWG6990] >>> has been answered >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] >>> switch_channel.c:3780 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >>> Callstate Change EARLY -> ACTIVE >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] >>> sofia.c:7247 Channel sofia/websocket/nobody at 52LX8LP8BBWG6990 entering >>> state [completed][200] >>> 2017-05-12 17:53:58.932851 [DEBUG] mod_conference.c:3289 using channel >>> sound prefix: /tmp/fs1.8/share/freeswitch/sounds/en/us/callie >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] >>> conference_member.c:1715 Raw Codec Activation Success L16 at 8000hz 1 >>> channel 20ms >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] >>> conference_member.c:1762 Raw Codec Activation Success L16 at 8000hz 1 >>> channel 20ms >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] >>> switch_core_codec.c:223 sofia/websocket/nobody at 52LX8LP8BBWG6990 Push >>> codec L16:100 >>> 2017-05-12 17:53:58.932851 [DEBUG] mod_conference.c:227 Setup timer >>> success interval: 20 samples: 160 >>> 2017-05-12 17:53:58.932851 [ERR] switch_core_video.c:2868 This function >>> is not available, libpng not installed >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:53:58.932851 [DEBUG] >>> conference_loop.c:1306 Setup timer soft success interval: 20 samples: 160 >>> from codec PCMU >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 >>> [NOTICE] switch_channel.c:1104 New Channel sofia/websocket/nobody at 3PXT9NIJPMV6KC8W >>> [02b30be2-373c-11e7-913a-499bb33ea37e] >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >>> switch_core_state_machine.c:584 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) >>> Running State Change CS_NEW (Cur 2 Tot 2) >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >>> sofia.c:10028 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W receiving invite >>> from 192.168.56.151:53568 version: 1.9.0 git db24869 2017-05-11 >>> 18:22:45Z 64bit >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >>> sofia.c:11325 Setting NAT mode based on websockets >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >>> sofia.c:7247 Channel sofia/websocket/nobody at 3PXT9NIJPMV6KC8W entering >>> state [received][100] >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >>> sofia.c:7257 Remote SDP: >>> 02b30be2-373c-11e7-913a-499bb33ea37e v=0 >>> 02b30be2-373c-11e7-913a-499bb33ea37e o=- 716582477 2 IN IP4 >>> 192.168.56.151 >>> 02b30be2-373c-11e7-913a-499bb33ea37e s=- >>> 02b30be2-373c-11e7-913a-499bb33ea37e c=IN IP4 192.168.56.151 >>> 02b30be2-373c-11e7-913a-499bb33ea37e t=0 0 >>> 02b30be2-373c-11e7-913a-499bb33ea37e m=audio 11866 RTP/AVP 0 100 >>> 02b30be2-373c-11e7-913a-499bb33ea37e a=rtpmap:0 PCMU/8000 >>> 02b30be2-373c-11e7-913a-499bb33ea37e a=maxptime:20 >>> 02b30be2-373c-11e7-913a-499bb33ea37e >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >>> switch_core_media.c:5110 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU >>> :0:8000:20:64000:1] >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >>> switch_core_media.c:5165 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ >>> is saved as a match >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >>> switch_core_media.c:5110 Audio Codec Compare [:100:0:20:0:1]/[PCMU:0:8000:2 >>> 0:64000:1] >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >>> switch_core_media.c:3425 Set Codec sofia/websocket/nobody at 3PXT9NIJPMV6KC8W >>> PCMU/8000 20 ms 160 samples 64000 bits 1 channels >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >>> switch_core_codec.c:111 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W >>> Original read codec set to PCMU:0 >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >>> switch_core_media.c:5398 No 2833 in SDP. Liberal DTMF mode adding 101 as >>> telephone-event. >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >>> switch_core_media.c:5427 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W Set >>> 2833 dtmf send payload to 101 recv payload to 101 >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >>> sofia.c:7670 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) State Change >>> CS_NEW -> CS_INIT >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >>> switch_core_state_machine.c:603 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) >>> State NEW >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >>> switch_core_state_machine.c:584 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) >>> Running State Change CS_INIT (Cur 2 Tot 2) >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >>> switch_core_state_machine.c:627 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) >>> State INIT >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >>> mod_sofia.c:93 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W SOFIA INIT >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >>> switch_core_state_machine.c:40 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W >>> Standard INIT >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >>> switch_core_state_machine.c:48 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) >>> State Change CS_INIT -> CS_ROUTING >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >>> switch_core_state_machine.c:627 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) >>> State INIT going to sleep >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >>> switch_core_state_machine.c:584 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) >>> Running State Change CS_ROUTING (Cur 2 Tot 2) >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >>> switch_channel.c:2249 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) >>> Callstate Change DOWN -> RINGING >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >>> switch_core_state_machine.c:643 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) >>> State ROUTING >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >>> mod_sofia.c:154 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W SOFIA ROUTING >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >>> switch_core_state_machine.c:236 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W >>> Standard ROUTING >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [INFO] >>> mod_dialplan_xml.c:637 Processing unknown <>->3PXT9NIJPMV6KC8W in context >>> wss-dialplan >>> 02b30be2-373c-11e7-913a-499bb33ea37e Dialplan: >>> sofia/websocket/nobody at 3PXT9NIJPMV6KC8W parsing >>> [wss-dialplan->wss-dialplan] continue=false >>> 02b30be2-373c-11e7-913a-499bb33ea37e Dialplan: >>> sofia/websocket/nobody at 3PXT9NIJPMV6KC8W Regex (PASS) [wss-dialplan] >>> destination_number(3PXT9NIJPMV6KC8W) =~ /.+/ break=on-false >>> 02b30be2-373c-11e7-913a-499bb33ea37e Dialplan: >>> sofia/websocket/nobody at 3PXT9NIJPMV6KC8W Action conference(test123) >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >>> switch_core_state_machine.c:286 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) >>> State Change CS_ROUTING -> CS_EXECUTE >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >>> switch_core_state_machine.c:643 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) >>> State ROUTING going to sleep >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >>> switch_core_state_machine.c:584 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) >>> Running State Change CS_EXECUTE (Cur 2 Tot 2) >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >>> switch_core_state_machine.c:650 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) >>> State EXECUTE >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >>> mod_sofia.c:209 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W SOFIA EXECUTE >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >>> switch_core_state_machine.c:328 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W >>> Standard EXECUTE >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >>> switch_core_session.c:2707 Application conference Requires media! >>> pre_answering channel sofia/websocket/nobody at 3PXT9NIJPMV6KC8W >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [INFO] >>> switch_core_session.c:2709 Sending early media >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >>> switch_core_media.c:8144 AUDIO RTP [sofia/websocket/nobody at 3PXT9NIJPMV6KC8W] >>> 172.16.19.215 port 28858 -> 192.168.56.151 port 11866 codec: 0 ms: 20 >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >>> switch_rtp.c:4096 Starting timer [soft] 160 bytes per 20ms >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >>> switch_core_media.c:8447 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W Set >>> 2833 dtmf send payload to 101 >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >>> switch_core_media.c:8454 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W Set >>> 2833 dtmf receive payload to 101 >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >>> switch_core_media.c:8477 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W Set >>> rtp dtmf delay to 40 >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >>> mod_sofia.c:2512 Ring SDP: >>> 02b30be2-373c-11e7-913a-499bb33ea37e v=0 >>> 02b30be2-373c-11e7-913a-499bb33ea37e o=FreeSWITCH 1494582796 1494582797 >>> IN IP4 172.16.19.215 >>> 02b30be2-373c-11e7-913a-499bb33ea37e s=FreeSWITCH >>> 02b30be2-373c-11e7-913a-499bb33ea37e c=IN IP4 172.16.19.215 >>> 02b30be2-373c-11e7-913a-499bb33ea37e t=0 0 >>> 02b30be2-373c-11e7-913a-499bb33ea37e m=audio 28858 RTP/AVP 0 101 >>> 02b30be2-373c-11e7-913a-499bb33ea37e a=rtpmap:0 PCMU/8000 >>> 02b30be2-373c-11e7-913a-499bb33ea37e a=rtpmap:101 telephone-event/8000 >>> 02b30be2-373c-11e7-913a-499bb33ea37e a=fmtp:101 0-16 >>> 02b30be2-373c-11e7-913a-499bb33ea37e a=ptime:20 >>> 02b30be2-373c-11e7-913a-499bb33ea37e a=sendrecv >>> 02b30be2-373c-11e7-913a-499bb33ea37e >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 >>> [NOTICE] mod_sofia.c:2515 Pre-Answer sofia/websocket/nobody at 3PXT9NI >>> JPMV6KC8W! >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >>> switch_channel.c:3481 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) >>> Callstate Change RINGING -> EARLY >>> 02b30be2-373c-11e7-913a-499bb33ea37e EXECUTE >>> sofia/websocket/nobody at 3PXT9NIJPMV6KC8W conference(test123) >>> 02b30be2-373c-11e7-913a-499bb33ea37e v=0 >>> 02b30be2-373c-11e7-913a-499bb33ea37e o=FreeSWITCH 1494582796 1494582798 >>> IN IP4 172.16.19.215 >>> 02b30be2-373c-11e7-913a-499bb33ea37e s=FreeSWITCH >>> 02b30be2-373c-11e7-913a-499bb33ea37e c=IN IP4 172.16.19.215 >>> 02b30be2-373c-11e7-913a-499bb33ea37e t=0 0 >>> 02b30be2-373c-11e7-913a-499bb33ea37e m=audio 28858 RTP/AVP 0 101 >>> 02b30be2-373c-11e7-913a-499bb33ea37e a=rtpmap:0 PCMU/8000 >>> 02b30be2-373c-11e7-913a-499bb33ea37e a=rtpmap:101 telephone-event/8000 >>> 02b30be2-373c-11e7-913a-499bb33ea37e a=fmtp:101 0-16 >>> 02b30be2-373c-11e7-913a-499bb33ea37e a=ptime:20 >>> 02b30be2-373c-11e7-913a-499bb33ea37e a=sendrecv >>> 02b30be2-373c-11e7-913a-499bb33ea37e >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >>> sofia.c:7247 Channel sofia/websocket/nobody at 3PXT9NIJPMV6KC8W entering >>> state [early][183] >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 >>> [NOTICE] mod_conference.c:1829 Channel [sofia/websocket/nobody at 3PXT9NIJPMV6KC8W] >>> has been answered >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >>> switch_channel.c:3780 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) >>> Callstate Change EARLY -> ACTIVE >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >>> sofia.c:7247 Channel sofia/websocket/nobody at 3PXT9NIJPMV6KC8W entering >>> state [completed][200] >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >>> conference_member.c:1715 Raw Codec Activation Success L16 at 8000hz 1 >>> channel 20ms >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >>> conference_member.c:1762 Raw Codec Activation Success L16 at 8000hz 1 >>> channel 20ms >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >>> switch_core_codec.c:223 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W Push >>> codec L16:100 >>> 2017-05-12 17:54:14.092839 [ERR] switch_core_video.c:2868 This function >>> is not available, libpng not installed >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:14.092839 [DEBUG] >>> conference_loop.c:1306 Setup timer soft success interval: 20 samples: 160 >>> from codec PCMU >>> >>> This is where session 02b30be2-373c-11e7-913a-499bb33ea37e wss socket >>> gets an ssl errror: >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.952843 [DEBUG] >>> sofia.c:7247 Channel sofia/websocket/nobody at 52LX8LP8BBWG6990 entering >>> state [terminating][0] >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.952843 >>> [NOTICE] sofia.c:8438 Hangup sofia/websocket/nobody at 52LX8LP8BBWG6990 >>> [CS_EXECUTE] [NORMAL_UNSPECIFIED] >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [INFO] >>> conference_loop.c:1621 Channel leaving conference, cause: NORMAL_UNSPECIFIED >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] >>> mod_conference.c:2404 sofia/websocket/nobody at 52LX8LP8BBWG6990 skip >>> receive message [TRANSFER] (channel is hungup already) >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] >>> switch_core_media.c:11838 sofia/websocket/nobody at 52LX8LP8BBWG6990 skip >>> receive message [BITRATE_REQ] (channel is hungup already) >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] >>> switch_core_codec.c:248 sofia/websocket/nobody at 52LX8LP8BBWG6990 Restore >>> previous codec PCMU:0. >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] >>> switch_core_session.c:2884 sofia/websocket/nobody at 52LX8LP8BBWG6990 skip >>> receive message [PHONE_EVENT] (channel is hungup already) >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] >>> switch_core_state_machine.c:650 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >>> State EXECUTE going to sleep >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] >>> switch_core_state_machine.c:584 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >>> Running State Change CS_HANGUP (Cur 2 Tot 2) >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] >>> switch_core_state_machine.c:850 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >>> Callstate Change ACTIVE -> HANGUP >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] >>> switch_core_state_machine.c:852 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >>> State HANGUP >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] >>> mod_sofia.c:449 Channel sofia/websocket/nobody at 52LX8LP8BBWG6990 hanging >>> up, cause: NORMAL_UNSPECIFIED >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] >>> switch_core_state_machine.c:60 sofia/websocket/nobody at 52LX8LP8BBWG6990 >>> Standard HANGUP, cause: NORMAL_UNSPECIFIED >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] >>> switch_core_state_machine.c:852 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >>> State HANGUP going to sleep >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] >>> switch_core_state_machine.c:619 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >>> State Change CS_HANGUP -> CS_REPORTING >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] >>> switch_core_state_machine.c:584 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >>> Running State Change CS_REPORTING (Cur 2 Tot 2) >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] >>> switch_core_state_machine.c:938 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >>> State REPORTING >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] >>> switch_core_state_machine.c:174 sofia/websocket/nobody at 52LX8LP8BBWG6990 >>> Standard REPORTING, cause: NORMAL_UNSPECIFIED >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] >>> switch_core_state_machine.c:938 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >>> State REPORTING going to sleep >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] >>> switch_core_state_machine.c:610 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >>> State Change CS_REPORTING -> CS_DESTROY >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] >>> switch_core_session.c:1712 Session 1 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >>> Locked, Waiting on external entities >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 >>> [NOTICE] switch_core_session.c:1730 Session 1 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >>> Ended >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 >>> [NOTICE] switch_core_session.c:1734 Close Channel >>> sofia/websocket/nobody at 52LX8LP8BBWG6990 [CS_DESTROY] >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] >>> switch_core_state_machine.c:741 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >>> Running State Change CS_DESTROY (Cur 1 Tot 2) >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] >>> switch_core_state_machine.c:751 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >>> State DESTROY >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] >>> mod_sofia.c:354 sofia/websocket/nobody at 52LX8LP8BBWG6990 SOFIA DESTROY >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] >>> switch_core_state_machine.c:181 sofia/websocket/nobody at 52LX8LP8BBWG6990 >>> Standard DESTROY >>> f9a97c98-373b-11e7-9136-499bb33ea37e 2017-05-12 17:54:30.972859 [DEBUG] >>> switch_core_state_machine.c:751 (sofia/websocket/nobody at 52LX8LP8BBWG6990) >>> State DESTROY going to sleep >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.112842 [DEBUG] >>> sofia.c:7247 Channel sofia/websocket/nobody at 3PXT9NIJPMV6KC8W entering >>> state [terminating][0] >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.112842 >>> [NOTICE] sofia.c:8438 Hangup sofia/websocket/nobody at 3PXT9NIJPMV6KC8W >>> [CS_EXECUTE] [NORMAL_UNSPECIFIED] >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [INFO] >>> conference_loop.c:1621 Channel leaving conference, cause: NORMAL_UNSPECIFIED >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] >>> mod_conference.c:2404 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W skip >>> receive message [TRANSFER] (channel is hungup already) >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] >>> switch_core_media.c:11838 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W skip >>> receive message [BITRATE_REQ] (channel is hungup already) >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] >>> switch_core_codec.c:248 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W Restore >>> previous codec PCMU:0. >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] >>> switch_core_session.c:2884 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W skip >>> receive message [PHONE_EVENT] (channel is hungup already) >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] >>> switch_core_state_machine.c:650 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) >>> State EXECUTE going to sleep >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] >>> switch_core_state_machine.c:584 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) >>> Running State Change CS_HANGUP (Cur 1 Tot 2) >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] >>> switch_core_state_machine.c:850 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) >>> Callstate Change ACTIVE -> HANGUP >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] >>> switch_core_state_machine.c:852 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) >>> State HANGUP >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] >>> mod_sofia.c:449 Channel sofia/websocket/nobody at 3PXT9NIJPMV6KC8W hanging >>> up, cause: NORMAL_UNSPECIFIED >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] >>> switch_core_state_machine.c:60 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W >>> Standard HANGUP, cause: NORMAL_UNSPECIFIED >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] >>> switch_core_state_machine.c:852 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) >>> State HANGUP going to sleep >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] >>> switch_core_state_machine.c:619 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) >>> State Change CS_HANGUP -> CS_REPORTING >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] >>> switch_core_state_machine.c:584 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) >>> Running State Change CS_REPORTING (Cur 1 Tot 2) >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] >>> switch_core_state_machine.c:938 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) >>> State REPORTING >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] >>> switch_core_state_machine.c:174 sofia/websocket/nobody at 3PXT9NIJPMV6KC8W >>> Standard REPORTING, cause: NORMAL_UNSPECIFIED >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] >>> switch_core_state_machine.c:938 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) >>> State REPORTING going to sleep >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] >>> switch_core_state_machine.c:610 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) >>> State Change CS_REPORTING -> CS_DESTROY >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 [DEBUG] >>> switch_core_session.c:1712 Session 2 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) >>> Locked, Waiting on external entities >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 >>> [NOTICE] switch_core_session.c:1730 Session 2 (sofia/websocket/nobody at 3PXT9NIJPMV6KC8W) >>> Ended >>> 02b30be2-373c-11e7-913a-499bb33ea37e 2017-05-12 17:54:46.132852 >>> [NOTICE] switch_core_session.c:1734 Close Channel >>> sofia/websocket/nobody at 3PXT9NIJPMV6KC8W [CS_DESTROY] >>> >>> On Fri, May 12, 2017 at 11:51 AM, Luke Wahlmeier >>> wrote: >>> >>>> Thanks Michael, >>>> >>>> I am more then happy to setup something with libks if needed. >>>> >>>> I have figured out some more however. It appears that this only >>>> happens when a wss connections session has not fully established and is >>>> cleaning up because of timing out. The problem is that it causes another >>>> wss connection it to get this ssl error, even if that other wss connection >>>> has a fully established and running audio session. It is important to note >>>> it does not seem to interrupt audio just the wss sip channel, which I am >>>> fairly sure can be reestablished for that audio session w/o an issue. >>>> >>>> The sessions that is being cleaned up sends the logs messages as its >>>> doing it: >>>> 2017-05-12 17:32:46.607768 [NOTICE] sofia.c:8438 Hangup >>>> sofia/websocket/nobody at 1LF3F6I924P9WH6U [CS_EXECUTE] >>>> [NORMAL_UNSPECIFIED] >>>> 2017-05-12 17:32:46.627768 [INFO] conference_loop.c:1621 Channel >>>> leaving conference, cause: NORMAL_UNSPECIFIED >>>> 2017-05-12 17:32:46.627768 [NOTICE] switch_core_session.c:1730 Session >>>> 46 (sofia/websocket/nobody at 1LF3F6I924P9WH6U) Ended >>>> 2017-05-12 17:32:46.627768 [NOTICE] switch_core_session.c:1734 Close >>>> Channel sofia/websocket/nobody at 1LF3F6I924P9WH6U [CS_DESTROY] >>>> >>>> I have attached the updated python script, it can duplicate this every >>>> time now with only 2 connections. I verified with a webRTC client that if >>>> I initiate this first connection in the script, let it close, then connect >>>> the webRTC client and get full audio, once the first session from the >>>> script times out it causes the webRTC wss connection to get an error and >>>> close. >>>> >>>> The webRTC connection is in chrome with sip.js. >>>> >>>> Sorry the python script is so nasty, was working through any possible >>>> duplicated sip session stuff in it to make sure that was not why it was >>>> hitting the second connection. >>>> >>>> >>>> On Fri, May 12, 2017 at 10:20 AM, Michael Jerris >>>> wrote: >>>> >>>>> test on master.. work a similar test for verto maybe, this might have >>>>> to do with sip specifically trying to keep state. Might make sense to >>>>> build something out of libks as it has basically the same web socket code, >>>>> and has both client and server web socket support in it, to do a ?real? >>>>> test?, instead of this fake sip without any state over web sockets. >>>>> >>>>> >>>>> On May 12, 2017, at 11:42 AM, Luke Wahlmeier >>>>> wrote: >>>>> >>>>> Just got done testing this on v1.6 head and master, both seem to still >>>>> have this issue. This box is using libssl version 1.0.1t-1+deb8u6. I am >>>>> gonna start digging more into the ws/wss/sofia code to see if I can figure >>>>> it out. Any suggestions on debugging this would be appreciated. >>>>> >>>>> Thanks >>>>> Luke >>>>> >>>>> On Thu, May 11, 2017 at 5:12 PM, Luke Wahlmeier >>>>> wrote: >>>>> >>>>>> Its just in our isolated lab, pretty normal dell xeon server running >>>>>> Jessie 8.6. I just want to get it building on the same box I am testing >>>>>> with so setting that all up. >>>>>> >>>>>> I was able to reproduce it w/o DTLS/Srtp. here is a much simpler and >>>>>> cleaned up version of the python script. >>>>>> >>>>>> >>>>>> >>>>>> On Thu, May 11, 2017 at 4:34 PM, Michael Jerris >>>>>> wrote: >>>>>> >>>>>>> what is ?this environment? ? >>>>>>> >>>>>>> On May 11, 2017, at 6:31 PM, Luke Wahlmeier >>>>>>> wrote: >>>>>>> >>>>>>> Yeah I can usually get it to happen within about 5 minutes or so of >>>>>>> testing. Still getting all setup to build freeswitch in this environment, >>>>>>> but I should have it working by tomorrow. I will try more w/o dtls/srtp as >>>>>>> well and make sure it does not need to be on. >>>>>>> >>>>>>> Thanks >>>>>>> Luke >>>>>>> >>>>>>> On Thu, May 11, 2017 at 4:20 PM, Michael Jerris >>>>>>> wrote: >>>>>>> >>>>>>>> if you can reproduce this reliably, i?d try master as well. Unless >>>>>>>> this is a bug in openssl, i can?t imagine how dtls would come into play in >>>>>>>> something like this. >>>>>>>> >>>>>>>> > On May 11, 2017, at 5:48 PM, Luke Wahlmeier >>>>>>>> wrote: >>>>>>>> > >>>>>>>> > I keep semi-regularly running into issues using the wss transport >>>>>>>> when using dtls/strp/ice. This is on the latest 1.6.17~34~0fc0946 on >>>>>>>> Debian jessie, but I am pretty sure it was happening on the last couple >>>>>>>> releases as well. >>>>>>>> > >>>>>>>> > It seems like something bad/wrong happens to the encrypted data >>>>>>>> going over the websocket coming from freeswitch when more then 1 websocket >>>>>>>> connection are going and so far ice/srtp/dtls also seem to be needed in the >>>>>>>> invite to duplicate it. >>>>>>>> > >>>>>>>> > I have tried many different languages and network/ssl stacks and >>>>>>>> keep running into this. It is always on data coming in from freeswitch on >>>>>>>> the websocket connection, and its very very random. Sometimes I will get >>>>>>>> it 20 times in a row, other times it takes thousands of >>>>>>>> connections/sessions before it happen. It also, obviously, completely goes >>>>>>>> away if I use plain ws instead wss. >>>>>>>> > >>>>>>>> > Here are the errors: >>>>>>>> > python: >>>>>>>> > SSLError: [SSL: DECRYPTION_FAILED_OR_BAD_RECORD_MAC] decryption >>>>>>>> failed or bad record mac (_ssl.c:1750) >>>>>>>> > c/c++ (stunnel4): >>>>>>>> > SSL_read: 1408F119: error:1408F119:SSL >>>>>>>> routines:SSL3_GET_RECORD:decryption failed or bad record mac >>>>>>>> > Java: >>>>>>>> > java.lang.IllegalArgumentException: Bad arguments >>>>>>>> > at javax.crypto.Mac.update(Mac.java:509) >>>>>>>> > at sun.security.ssl.MAC.compute(MAC.java:135) >>>>>>>> > at sun.security.ssl.InputRecord.c >>>>>>>> heckMacTags(InputRecord.java:265) >>>>>>>> > at sun.security.ssl.InputRecord.decrypt(InputRecord.java:216) >>>>>>>> > at sun.security.ssl.EngineInputRe >>>>>>>> cord.decrypt(EngineInputRecord.java:177) >>>>>>>> > at sun.security.ssl.SSLEngineImpl >>>>>>>> .readRecord(SSLEngineImpl.java:974) >>>>>>>> > at sun.security.ssl.SSLEngineImpl >>>>>>>> .readNetRecord(SSLEngineImpl.java:907) >>>>>>>> > at sun.security.ssl.SSLEngineImpl >>>>>>>> .unwrap(SSLEngineImpl.java:781) >>>>>>>> > at javax.net.ssl.SSLEngine.unwrap(SSLEngine.java:624) >>>>>>>> > >>>>>>>> > Attached are a simple python script to do the load, my dialplan >>>>>>>> and sip_profile. The python script can take a few runs before it see the >>>>>>>> error, and I know its not completing the sip or rtp, but even if it does >>>>>>>> this still happens. >>>>>>>> > >>>>>>>> > I have also looked at libsofia-sip-ua/tport/ws.c and I dont see >>>>>>>> anything obvious. I am getting setup to build v1.6 head and test this any >>>>>>>> guidance on ways I can trouble shoot this better or requests for more info >>>>>>>> are very welcome. >>>>>>>> > >>>>>>>> >>>>>>> >>>>>>> >>>>>>> ____________________________________________________________ >>>>>>> _____________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://confluence.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-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/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 >> > > > > -- > 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 <(919)%20386-9900> > > 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170518/53825c63/attachment-0001.html From brian at freeswitch.org Fri May 19 01:18:13 2017 From: brian at freeswitch.org (Brian West) Date: Thu, 18 May 2017 16:18:13 -0500 Subject: [Freeswitch-users] Busy In-Reply-To: <15c1d475890.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com> References: <6075eed2-9e19-080a-87a3-f4f98fca9f11@ednt.de> <15c1d475890.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com> Message-ID: mod_limit. On Thu, May 18, 2017 at 3:37 PM, Bipin Patel wrote: > Hi, > > I had a similar situation, what I did is when some one places a call I set > a counter for the calling and called party, then when someone tries to call > any one of them I check the counter and if it's one I send them busy or > change ringback tone. I'll check my setup tomorrow and let you know the > actual module I use > > > > > On May 18, 2017 8:16:39 PM kk Mailinglist wrote: > > > When a phone have a running call and a second call comes in i get no > > busy and i have redirect to voicemail by busy and the new incoming call > > will not redirect to VM. > > > > I am aware that my phone can have multiple Lines, but how i can signal > > that i am busy ? > > > > 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 > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Book a phone call (CST) Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170518/5aa669da/attachment.html From brian at freeswitch.org Fri May 19 01:19:43 2017 From: brian at freeswitch.org (Brian West) Date: Thu, 18 May 2017 16:19:43 -0500 Subject: [Freeswitch-users] Uninstalling freeswitch after compiling and installing from source In-Reply-To: References: Message-ID: The debian packages we build have SCTP enabled by default. Just an FYI. /b On Thu, May 18, 2017 at 11:46 AM, David Villasmil < david.villasmil.work at gmail.com> wrote: > I'm assuming you're starting off with FS, If you're on debian and this is > a fresh install, you're probably better off reinstalling debian and > installing from apt following: > > https://freeswitch.org/confluence/plugins/servlet/ > mobile#content/view/9633817 > > On the "easy way" part. > > > > On Thu, May 18, 2017 at 6:21 PM Michael Jerris wrote: > >> Depends how you configured, when compiling from source configs default to >> /usr/local/freeswitch/conf >> >> On May 17, 2017, at 7:02 PM, Ravi sanyal >> wrote: >> >> Hi, >> >> I made a noob mistake and ran the script for installing freeswitch from >> binaries without doing logs on the process and fs_cli is complaining that >> commands are not found. >> >> specifically, sofia status, or sofia status profile but im sure >> that it extends further than that. I'm dont know how to get a list of the >> commands available. typing in help (not /help) at the prompt says it can't >> find a command. >> >> So i've installed it and it doesn't seem to be working, and don't know >> how to uninstall it. >> >> Where can i find a list of acceptable commands for the configure command? >> >> Im trying to enable sctp, but when looking through the logs for it you >> have a line: >> >> checking whether SCTP is supported... yes >> >> is this the same as checking with the --enable-sctp flag? (when looking >> at the logs for the added command there's no extra lines pertaining to >> enable-sctp as if you didn't specify it). >> >> I'm running the latest build from the github repository. >> >> When running this command: >> >> ./configure --enable-sctp > /home/fleetchat/config.log >> >> it keeps on displaying errors on output, but don't know if they're >> serious or not. >> >> ./configure: line 31222: php-config: command not found >> ./configure: line 31223: php-config: command not found >> ./configure: line 31224: php-config: command not found >> ./configure: line 31225: php: command not found >> ./configure: line 31226: php-config: command not found >> ./configure: line 31227: php-config: command not found >> ./configure: line 5306: AX_COMPILER_VENDOR: command not found >> rm: cannot remove 'conftest*': No such file or directory >> ./configure: line 19537: test: 3.16.0-4-amd64: integer expression expected >> ./configure: line 19819: test: 3.16.0-4-amd64: integer expression expected >> config.status: WARNING: 'apr-config.in' seems to ignore the >> --datarootdir setting >> rm: cannot remove 'libtoolT': No such file or directory >> config.status: WARNING: 'Makefile.in' seems to ignore the --datarootdir >> setting >> config.status: WARNING: 'lib/Makefile.in' seems to ignore the >> --datarootdir setting >> configure: WARNING: ** STUN support disabled ** >> >> Im sure it's somewhere but i also don't know where the configuration >> files should be located when building from source. i placed them in >> /etc/freeswitch. From memory that was where the installation instructions >> said they should be located. >> >> The configuration files were downloaded from the ubuntu repository >> command freeswitch-meta-vanilla >> >> The reason they come from the ubuntu repository is because i had >> freeswitch set up on ubuntu 16.04 but found out debian was the recommended >> OS. I'm running freeswitch on debian 8.8 now. >> >> Nathan >> >> -- >> >> *Virtual Guard Ltd* >> *info at virtualguard.co.nz * >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Book a phone call (CST) Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170518/5f022afa/attachment-0001.html From david.villasmil.work at gmail.com Fri May 19 03:02:24 2017 From: david.villasmil.work at gmail.com (David Villasmil) Date: Thu, 18 May 2017 23:02:24 +0000 Subject: [Freeswitch-users] Busy In-Reply-To: References: <6075eed2-9e19-080a-87a3-f4f98fca9f11@ednt.de> <15c1d475890.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com> Message-ID: This guy is talking about fusionpbx... On Thu, May 18, 2017 at 11:19 PM Brian West wrote: > mod_limit. > > On Thu, May 18, 2017 at 3:37 PM, Bipin Patel wrote: > >> Hi, >> >> I had a similar situation, what I did is when some one places a call I set >> a counter for the calling and called party, then when someone tries to >> call >> any one of them I check the counter and if it's one I send them busy or >> change ringback tone. I'll check my setup tomorrow and let you know the >> actual module I use >> >> >> >> >> On May 18, 2017 8:16:39 PM kk Mailinglist wrote: >> >> > When a phone have a running call and a second call comes in i get no >> > busy and i have redirect to voicemail by busy and the new incoming call >> > will not redirect to VM. >> > >> > I am aware that my phone can have multiple Lines, but how i can signal >> > that i am busy ? >> > >> > 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 >> > > > > -- > > *Brian West* > brian at freeswitch.org > > *Twitter: @FreeSWITCH , @briankwest* > > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > Book a phone call (CST) > > Allison prompts for FreeSWITCH: > > *https://www.gofundme.com/allison-prompts-for-freeswitch* > > > Got Bugs? Report them here ! | Reddit: > /r/freeswitch > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *Skype:*briankwest > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170518/7bb72e97/attachment.html From italo at freeswitch.org Fri May 19 04:04:12 2017 From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=) Date: Fri, 19 May 2017 00:04:12 +0000 Subject: [Freeswitch-users] Callcenter modul error:: register subclass In-Reply-To: References: <13C9308F-F688-4A24-8175-76E761CAB9AB@jerris.com> Message-ID: That's a bug, file a jira and I'll fix it. There's one spot in the code where we don't unregister the bind before unloading. Em qui, 18 de mai de 2017 ?s 14:13, Gregor Nanger escreveu: > Well, fs restart solved issue. > > Will keep an eye if problem repeats. > > On Thu, May 18, 2017, 18:18 Michael Jerris wrote: > >> Thats strange, that means some other module already has that loaded? did >> you do something strange like copy the module and try to load it again? >> >> On May 18, 2017, at 3:51 AM, Gregor Nanger wrote: >> >> ?Can anyone help with error that I get when try to load callcenter module: >> >> 2017-05-18 09:48:49.828297 [ERR] mod_callcenter.c:3810 Couldn't register >> subclass callcenter::info! >> 2017-05-18 09:48:49.828297 [CRIT] switch_loadable_module.c:1522 Error >> Loading module /usr/lib/freeswitch/mod/mod_callcenter.so >> **Module load routine returned an error**? >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170519/72b7e74d/attachment.html From gregor at infomedia.si Fri May 19 09:09:40 2017 From: gregor at infomedia.si (Gregor Nanger) Date: Fri, 19 May 2017 07:09:40 +0200 Subject: [Freeswitch-users] Callcenter modul error:: register subclass In-Reply-To: References: <13C9308F-F688-4A24-8175-76E761CAB9AB@jerris.com> Message-ID: Just to let you know that this happens with FusionPBX. Should I open bug anyway? 2017-05-19 2:04 GMT+02:00 ?talo Rossi : > That's a bug, file a jira and I'll fix it. There's one spot in the code > where we don't unregister the bind before unloading. > > Em qui, 18 de mai de 2017 ?s 14:13, Gregor Nanger > escreveu: > >> Well, fs restart solved issue. >> >> Will keep an eye if problem repeats. >> >> On Thu, May 18, 2017, 18:18 Michael Jerris wrote: >> >>> Thats strange, that means some other module already has that loaded? did >>> you do something strange like copy the module and try to load it again? >>> >>> On May 18, 2017, at 3:51 AM, Gregor Nanger wrote: >>> >>> ?Can anyone help with error that I get when try to load callcenter >>> module: >>> >>> 2017-05-18 09:48:49.828297 [ERR] mod_callcenter.c:3810 Couldn't register >>> subclass callcenter::info! >>> 2017-05-18 09:48:49.828297 [CRIT] switch_loadable_module.c:1522 Error >>> Loading module /usr/lib/freeswitch/mod/mod_callcenter.so >>> **Module load routine returned an error**? >>> >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170519/4a42a086/attachment-0001.html From bipin at xbipin.com Fri May 19 10:41:32 2017 From: bipin at xbipin.com (Bipin Patel) Date: Fri, 19 May 2017 10:41:32 +0400 Subject: [Freeswitch-users] Busy In-Reply-To: References: <6075eed2-9e19-080a-87a3-f4f98fca9f11@ednt.de> <15c1d475890.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com> Message-ID: <2a4a0ad6-624d-686f-fbee-e5b197a1e343@xbipin.com> hi, if its about fusionpbx then u got the wrong mailing list, if not then like Brian said, use mod_limit set the below when some1 places a call this will set a counter for both caller and callee then before the bridge check counter and redirect them to ChangeRingBack extension where u can changeringback or in ur case set to busy Regards, Bipin ------------------------------------------------------------------------ -------- Original Message -------- Subject: Re: [Freeswitch-users] Busy From: David Villasmil To: FreeSWITCH Users Help Date: 5/19/2017, 3:02:24 AM > This guy is talking about fusionpbx... > On Thu, May 18, 2017 at 11:19 PM Brian West > wrote: > > mod_limit. > > On Thu, May 18, 2017 at 3:37 PM, Bipin Patel > wrote: > > Hi, > > I had a similar situation, what I did is when some one places > a call I set > a counter for the calling and called party, then when someone > tries to call > any one of them I check the counter and if it's one I send > them busy or > change ringback tone. I'll check my setup tomorrow and let you > know the > actual module I use > > > > > On May 18, 2017 8:16:39 PM kk Mailinglist > > wrote: > > > When a phone have a running call and a second call comes in > i get no > > busy and i have redirect to voicemail by busy and the new > incoming call > > will not redirect to VM. > > > > I am aware that my phone can have multiple Lines, but how i > can signal > > that i am busy ? > > > > 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 > > > > > -- > > */Brian West/* > brian at freeswitch.org > > */Twitter: @FreeSWITCH , @briankwest/* > > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > Book a phone call (CST) > > Allison prompts for FreeSWITCH: > > *https://www.gofundme.com/allison-prompts-for-freeswitch* > > Got Bugs? Report them here ! | > Reddit: /r/freeswitch > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *Skype:*briankwest > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170519/493fc00a/attachment.html From tculjaga at gmail.com Fri May 19 15:10:36 2017 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Fri, 19 May 2017 13:10:36 +0200 Subject: [Freeswitch-users] DTMF In-Reply-To: References: Message-ID: yeah... doggy carrier, you should somehow get other leg SDP and check that against telephone-event string. i'd use execute_on_answer hook to try to get other leg SDP and than would evaluate that to either start_dtmf or not. anyhow, this is theoretical, but if there is no other straight method, i'd try this. T. On 13 May 2017 at 23:27, Brian : wrote: > Hello List, > > We have a PITA carrier that we do 2833 with 99.8% of the time. > Sometimes if the carrier they are sending the call to doesn't support > 2833 when they will send an OK back to us the SDP won't have 2833 in > the RTP Map. Is it possible to detect this in dialog and start dtmf > generate so we will send tones inband for these calls? > > For inbound calls we detect it with something like > > expression="a=rtpmap:(\d+)\stelephone-event/8000"> > on the initial invite. > > Just not sure how to do this for an OK in dialog or if its possible. > > > Thanks! > Brian > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170519/9a9cfceb/attachment-0001.html From tculjaga at gmail.com Fri May 19 15:11:43 2017 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Fri, 19 May 2017 13:11:43 +0200 Subject: [Freeswitch-users] DTMF Message-ID: hello FS uses, On incoming calls, i got a carrier that advertises rfc2833 (telephone-event) while supporting inBand DTMF only. is there a chance FreeSWITCH can ignore telephone-event for incoming calls from a specific gateway ( profile ) and switch to inBand ? Regards, T. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170519/92df789a/attachment.html From tculjaga at gmail.com Fri May 19 15:41:06 2017 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Fri, 19 May 2017 13:41:06 +0200 Subject: [Freeswitch-users] DTMF In-Reply-To: References: Message-ID: well, this solved my issue :=) not the best practice mode but for now, i can accept InBand DTMF On 19 May 2017 at 13:11, Tihomir Culjaga wrote: > hello FS uses, > > > On incoming calls, i got a carrier that advertises rfc2833 > (telephone-event) while supporting inBand DTMF only. > > is there a chance FreeSWITCH can ignore telephone-event for incoming calls > from a specific gateway ( profile ) and switch to inBand ? > > > Regards, > T. > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170519/454101f9/attachment.html From tculjaga at gmail.com Fri May 19 15:53:55 2017 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Fri, 19 May 2017 13:53:55 +0200 Subject: [Freeswitch-users] DTMF In-Reply-To: References: Message-ID: and for you my friend its like this: On 19 May 2017 at 13:41, Tihomir Culjaga wrote: > well, this solved my issue :=) > > > break="never"> > > > > > > > > > not the best practice mode but for now, i can accept InBand DTMF > > > break="never"> > > > > > > > > > On 19 May 2017 at 13:11, Tihomir Culjaga wrote: > >> hello FS uses, >> >> >> On incoming calls, i got a carrier that advertises rfc2833 >> (telephone-event) while supporting inBand DTMF only. >> >> is there a chance FreeSWITCH can ignore telephone-event for incoming >> calls from a specific gateway ( profile ) and switch to inBand ? >> >> >> Regards, >> T. >> >> >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170519/a0fbbc22/attachment.html From lesley.pervis at gmail.com Fri May 19 16:04:33 2017 From: lesley.pervis at gmail.com (Lesley Pervis) Date: Fri, 19 May 2017 06:04:33 -0600 Subject: [Freeswitch-users] Paging In-Reply-To: <8AE4D7F5-1AE6-44F8-B85D-13C4FADC22CA@magicmail.mooo.com> References: <8AE4D7F5-1AE6-44F8-B85D-13C4FADC22CA@magicmail.mooo.com> Message-ID: I've done this in Lua with outbound conference calls, but it was pretty complicated and you have to have endpoints that will auto-answer. https://freeswitch.org/confluence/display/FREESWITCH/Outbound+Conference+Calls On Wed, May 17, 2017 at 10:02 AM, Rick Jarvis wrote: > Looking to set up paging (not multicast). What?s the best way of achieving > this? Specifically, I want to have the receiving handset(s) answer muted > for privacy reasons, so it?s literally like a PA system rather than just > auto answer?? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170519/2c9f8d91/attachment.html From bipin at xbipin.com Fri May 19 18:36:34 2017 From: bipin at xbipin.com (Bipin Patel) Date: Fri, 19 May 2017 18:36:34 +0400 Subject: [Freeswitch-users] return to shell only when call from originate is complete? Message-ID: hi, i was wondering if its possible to to release handle from an originate command only once the call is completed, i mean i send a originate command from shell using fs_cli but only return back to prompt once the call is completed rather than as soon as a bridge is created or a UUID generated -- Regards, Bipin ------------------------------------------------------------------------ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170519/11b1b3d4/attachment-0001.html From mike at jerris.com Fri May 19 19:34:35 2017 From: mike at jerris.com (Michael Jerris) Date: Fri, 19 May 2017 11:34:35 -0400 Subject: [Freeswitch-users] return to shell only when call from originate is complete? In-Reply-To: References: Message-ID: originate returns when we get call progress, or in the case of ignore_early_media, on answer or failure. > On May 19, 2017, at 10:36 AM, Bipin Patel wrote: > > hi, > > i was wondering if its possible to to release handle from an originate command only once the call is completed, i mean i send a originate command from shell using fs_cli but only return back to prompt once the call is completed rather than as soon as a bridge is created or a UUID generated > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170519/bf98d4c0/attachment.html From bipin at xbipin.com Fri May 19 19:43:19 2017 From: bipin at xbipin.com (Bipin Patel) Date: Fri, 19 May 2017 19:43:19 +0400 Subject: [Freeswitch-users] return to shell only when call from originate is complete? In-Reply-To: References: Message-ID: <69660425-2e51-1943-4fba-fc1ff5ea7094@xbipin.com> hi, does it happen right after answer? is there any way to lock it so it returns only after failure or after hangup? yes im using ignore_early_media Regards, Bipin ------------------------------------------------------------------------ -------- Original Message -------- Subject: Re: [Freeswitch-users] return to shell only when call from originate is complete? From: Michael Jerris To: FreeSWITCH Users Help Date: 5/19/2017, 7:34:35 PM > originate returns when we get call progress, or in the case of > ignore_early_media, on answer or failure. > >> On May 19, 2017, at 10:36 AM, Bipin Patel > > wrote: >> >> hi, >> >> i was wondering if its possible to to release handle from an >> originate command only once the call is completed, i mean i send a >> originate command from shell using fs_cli but only return back to >> prompt once the call is completed rather than as soon as a bridge is >> created or a UUID generated >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170519/b2f5920b/attachment.html From mike at jerris.com Fri May 19 19:48:22 2017 From: mike at jerris.com (Michael Jerris) Date: Fri, 19 May 2017 11:48:22 -0400 Subject: [Freeswitch-users] return to shell only when call from originate is complete? In-Reply-To: <69660425-2e51-1943-4fba-fc1ff5ea7094@xbipin.com> References: <69660425-2e51-1943-4fba-fc1ff5ea7094@xbipin.com> Message-ID: <4E8BF635-E985-4056-B301-A1EEFF702B09@jerris.com> No, thats not what originate does. > On May 19, 2017, at 11:43 AM, Bipin Patel wrote: > > hi, > > does it happen right after answer? > is there any way to lock it so it returns only after failure or after hangup? yes im using ignore_early_media > > > Regards, > Bipin > > > -------- Original Message -------- > Subject: Re: [Freeswitch-users] return to shell only when call from originate is complete? > From: Michael Jerris > To: FreeSWITCH Users Help > Date: 5/19/2017, 7:34:35 PM >> originate returns when we get call progress, or in the case of ignore_early_media, on answer or failure. >> >>> On May 19, 2017, at 10:36 AM, Bipin Patel > wrote: >>> >>> hi, >>> >>> i was wondering if its possible to to release handle from an originate command only once the call is completed, i mean i send a originate command from shell using fs_cli but only return back to prompt once the call is completed rather than as soon as a bridge is created or a UUID generated >>> >> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170519/638b0078/attachment.html From brian at freeswitch.org Fri May 19 19:59:57 2017 From: brian at freeswitch.org (Brian West) Date: Fri, 19 May 2017 10:59:57 -0500 Subject: [Freeswitch-users] Busy In-Reply-To: References: <6075eed2-9e19-080a-87a3-f4f98fca9f11@ednt.de> <15c1d475890.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com> Message-ID: You can still use that in fusion, I do it! On Thu, May 18, 2017 at 6:02 PM, David Villasmil < david.villasmil.work at gmail.com> wrote: > This guy is talking about fusionpbx... > > On Thu, May 18, 2017 at 11:19 PM Brian West wrote: > >> mod_limit. >> >> On Thu, May 18, 2017 at 3:37 PM, Bipin Patel wrote: >> >>> Hi, >>> >>> I had a similar situation, what I did is when some one places a call I >>> set >>> a counter for the calling and called party, then when someone tries to >>> call >>> any one of them I check the counter and if it's one I send them busy or >>> change ringback tone. I'll check my setup tomorrow and let you know the >>> actual module I use >>> >>> >>> >>> >>> On May 18, 2017 8:16:39 PM kk Mailinglist >>> wrote: >>> >>> > When a phone have a running call and a second call comes in i get no >>> > busy and i have redirect to voicemail by busy and the new incoming call >>> > will not redirect to VM. >>> > >>> > I am aware that my phone can have multiple Lines, but how i can signal >>> > that i am busy ? >>> > >>> > 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 >>> >> >> >> >> -- >> >> *Brian West* >> brian at freeswitch.org >> >> *Twitter: @FreeSWITCH , @briankwest* >> >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> Book a phone call (CST) >> >> Allison prompts for FreeSWITCH: >> >> *https://www.gofundme.com/allison-prompts-for-freeswitch* >> >> >> Got Bugs? Report them here ! | Reddit: >> /r/freeswitch >> >> *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002> >> | *M:*+1918424WEST (9378) >> *Skype:*briankwest >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Book a phone call (CST) Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170519/fa1953aa/attachment-0001.html From bipin at xbipin.com Fri May 19 20:31:16 2017 From: bipin at xbipin.com (Bipin Patel) Date: Fri, 19 May 2017 20:31:16 +0400 Subject: [Freeswitch-users] return to shell only when call from originate is complete? In-Reply-To: <4E8BF635-E985-4056-B301-A1EEFF702B09@jerris.com> References: <69660425-2e51-1943-4fba-fc1ff5ea7094@xbipin.com> <4E8BF635-E985-4056-B301-A1EEFF702B09@jerris.com> Message-ID: <8eb5928b-d77f-eb70-fa0b-84f9d21f7d5b@xbipin.com> hi, any other way this can be done Regards, Bipin ------------------------------------------------------------------------ -------- Original Message -------- Subject: Re: [Freeswitch-users] return to shell only when call from originate is complete? From: Michael Jerris To: Bipin Patel Date: 5/19/2017, 7:48:22 PM > No, thats not what originate does. > >> On May 19, 2017, at 11:43 AM, Bipin Patel > > wrote: >> >> hi, >> >> does it happen right after answer? >> is there any way to lock it so it returns only after failure or after >> hangup? yes im using ignore_early_media >> >> >> Regards, >> Bipin >> >> >> ------------------------------------------------------------------------ >> -------- Original Message -------- >> Subject: Re: [Freeswitch-users] return to shell only when call from >> originate is complete? >> From: Michael Jerris >> To: FreeSWITCH Users Help >> Date: 5/19/2017, 7:34:35 PM >>> originate returns when we get call progress, or in the case of >>> ignore_early_media, on answer or failure. >>> >>>> On May 19, 2017, at 10:36 AM, Bipin Patel >>> > wrote: >>>> >>>> hi, >>>> >>>> i was wondering if its possible to to release handle from an >>>> originate command only once the call is completed, i mean i send a >>>> originate command from shell using fs_cli but only return back to >>>> prompt once the call is completed rather than as soon as a bridge >>>> is created or a UUID generated >>>> >>> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170519/4b3ca34a/attachment.html From italo at freeswitch.org Fri May 19 20:37:01 2017 From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=) Date: Fri, 19 May 2017 16:37:01 +0000 Subject: [Freeswitch-users] Callcenter modul error:: register subclass In-Reply-To: References: <13C9308F-F688-4A24-8175-76E761CAB9AB@jerris.com> Message-ID: Yes Em sex, 19 de mai de 2017 ?s 02:11, Gregor Nanger escreveu: > Just to let you know that this happens with FusionPBX. Should I open bug > anyway? > > 2017-05-19 2:04 GMT+02:00 ?talo Rossi : > >> That's a bug, file a jira and I'll fix it. There's one spot in the code >> where we don't unregister the bind before unloading. >> >> Em qui, 18 de mai de 2017 ?s 14:13, Gregor Nanger >> escreveu: >> >>> Well, fs restart solved issue. >>> >>> Will keep an eye if problem repeats. >>> >>> On Thu, May 18, 2017, 18:18 Michael Jerris wrote: >>> >>>> Thats strange, that means some other module already has that loaded? >>>> did you do something strange like copy the module and try to load it again? >>>> >>>> On May 18, 2017, at 3:51 AM, Gregor Nanger wrote: >>>> >>>> ?Can anyone help with error that I get when try to load callcenter >>>> module: >>>> >>>> 2017-05-18 09:48:49.828297 [ERR] mod_callcenter.c:3810 Couldn't >>>> register subclass callcenter::info! >>>> 2017-05-18 09:48:49.828297 [CRIT] switch_loadable_module.c:1522 Error >>>> Loading module /usr/lib/freeswitch/mod/mod_callcenter.so >>>> **Module load routine returned an error**? >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170519/a87f48b0/attachment-0001.html From mike at jerris.com Fri May 19 20:41:28 2017 From: mike at jerris.com (Michael Jerris) Date: Fri, 19 May 2017 12:41:28 -0400 Subject: [Freeswitch-users] return to shell only when call from originate is complete? In-Reply-To: <8eb5928b-d77f-eb70-fa0b-84f9d21f7d5b@xbipin.com> References: <69660425-2e51-1943-4fba-fc1ff5ea7094@xbipin.com> <4E8BF635-E985-4056-B301-A1EEFF702B09@jerris.com> <8eb5928b-d77f-eb70-fa0b-84f9d21f7d5b@xbipin.com> Message-ID: Not specifically what you asked, thats not how it works. What exactly are you trying to accomplish? > On May 19, 2017, at 12:31 PM, Bipin Patel wrote: > > hi, > > any other way this can be done > > > Regards, > Bipin > > > -------- Original Message -------- > Subject: Re: [Freeswitch-users] return to shell only when call from originate is complete? > From: Michael Jerris > To: Bipin Patel > Date: 5/19/2017, 7:48:22 PM >> No, thats not what originate does. >> >>> On May 19, 2017, at 11:43 AM, Bipin Patel > wrote: >>> >>> hi, >>> >>> does it happen right after answer? >>> is there any way to lock it so it returns only after failure or after hangup? yes im using ignore_early_media >>> >>> >>> Regards, >>> Bipin >>> >>> >>> -------- Original Message -------- >>> Subject: Re: [Freeswitch-users] return to shell only when call from originate is complete? >>> From: Michael Jerris >>> To: FreeSWITCH Users Help >>> Date: 5/19/2017, 7:34:35 PM >>>> originate returns when we get call progress, or in the case of ignore_early_media, on answer or failure. >>>> >>>>> On May 19, 2017, at 10:36 AM, Bipin Patel > wrote: >>>>> >>>>> hi, >>>>> >>>>> i was wondering if its possible to to release handle from an originate command only once the call is completed, i mean i send a originate command from shell using fs_cli but only return back to prompt once the call is completed rather than as soon as a bridge is created or a UUID generated >>>>> >>>> >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170519/1f7d3a27/attachment.html From krice at freeswitch.org Fri May 19 20:07:49 2017 From: krice at freeswitch.org (Ken Rice) Date: Fri, 19 May 2017 15:07:49 -0500 Subject: [Freeswitch-users] FW: We've been Busy! In-Reply-To: <04ad01d2d0da$ba3ddcb0$2eb99610$@freeswitch.org> References: <04ad01d2d0da$ba3ddcb0$2eb99610$@freeswitch.org> Message-ID: <04b401d2d0db$97827450$c6875cf0$@freeswitch.org> This is both a test message and a user notification. We've been busy and are going to be all weekend. So today I've done a pile of upgrades to the list server here. And tomorrow we'll have a team lead by Mr West attacking the data center! So don't be surprised if you see services offline a bit tomorrow. This will include everything at the FreeSWITC.org datacenter as we replace some critical infrastructure. What does this mean for you, if you need to do an install from our package repos or do a git clone etc, do it before about 0900 US Eastern in the AM or plan on doing it much later in the day. This will be an extended maintenance window. If you want to help us avoid these outages in the future please visit https://freeswitch.org/ and use the donate link in the menu. Contributions from the FreeSWITCH userbase is what allows us to keep the hardware up and running and where possible deploy redundant hardware so we can fail over and not miss a beat. Thanks! Ken -------------- next part -------------- An HTML attachment was scrubbed... URL: From Agusti.Ubalde at enghouse.com Fri May 19 07:04:23 2017 From: Agusti.Ubalde at enghouse.com (Agusti Ubalde) Date: Fri, 19 May 2017 07:04:23 +0000 Subject: [Freeswitch-users] Verto calls In-Reply-To: References: Message-ID: Hi Giovanni, This is the link to download console logs. https://pastebin.freeswitch.org/view/8f7cfc0c Best regards, PRESENCE TECHNOLOGY An ENGHOUSE INTERACTIVE Company Agustí Ubalde Bellot Chief Developer C/ Comte Urgell 240 3º-A Barcelona 08036 aubalde at presenceco.com Ph: +34 93 10 10 322 Enghouse is listed on the Toronto Stock Exchange, stock symbol: ENGH From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Giovanni Maruzzelli Sent: jueves, 18 de mayo de 2017 18:22 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Verto calls from FreeSWITCH command line: fsctl loglevel 7 console loglevel 7 then repeat the call, and post the *entire*, since beginning, not edited console output in https://pastebin.freeswitch.org/ (***not here in mail***), then write here again and tell us the pastebin address On 18 May 2017 at 09:15, Agustí Ubalde Bellot > wrote: Hi all, I am trying to call from Verto extension to another Verto extension. Both are successfully registered (Verto status show the successfully register) but the call between is not established. The call remains in ring state. This is the last dialplan function executed (calling 1000 to 1001): EXECUTE verto.rtc/1001 bridge() Regards, Tineli _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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 krice at freeswitch.org Fri May 19 20:14:13 2017 From: krice at freeswitch.org (Ken Rice) Date: Fri, 19 May 2017 15:14:13 -0500 Subject: [Freeswitch-users] and another test.. Message-ID: <04c401d2d0dc$7c614e20$7523ea60$@freeswitch.org> Did it blend? -------------- next part -------------- An HTML attachment was scrubbed... URL: From krice at freeswitch.org Fri May 19 20:35:50 2017 From: krice at freeswitch.org (Ken Rice) Date: Fri, 19 May 2017 15:35:50 -0500 Subject: [Freeswitch-users] We've been Busy! In-Reply-To: <04ad01d2d0da$ba3ddcb0$2eb99610$@freeswitch.org> References: <04ad01d2d0da$ba3ddcb0$2eb99610$@freeswitch.org> Message-ID: <060c01d2d0df$81512970$83f37c50$@freeswitch.org> This is both a test message and a user notification. (if you saw this twice its because I sent it again to make sure everyone sees it!) We've been busy and are going to be all weekend. So today I've done a pile of upgrades to the list server here. And tomorrow we'll have a team lead by Mr West attacking the data center! So don't be surprised if you see services offline a bit tomorrow. This will include everything at the FreeSWITC.org datacenter as we replace some critical infrastructure. What does this mean for you, if you need to do an install from our package repos or do a git clone etc, do it before about 0900 US Eastern in the AM or plan on doing it much later in the day. This will be an extended maintenance window. If you want to help us avoid these outages in the future please visit https://freeswitch.org/ and use the donate link in the menu. Contributions from the FreeSWITCH userbase is what allows us to keep the hardware up and running and where possible deploy redundant hardware so we can fail over and not miss a beat. Thanks! Ken -------------- next part -------------- An HTML attachment was scrubbed... URL: From bipin at xbipin.com Fri May 19 20:37:18 2017 From: bipin at xbipin.com (Bipin Patel) Date: Sat, 20 May 2017 00:37:18 +0400 Subject: [Freeswitch-users] return to shell only when call from originate is complete? In-Reply-To: References: <69660425-2e51-1943-4fba-fc1ff5ea7094@xbipin.com> <4E8BF635-E985-4056-B301-A1EEFF702B09@jerris.com> <8eb5928b-d77f-eb70-fa0b-84f9d21f7d5b@xbipin.com> Message-ID: <15c226dc430.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com> Well I have a script that calls numbers over a fxo gateway and plays a ivr. The fxo gateway has one channel available at times out of 4, the originate keeps calling them non stop without even the first call finished, I want it to pause till one call is over and then move over to the next so the others don't fail due to no channels available. On May 19, 2017 8:41:30 PM Michael Jerris wrote: > Not specifically what you asked, thats not how it works. What exactly are > you trying to accomplish… > >> On May 19, 2017, at 12:31 PM, Bipin Patel wrote: >> >> hi, >> >> any other way this can be done >> >> >> Regards, >> Bipin >> >> >> -------- Original Message -------- >> Subject: Re: [Freeswitch-users] return to shell only when call from >> originate is complete? >> From: Michael Jerris >> To: Bipin Patel >> Date: 5/19/2017, 7:48:22 PM >>> No, thats not what originate does. >>> >>>> On May 19, 2017, at 11:43 AM, Bipin Patel >>> > wrote: >>>> >>>> hi, >>>> >>>> does it happen right after answer? >>>> is there any way to lock it so it returns only after failure or after >>>> hangup? yes im using ignore_early_media >>>> >>>> >>>> Regards, >>>> Bipin >>>> >>>> >>>> -------- Original Message -------- >>>> Subject: Re: [Freeswitch-users] return to shell only when call from >>>> originate is complete? >>>> From: Michael Jerris >>>> To: FreeSWITCH Users Help >>>> >>>> Date: 5/19/2017, 7:34:35 PM >>>>> originate returns when we get call progress, or in the case of >>>>> ignore_early_media, on answer or failure. >>>>> >>>>>> On May 19, 2017, at 10:36 AM, Bipin Patel >>>>> > wrote: >>>>>> >>>>>> hi, >>>>>> >>>>>> i was wondering if its possible to to release handle from an originate >>>>>> command only once the call is completed, i mean i send a originate command >>>>>> from shell using fs_cli but only return back to prompt once the call is >>>>>> completed rather than as soon as a bridge is created or a UUID generated >>>>>> >>>>> >>> >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From aqsyounas at gmail.com Fri May 19 21:25:36 2017 From: aqsyounas at gmail.com (Aqs Younas) Date: Sat, 20 May 2017 02:25:36 +0500 Subject: [Freeswitch-users] return to shell only when call from originate is complete? In-Reply-To: <15c226dc430.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com> References: <69660425-2e51-1943-4fba-fc1ff5ea7094@xbipin.com> <4E8BF635-E985-4056-B301-A1EEFF702B09@jerris.com> <8eb5928b-d77f-eb70-fa0b-84f9d21f7d5b@xbipin.com> <15c226dc430.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com> Message-ID: You originate a call and subscribe to events and based on those events you originate another call. On 20 May 2017 1:39 am, "Bipin Patel" wrote: > Well I have a script that calls numbers over a fxo gateway and plays a > ivr. The fxo gateway has one channel available at times out of 4, the > originate keeps calling them non stop without even the first call finished, > I want it to pause till one call is over and then move over to the next so > the others don't fail due to no channels available. > > On May 19, 2017 8:41:30 PM Michael Jerris wrote: > >> Not specifically what you asked, thats not how it works. What exactly >> are you trying to accomplish… >> >> On May 19, 2017, at 12:31 PM, Bipin Patel wrote: >> >> hi, >> >> any other way this can be done >> >> >> Regards, >> Bipin >> >> >> ------------------------------ >> -------- Original Message -------- >> Subject: Re: [Freeswitch-users] return to shell only when call from >> originate is complete? >> From: Michael Jerris >> To: Bipin Patel >> Date: 5/19/2017, 7:48:22 PM >> >> No, thats not what originate does. >> >> On May 19, 2017, at 11:43 AM, Bipin Patel wrote: >> >> hi, >> >> does it happen right after answer? >> is there any way to lock it so it returns only after failure or after >> hangup? yes im using ignore_early_media >> >> >> Regards, >> Bipin >> >> >> ------------------------------ >> -------- Original Message -------- >> Subject: Re: [Freeswitch-users] return to shell only when call from >> originate is complete? >> From: Michael Jerris >> To: FreeSWITCH Users Help >> >> Date: 5/19/2017, 7:34:35 PM >> >> originate returns when we get call progress, or in the case of >> ignore_early_media, on answer or failure. >> >> On May 19, 2017, at 10:36 AM, Bipin Patel wrote: >> >> hi, >> >> i was wondering if its possible to to release handle from an originate >> command only once the call is completed, i mean i send a originate command >> from shell using fs_cli but only return back to prompt once the call is >> completed rather than as soon as a bridge is created or a UUID generated >> >> >> >> >> >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 wsimon at stratusvideo.com Fri May 19 21:29:05 2017 From: wsimon at stratusvideo.com (William Simon) Date: Fri, 19 May 2017 21:29:05 +0000 Subject: [Freeswitch-users] return to shell only when call from originate is complete? In-Reply-To: <8eb5928b-d77f-eb70-fa0b-84f9d21f7d5b@xbipin.com> References: <69660425-2e51-1943-4fba-fc1ff5ea7094@xbipin.com> <4E8BF635-E985-4056-B301-A1EEFF702B09@jerris.com> <8eb5928b-d77f-eb70-fa0b-84f9d21f7d5b@xbipin.com> Message-ID: <4112C9C4-5BAA-408F-9731-8B11F3CCDAD8@stratusvideo.com> If you are using fs_cli then a simple shell script can monitor the call and return when the call completes: #!/bin/bash RESULT=$( fs_cli -x "originate leg1 leg2" ) if [[ $RESULT =~ "OK" ]]; then UUID=$( echo "$RESULT" | cut -f 2 -d ' ' ) while fs_cli -x "show channels" | grep -q $UUID ; do sleep 5; done echo "Call completed." fi On May 19, 2017, at 12:31 PM, Bipin Patel > wrote: hi, any other way this can be done Regards, Bipin ________________________________ -------- Original Message -------- Subject: Re: [Freeswitch-users] return to shell only when call from originate is complete? From: Michael Jerris To: Bipin Patel Date: 5/19/2017, 7:48:22 PM No, thats not what originate does. On May 19, 2017, at 11:43 AM, Bipin Patel > wrote: hi, does it happen right after answer? is there any way to lock it so it returns only after failure or after hangup? yes im using ignore_early_media Regards, Bipin ________________________________ -------- Original Message -------- Subject: Re: [Freeswitch-users] return to shell only when call from originate is complete? From: Michael Jerris To: FreeSWITCH Users Help Date: 5/19/2017, 7:34:35 PM originate returns when we get call progress, or in the case of ignore_early_media, on answer or failure. On May 19, 2017, at 10:36 AM, Bipin Patel > wrote: hi, i was wondering if its possible to to release handle from an originate command only once the call is completed, i mean i send a originate command from shell using fs_cli but only return back to prompt once the call is completed rather than as soon as a bridge is created or a UUID generated “The information transmitted is intended only for the person or entity to which it is addressed and may contain proprietary, business-confidential and/or privileged material. If you are not the intended recipient of this message you are hereby notified that any use, review, retransmission, dissemination, distribution, reproduction or any action taken in reliance upon this message is prohibited. If you received this in error, please contact the sender and delete the material from any computer.” _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org “The information transmitted is intended only for the person or entity to which it is addressed and may contain proprietary, business-confidential and/or privileged material. If you are not the intended recipient of this message you are hereby notified that any use, review, retransmission, dissemination, distribution, reproduction or any action taken in reliance upon this message is prohibited. If you received this in error, please contact the sender and delete the material from any computer.” -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Fri May 19 22:28:47 2017 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Sat, 20 May 2017 00:28:47 +0200 Subject: [Freeswitch-users] We've been Busy! In-Reply-To: <060c01d2d0df$81512970$83f37c50$@freeswitch.org> References: <04ad01d2d0da$ba3ddcb0$2eb99610$@freeswitch.org> <060c01d2d0df$81512970$83f37c50$@freeswitch.org> Message-ID: Donate, already !!! On 19 May 2017 at 22:35, Ken Rice wrote: > This is both a test message and a user notification. (if you saw this > twice its because I sent it again to make sure everyone sees it!) > > > > We’ve been busy and are going to be all weekend. > > > > So today I’ve done a pile of upgrades to the list server here. And > tomorrow we’ll have a team lead by Mr West attacking the data center! > > > > So don’t be surprised if you see services offline a bit tomorrow. This > will include everything at the FreeSWITC.org datacenter as we replace some > critical infrastructure. > > > > What does this mean for you, if you need to do an install from our package > repos or do a git clone etc, do it before about 0900 US Eastern in the AM > or plan on doing it much later in the day. > > > > This will be an extended maintenance window. > > > > If you want to help us avoid these outages in the future please visit > https://freeswitch.org/ and use the donate link in the menu. > Contributions from the FreeSWITCH userbase is what allows us to keep the > hardware up and running and where possible deploy redundant hardware so we > can fail over and not miss a beat. > > > > Thanks! > > Ken > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 servtelar at gmail.com Fri May 19 22:38:17 2017 From: servtelar at gmail.com (GM phy) Date: Fri, 19 May 2017 19:38:17 -0300 Subject: [Freeswitch-users] We've been Busy! In-Reply-To: References: <04ad01d2d0da$ba3ddcb0$2eb99610$@freeswitch.org> <060c01d2d0df$81512970$83f37c50$@freeswitch.org> Message-ID: <145BD7B6-CE65-4DA3-955E-D7ED436D2D88@gmail.com> Done too > On May 19, 2017, at 7:28 PM, Giovanni Maruzzelli wrote: > > Donate, already !!! > > > > On 19 May 2017 at 22:35, Ken Rice > wrote: > This is both a test message and a user notification. (if you saw this twice its because I sent it again to make sure everyone sees it!) > > > > We’ve been busy and are going to be all weekend. > > > > So today I’ve done a pile of upgrades to the list server here. And tomorrow we’ll have a team lead by Mr West attacking the data center! > > > > So don’t be surprised if you see services offline a bit tomorrow. This will include everything at the FreeSWITC.org datacenter as we replace some critical infrastructure. > > > > What does this mean for you, if you need to do an install from our package repos or do a git clone etc, do it before about 0900 US Eastern in the AM or plan on doing it much later in the day. > > > > This will be an extended maintenance window. > > > > If you want to help us avoid these outages in the future please visit https://freeswitch.org/ and use the donate link in the menu. Contributions from the FreeSWITCH userbase is what allows us to keep the hardware up and running and where possible deploy redundant hardware so we can fail over and not miss a beat. > > > > Thanks! > > Ken > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 grcamauer at gmail.com Fri May 19 22:44:13 2017 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Fri, 19 May 2017 19:44:13 -0300 Subject: [Freeswitch-users] We've been Busy! In-Reply-To: <145BD7B6-CE65-4DA3-955E-D7ED436D2D88@gmail.com> References: <04ad01d2d0da$ba3ddcb0$2eb99610$@freeswitch.org> <060c01d2d0df$81512970$83f37c50$@freeswitch.org> <145BD7B6-CE65-4DA3-955E-D7ED436D2D88@gmail.com> Message-ID: Donated! On Fri, May 19, 2017 at 7:38 PM, GM phy wrote: > Done too > > > On May 19, 2017, at 7:28 PM, Giovanni Maruzzelli > wrote: > > Donate, already !!! > > > > On 19 May 2017 at 22:35, Ken Rice wrote: > >> This is both a test message and a user notification. (if you saw this >> twice its because I sent it again to make sure everyone sees it!) >> >> >> >> We’ve been busy and are going to be all weekend. >> >> >> >> So today I’ve done a pile of upgrades to the list server here. And >> tomorrow we’ll have a team lead by Mr West attacking the data center! >> >> >> >> So don’t be surprised if you see services offline a bit tomorrow. This >> will include everything at the FreeSWITC.org >> datacenter as we replace some critical infrastructure. >> >> >> >> What does this mean for you, if you need to do an install from our >> package repos or do a git clone etc, do it before about 0900 US Eastern in >> the AM or plan on doing it much later in the day. >> >> >> >> This will be an extended maintenance window. >> >> >> >> If you want to help us avoid these outages in the future please visit >> https://freeswitch.org/ and use the donate link in the menu. >> Contributions from the FreeSWITCH userbase is what allows us to keep the >> hardware up and running and where possible deploy redundant hardware so we >> can fail over and not miss a beat. >> >> >> >> Thanks! >> >> Ken >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 > -- Guillermo Ruiz Camauer -------------- next part -------------- An HTML attachment was scrubbed... URL: From bipin at xbipin.com Sat May 20 05:37:57 2017 From: bipin at xbipin.com (Bipin Patel) Date: Sat, 20 May 2017 09:37:57 +0400 Subject: [Freeswitch-users] return to shell only when call from originate is complete? In-Reply-To: <4112C9C4-5BAA-408F-9731-8B11F3CCDAD8@stratusvideo.com> References: <69660425-2e51-1943-4fba-fc1ff5ea7094@xbipin.com> <4E8BF635-E985-4056-B301-A1EEFF702B09@jerris.com> <8eb5928b-d77f-eb70-fa0b-84f9d21f7d5b@xbipin.com> <4112C9C4-5BAA-408F-9731-8B11F3CCDAD8@stratusvideo.com> Message-ID: <6f17d853-9ee6-15ba-bb93-a48862329bc0@xbipin.com> hi, this seems like a decent way, now just need to convert this to python, thanks anyways Regards, Bipin ------------------------------------------------------------------------ -------- Original Message -------- Subject: Re: [Freeswitch-users] return to shell only when call from originate is complete? From: William Simon To: FreeSWITCH Users Help Date: 5/20/2017, 1:29:05 AM > If you are using fs_cli then a simple shell script can monitor the > call and return when the call completes: > > #!/bin/bash > > RESULT=$( fs_cli -x "originate leg1 leg2" ) > > if [[ $RESULT =~ "OK" ]]; then > UUID=$( echo "$RESULT" | cut -f 2 -d ' ' ) > while fs_cli -x "show channels" | grep -q $UUID ; > do > sleep 5; > done > echo "Call completed." > fi > > > > >> On May 19, 2017, at 12:31 PM, Bipin Patel > > wrote: >> >> hi, >> >> any other way this can be done >> >> >> Regards, >> Bipin >> >> >> ------------------------------------------------------------------------ >> -------- Original Message -------- >> Subject: Re: [Freeswitch-users] return to shell only when call from >> originate is complete? >> From: Michael Jerris >> To: Bipin Patel >> Date: 5/19/2017, 7:48:22 PM >>> No, thats not what originate does. >>> >>>> On May 19, 2017, at 11:43 AM, Bipin Patel >>> > wrote: >>>> >>>> hi, >>>> >>>> does it happen right after answer? >>>> is there any way to lock it so it returns only after failure or >>>> after hangup? yes im using ignore_early_media >>>> >>>> >>>> Regards, >>>> Bipin >>>> >>>> >>>> ------------------------------------------------------------------------ >>>> -------- Original Message -------- >>>> Subject: Re: [Freeswitch-users] return to shell only when call from >>>> originate is complete? >>>> From: Michael Jerris >>>> To: FreeSWITCH Users Help >>>> Date: 5/19/2017, 7:34:35 PM >>>>> originate returns when we get call progress, or in the case of >>>>> ignore_early_media, on answer or failure. >>>>> >>>>>> On May 19, 2017, at 10:36 AM, Bipin Patel >>>>> > wrote: >>>>>> >>>>>> hi, >>>>>> >>>>>> i was wondering if its possible to to release handle from an >>>>>> originate command only once the call is completed, i mean i send >>>>>> a originate command from shell using fs_cli but only return back >>>>>> to prompt once the call is completed rather than as soon as a >>>>>> bridge is created or a UUID generated >>>>>> >>>>> >>> >> >> >> >> “The information transmitted is intended only for the person or >> entity to which it is addressed and may contain proprietary, >> business-confidential and/or privileged material. If you are not the >> intended recipient of this message you are hereby notified that any >> use, review, retransmission, dissemination, distribution, >> reproduction or any action taken in reliance upon this message is >> prohibited. If you received this in error, please contact the sender >> and delete the material from any computer.” >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > “The information transmitted is intended only for the person or entity > to which it is addressed and may contain proprietary, > business-confidential and/or privileged material. If you are not the > intended recipient of this message you are hereby notified that any > use, review, retransmission, dissemination, distribution, reproduction > or any action taken in reliance upon this message is prohibited. If > you received this in error, please contact the sender and delete the > material from any computer.” > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 fernando at softov.com.br Sun May 21 00:02:24 2017 From: fernando at softov.com.br (Luiz Fernando Softov) Date: Sat, 20 May 2017 20:02:24 -0400 Subject: [Freeswitch-users] Freeswitch PRI support In-Reply-To: References: <27a9d063.ad3b.15bd85e9762.Coremail.xxxman2008@126.com> <6fc1c046-45f9-cdbe-2896-a077dbcc5dad@xbipin.com> Message-ID: Hello, here I use GsmOpen with some modifications. This work very well in my clients, some of then using with 4, 16, 27, 32 even 64 modens. There are different user cases. Some of then are using in ISPs making a Point to Point in rural areas in Brazil. There are others clients, using in CallCenter, to call clients. Here there is 4 main operators, VIVO, OI, CLARO, TIM, and client can make portability. So, the system has a database of all portability. And recognizes when a number is from VIVO, but got portability to CLARO, etc. The system calls using one of CLARO chips, according with the rules of the routes. There is another user case, when a company want to give the clients more numbers to receive calls, so client doesn't make expensive calls. And, there is a lot of user cases more. This is just the main. The main problem is to manage dongles, because sometimes, with a bad power supply, the modem get down and on (RESET). So the TTY port can change. I tried to use GSMOpen to recognize the tty by IMSI, without success. So I made my own daemon, that load all modems and tty, and using gsmopen reload + XML Curl, I can made changes in FreeSwitch on the fly. Here is expensive the cost of GSM Gateways. Some of then are R$ 4000 with 8 ports. Or R$ 16000 with 32 ports. Using GSMOpen, I can make a system with 32 modems with only R$ 500. I have at least 50 companies using this solution. And ~ 200 companies using my system. Search in the list for my name, there is a lot of e-mails talking about it. If you wan't I can give you my patch, there is a lot of corrections. The main of then are about Thread getting deadlock. 2017-05-11 11:31 GMT-04:00 Deepika Yadav : > This is for the usecase of community heath worker in rural areas of India > who only have access to feature phones i.e. no Internet, no data plan. > > Regards, > Deepika > > On Thu, May 11, 2017 at 8:05 PM, Colton Conor > wrote: > >> What is the value? Mobile voice plans are expensive (realtive to sip >> trunks). Also, call quality is not even G711 in most cases. >> >> I could see the use case is you wanted a mobile backup with GSM for voice >> in the case your hardline internet were down, but even then I would think >> it would be cheaper and more benefical to get a mobile data backup plan, >> and continue to use the SIP trunks over the internet. >> >> I honestly had now clue these large GSM banks existed. >> >> Anyone used them in North America? >> >> On Wed, May 10, 2017 at 10:06 AM, David Villasmil < >> david.villasmil.work at gmail.com> wrote: >> >>> I think it's illegal everywhere... but value? there's lots and lots of >>> values, it only depends on where you are doing it... :) >>> ᐧ >>> >>> Regards, >>> >>> David Villasmil >>> email: david.villasmil.work at gmail.com >>> phone: +34669448337 <+34%20669%2044%2083%2037> >>> >>> On Wed, May 10, 2017 at 4:12 PM, Raúl Alexis Betancor Santana < >>> rbetancor at gmail.com> wrote: >>> >>>> The only real advantage over a regular SIP trunk from any provider, >>>> it's that you could select a plan than MAYBE allow you some sort of >>>> flat-rate and not per-minute use plans ... but it doesn't worth the effort >>>> and money you have to spend to get that to generate revenue ... also most >>>> of celullar carriers withh put some restrictions on the usage of the >>>> flat-rate, as all of them have 'fair use' policies. >>>> At least on Spain, it's ilegal to resell that kind of traffic and I >>>> asume its the same in most of the world. >>>> >>>> 2017-05-10 15:04 GMT+01:00 Colton Conor : >>>> >>>>> So how does this GSM stuff work in the USA? Would I got buy a cell >>>>> phone plan from someone like AT&T or T-Mobile, and then insert the SIM into >>>>> these devices. The device would then covert the audio from GSM to SIP, and >>>>> then I could use it like a SIP trunk? >>>>> >>>>> What is the advantage of doing this over just buying a pure SIP trunk >>>>> from an internet provider? >>>>> >>>>> >>>>> >>>>> On Wed, May 10, 2017 at 4:35 AM, David Villasmil < >>>>> david.villasmil.work at gmail.com> wrote: >>>>> >>>>>> There's another chinese brand that has a 32 port, I've used it, works >>>>>> pretty well, i think it's "GoIP" or something. >>>>>> >>>>>> On Wed, May 10, 2017 at 10:39 AM Bipin Patel >>>>>> wrote: >>>>>> >>>>>>> hi, >>>>>>> >>>>>>> we use matrix boxes but 4 port ones, they work quiet well >>>>>>> >>>>>>> >>>>>>> Regards, >>>>>>> Bipin >>>>>>> >>>>>>> >>>>>>> ------------------------------ >>>>>>> -------- Original Message -------- >>>>>>> Subject: Re: [Freeswitch-users] Freeswitch PRI support >>>>>>> From: Deepika Yadav >>>>>>> To: FreeSWITCH Users Help >>>>>>> >>>>>>> Date: 5/10/2017, 11:02:49 AM >>>>>>> >>>>>>> Hi, >>>>>>> >>>>>>> I searched some gateways online. The ones available at ebay are >>>>>>> cheap, the following link shows a gateway supporting 32 SIMS for $859 : >>>>>>> >>>>>>> http://www.ebay.com/itm/GoIP-32-32-Port-VoIP-GSM-Gateway-wit >>>>>>> h-32-External-Antenna-/291811535942 >>>>>>> >>>>>>> However, when I contacted a company called "Matrix Telecom >>>>>>> Solution", they gave the quote of $4150 for similar number of SIM support. >>>>>>> On asking them to compare the two gateways, they said that the one >>>>>>> available at ebay is a Chinese gateway about which they are apprehensive >>>>>>> for the quality, warranty and working. >>>>>>> >>>>>>> Regards, >>>>>>> Deepika >>>>>>> >>>>>>> >>>>>>> On Fri, May 5, 2017 at 4:58 PM, Raymond wrote: >>>>>>> >>>>>>>> >>>>>>>> HI, >>>>>>>> >>>>>>>> Regarding this. i have asked some question of module "GSMOpen". >>>>>>>> >>>>>>>> >>>>>>>> In fact ,i want to make Freeswitch box as "GSM VOIP Gateway" by >>>>>>>> using "gsmopen". But...seems no one help me . :-(( . If it worked , can >>>>>>>> save more money than hardware gsm voip gateway ( GOIP ). >>>>>>>> >>>>>>>> Raymond >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> At 2017-05-04 17:24:13, "Giovanni Maruzzelli" >>>>>>>> wrote: >>>>>>>> >>>>>>>> For PRI you use Sangoma or Patton. >>>>>>>> >>>>>>>> But, why don't you use an hardware gateway sip<->gsm? >>>>>>>> >>>>>>>> It would save you very big money. >>>>>>>> >>>>>>>> Check on ebay and google, there are many of them, you put SIMs >>>>>>>> inside, and you are good to go. >>>>>>>> >>>>>>>> -giovanni >>>>>>>> >>>>>>>> >>>>>>>> sent from mobile >>>>>>>> cell: +39 347 266 56 18 >>>>>>>> Giovanni Maruzzelli >>>>>>>> OpenTelecom.IT >>>>>>>> >>>>>>>> Il 04 mag 2017 11:13, "Deepika Yadav" ha >>>>>>>> scritto: >>>>>>>> >>>>>>>>> Hi, >>>>>>>>> >>>>>>>>> I am using Freeswitch in an application that initiates conference >>>>>>>>> calls amongst 20 users in cellular network (mobile phones in GSM network). >>>>>>>>> Currently, for the VOIP-GSM gateway, we are using the service from a >>>>>>>>> company called Doorvaani. >>>>>>>>> >>>>>>>>> But, since, the cost of call estimates to be high and we cannot >>>>>>>>> debug the call drops; we are thinking to buy our own PRI card. >>>>>>>>> >>>>>>>>> I am seeking recommendation on following points: >>>>>>>>> >>>>>>>>> 1. Which card should I buy that is most easily configurable >>>>>>>>> with the Freeswitch i.e. company and type. >>>>>>>>> 2. Reference on how should I start to make Freeswitch >>>>>>>>> configure with the PRI card and start sending and receiving calls. For the >>>>>>>>> current gateway service in use, I simply put the authentication credentials >>>>>>>>> for the corresponding VOIP line offered by the company Doorvaani in the >>>>>>>>> external SIP Profile. In case of PRI, what all changes are needed? >>>>>>>>> >>>>>>>>> Regards, >>>>>>>>> Deepika >>>>>>>>> >>>>>>>>> -- >>>>>>>>> Regards >>>>>>>>> Deepika >>>>>>>>> https://www.iiitd.edu.in/~deepikay/ >>>>>>>>> >>>>>>>>> ____________________________________________________________ >>>>>>>>> _____________ >>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>> consulting at freeswitch.org >>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>> >>>>>>>>> Official FreeSWITCH Sites >>>>>>>>> http://www.freeswitch.org >>>>>>>>> http://confluence.freeswitch.org >>>>>>>>> http://www.cluecon.com >>>>>>>>> >>>>>>>>> FreeSWITCH-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 >>>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> Regards >>>>>>> Deepika >>>>>>> https://www.iiitd.edu.in/~deepikay/ >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> 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/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/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 >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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://www.iiitd.edu.in/~deepikay/ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 Sun May 21 04:11:47 2017 From: mike at jerris.com (Michael Jerris) Date: Sun, 21 May 2017 04:11:47 +0000 Subject: [Freeswitch-users] Freeswitch PRI support In-Reply-To: References: <27a9d063.ad3b.15bd85e9762.Coremail.xxxman2008@126.com> <6fc1c046-45f9-cdbe-2896-a077dbcc5dad@xbipin.com> Message-ID: Please share any improvements or patches you have with the whole community by submitting a bug report and a pull request with your patch so it can be properly reviewed. https://freeswitch.org/confluence/display/FREESWITCH/Pull+Requests On Sat, May 20, 2017 at 8:07 PM Luiz Fernando Softov wrote: > Hello, here I use GsmOpen with some modifications. > > This work very well in my clients, some of then using with 4, 16, 27, 32 > even 64 modens. > > There are different user cases. > > Some of then are using in ISPs making a Point to Point in rural areas in > Brazil. > > There are others clients, using in CallCenter, to call clients. > > Here there is 4 main operators, VIVO, OI, CLARO, TIM, and client can make > portability. > > So, the system has a database of all portability. And recognizes when a > number is from VIVO, but got portability to CLARO, etc. > The system calls using one of CLARO chips, according with the rules of the > routes. > > There is another user case, when a company want to give the clients more > numbers to receive calls, so client doesn't make expensive calls. > > And, there is a lot of user cases more. This is just the main. > > > The main problem is to manage dongles, because sometimes, with a bad power > supply, the modem get down and on (RESET). > So the TTY port can change. > > I tried to use GSMOpen to recognize the tty by IMSI, without success. > > So I made my own daemon, that load all modems and tty, and using gsmopen > reload + XML Curl, I can made changes in FreeSwitch on the fly. > > Here is expensive the cost of GSM Gateways. > Some of then are R$ 4000 with 8 ports. > Or R$ 16000 with 32 ports. > > Using GSMOpen, I can make a system with 32 modems with only R$ 500. > > I have at least 50 companies using this solution. > And ~ 200 companies using my system. > > Search in the list for my name, there is a lot of e-mails talking about it. > > If you wan't I can give you my patch, there is a lot of corrections. > The main of then are about Thread getting deadlock. > > > > > > > 2017-05-11 11:31 GMT-04:00 Deepika Yadav : > >> This is for the usecase of community heath worker in rural areas of India >> who only have access to feature phones i.e. no Internet, no data plan. >> >> Regards, >> Deepika >> >> On Thu, May 11, 2017 at 8:05 PM, Colton Conor >> wrote: >> >>> What is the value? Mobile voice plans are expensive (realtive to sip >>> trunks). Also, call quality is not even G711 in most cases. >>> >>> I could see the use case is you wanted a mobile backup with GSM for >>> voice in the case your hardline internet were down, but even then I would >>> think it would be cheaper and more benefical to get a mobile data backup >>> plan, and continue to use the SIP trunks over the internet. >>> >>> I honestly had now clue these large GSM banks existed. >>> >>> Anyone used them in North America? >>> >>> On Wed, May 10, 2017 at 10:06 AM, David Villasmil < >>> david.villasmil.work at gmail.com> wrote: >>> >>>> I think it's illegal everywhere... but value? there's lots and lots of >>>> values, it only depends on where you are doing it... :) >>>> ᐧ >>>> >>>> Regards, >>>> >>>> David Villasmil >>>> email: david.villasmil.work at gmail.com >>>> phone: +34669448337 <+34%20669%2044%2083%2037> >>>> >>>> On Wed, May 10, 2017 at 4:12 PM, Raúl Alexis Betancor Santana < >>>> rbetancor at gmail.com> wrote: >>>> >>>>> The only real advantage over a regular SIP trunk from any provider, >>>>> it's that you could select a plan than MAYBE allow you some sort of >>>>> flat-rate and not per-minute use plans ... but it doesn't worth the effort >>>>> and money you have to spend to get that to generate revenue ... also most >>>>> of celullar carriers withh put some restrictions on the usage of the >>>>> flat-rate, as all of them have 'fair use' policies. >>>>> At least on Spain, it's ilegal to resell that kind of traffic and I >>>>> asume its the same in most of the world. >>>>> >>>>> 2017-05-10 15:04 GMT+01:00 Colton Conor : >>>>> >>>>>> So how does this GSM stuff work in the USA? Would I got buy a cell >>>>>> phone plan from someone like AT&T or T-Mobile, and then insert the SIM into >>>>>> these devices. The device would then covert the audio from GSM to SIP, and >>>>>> then I could use it like a SIP trunk? >>>>>> >>>>>> What is the advantage of doing this over just buying a pure SIP trunk >>>>>> from an internet provider? >>>>>> >>>>>> >>>>>> >>>>>> On Wed, May 10, 2017 at 4:35 AM, David Villasmil < >>>>>> david.villasmil.work at gmail.com> wrote: >>>>>> >>>>>>> There's another chinese brand that has a 32 port, I've used it, >>>>>>> works pretty well, i think it's "GoIP" or something. >>>>>>> >>>>>>> On Wed, May 10, 2017 at 10:39 AM Bipin Patel >>>>>>> wrote: >>>>>>> >>>>>>>> hi, >>>>>>>> >>>>>>>> we use matrix boxes but 4 port ones, they work quiet well >>>>>>>> >>>>>>>> >>>>>>>> Regards, >>>>>>>> Bipin >>>>>>>> >>>>>>>> >>>>>>>> ------------------------------ >>>>>>>> -------- Original Message -------- >>>>>>>> Subject: Re: [Freeswitch-users] Freeswitch PRI support >>>>>>>> From: Deepika Yadav >>>>>>>> To: FreeSWITCH Users Help >>>>>>>> >>>>>>>> Date: 5/10/2017, 11:02:49 AM >>>>>>>> >>>>>>>> Hi, >>>>>>>> >>>>>>>> I searched some gateways online. The ones available at ebay are >>>>>>>> cheap, the following link shows a gateway supporting 32 SIMS for $859 : >>>>>>>> >>>>>>>> >>>>>>>> http://www.ebay.com/itm/GoIP-32-32-Port-VoIP-GSM-Gateway-with-32-External-Antenna-/291811535942 >>>>>>>> >>>>>>>> However, when I contacted a company called "Matrix Telecom >>>>>>>> Solution", they gave the quote of $4150 for similar number of SIM support. >>>>>>>> On asking them to compare the two gateways, they said that the one >>>>>>>> available at ebay is a Chinese gateway about which they are apprehensive >>>>>>>> for the quality, warranty and working. >>>>>>>> >>>>>>>> Regards, >>>>>>>> Deepika >>>>>>>> >>>>>>>> >>>>>>>> On Fri, May 5, 2017 at 4:58 PM, Raymond wrote: >>>>>>>> >>>>>>>>> >>>>>>>>> HI, >>>>>>>>> >>>>>>>>> Regarding this. i have asked some question of module "GSMOpen". >>>>>>>>> >>>>>>>>> >>>>>>>>> In fact ,i want to make Freeswitch box as "GSM VOIP Gateway" by >>>>>>>>> using "gsmopen". But...seems no one help me . :-(( . If it worked , can >>>>>>>>> save more money than hardware gsm voip gateway ( GOIP ). >>>>>>>>> >>>>>>>>> Raymond >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> At 2017-05-04 17:24:13, "Giovanni Maruzzelli" >>>>>>>>> wrote: >>>>>>>>> >>>>>>>>> For PRI you use Sangoma or Patton. >>>>>>>>> >>>>>>>>> But, why don't you use an hardware gateway sip<->gsm? >>>>>>>>> >>>>>>>>> It would save you very big money. >>>>>>>>> >>>>>>>>> Check on ebay and google, there are many of them, you put SIMs >>>>>>>>> inside, and you are good to go. >>>>>>>>> >>>>>>>>> -giovanni >>>>>>>>> >>>>>>>>> >>>>>>>>> sent from mobile >>>>>>>>> cell: +39 347 266 56 18 >>>>>>>>> Giovanni Maruzzelli >>>>>>>>> OpenTelecom.IT >>>>>>>>> >>>>>>>>> Il 04 mag 2017 11:13, "Deepika Yadav" ha >>>>>>>>> scritto: >>>>>>>>> >>>>>>>>>> Hi, >>>>>>>>>> >>>>>>>>>> I am using Freeswitch in an application that initiates conference >>>>>>>>>> calls amongst 20 users in cellular network (mobile phones in GSM network). >>>>>>>>>> Currently, for the VOIP-GSM gateway, we are using the service from a >>>>>>>>>> company called Doorvaani. >>>>>>>>>> >>>>>>>>>> But, since, the cost of call estimates to be high and we cannot >>>>>>>>>> debug the call drops; we are thinking to buy our own PRI card. >>>>>>>>>> >>>>>>>>>> I am seeking recommendation on following points: >>>>>>>>>> >>>>>>>>>> 1. Which card should I buy that is most easily configurable >>>>>>>>>> with the Freeswitch i.e. company and type. >>>>>>>>>> 2. Reference on how should I start to make Freeswitch >>>>>>>>>> configure with the PRI card and start sending and receiving calls. For the >>>>>>>>>> current gateway service in use, I simply put the authentication credentials >>>>>>>>>> for the corresponding VOIP line offered by the company Doorvaani in the >>>>>>>>>> external SIP Profile. In case of PRI, what all changes are needed? >>>>>>>>>> >>>>>>>>>> Regards, >>>>>>>>>> Deepika >>>>>>>>>> >>>>>>>>>> -- >>>>>>>>>> Regards >>>>>>>>>> Deepika >>>>>>>>>> https://www.iiitd.edu.in/~deepikay/ >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> _________________________________________________________________________ >>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>> consulting at freeswitch.org >>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>> >>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>> http://www.freeswitch.org >>>>>>>>>> http://confluence.freeswitch.org >>>>>>>>>> http://www.cluecon.com >>>>>>>>>> >>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>> FreeSWITCH-users at 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://www.iiitd.edu.in/~deepikay/ >>>>>>>> >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> 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 >>>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at 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://www.iiitd.edu.in/~deepikay/ >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 deepikay at iiitd.ac.in Sun May 21 05:25:33 2017 From: deepikay at iiitd.ac.in (Deepika Yadav) Date: Sun, 21 May 2017 10:55:33 +0530 Subject: [Freeswitch-users] Freeswitch PRI support In-Reply-To: References: <27a9d063.ad3b.15bd85e9762.Coremail.xxxman2008@126.com> <6fc1c046-45f9-cdbe-2896-a077dbcc5dad@xbipin.com> Message-ID: Thanks Luiz for sharing the information. Right now I don't have any clue about GSMOpen, I will definitely look into it, seems interesting. Regards, Deepika On May 21, 2017 5:36 AM, "Luiz Fernando Softov" wrote: > Hello, here I use GsmOpen with some modifications. > > This work very well in my clients, some of then using with 4, 16, 27, 32 > even 64 modens. > > There are different user cases. > > Some of then are using in ISPs making a Point to Point in rural areas in > Brazil. > > There are others clients, using in CallCenter, to call clients. > > Here there is 4 main operators, VIVO, OI, CLARO, TIM, and client can make > portability. > > So, the system has a database of all portability. And recognizes when a > number is from VIVO, but got portability to CLARO, etc. > The system calls using one of CLARO chips, according with the rules of the > routes. > > There is another user case, when a company want to give the clients more > numbers to receive calls, so client doesn't make expensive calls. > > And, there is a lot of user cases more. This is just the main. > > > The main problem is to manage dongles, because sometimes, with a bad power > supply, the modem get down and on (RESET). > So the TTY port can change. > > I tried to use GSMOpen to recognize the tty by IMSI, without success. > > So I made my own daemon, that load all modems and tty, and using gsmopen > reload + XML Curl, I can made changes in FreeSwitch on the fly. > > Here is expensive the cost of GSM Gateways. > Some of then are R$ 4000 with 8 ports. > Or R$ 16000 with 32 ports. > > Using GSMOpen, I can make a system with 32 modems with only R$ 500. > > I have at least 50 companies using this solution. > And ~ 200 companies using my system. > > Search in the list for my name, there is a lot of e-mails talking about it. > > If you wan't I can give you my patch, there is a lot of corrections. > The main of then are about Thread getting deadlock. > > > > > > > 2017-05-11 11:31 GMT-04:00 Deepika Yadav : > >> This is for the usecase of community heath worker in rural areas of India >> who only have access to feature phones i.e. no Internet, no data plan. >> >> Regards, >> Deepika >> >> On Thu, May 11, 2017 at 8:05 PM, Colton Conor >> wrote: >> >>> What is the value? Mobile voice plans are expensive (realtive to sip >>> trunks). Also, call quality is not even G711 in most cases. >>> >>> I could see the use case is you wanted a mobile backup with GSM for >>> voice in the case your hardline internet were down, but even then I would >>> think it would be cheaper and more benefical to get a mobile data backup >>> plan, and continue to use the SIP trunks over the internet. >>> >>> I honestly had now clue these large GSM banks existed. >>> >>> Anyone used them in North America? >>> >>> On Wed, May 10, 2017 at 10:06 AM, David Villasmil < >>> david.villasmil.work at gmail.com> wrote: >>> >>>> I think it's illegal everywhere... but value? there's lots and lots of >>>> values, it only depends on where you are doing it... :) >>>> ᐧ >>>> >>>> Regards, >>>> >>>> David Villasmil >>>> email: david.villasmil.work at gmail.com >>>> phone: +34669448337 <+34%20669%2044%2083%2037> >>>> >>>> On Wed, May 10, 2017 at 4:12 PM, Raúl Alexis Betancor Santana < >>>> rbetancor at gmail.com> wrote: >>>> >>>>> The only real advantage over a regular SIP trunk from any provider, >>>>> it's that you could select a plan than MAYBE allow you some sort of >>>>> flat-rate and not per-minute use plans ... but it doesn't worth the effort >>>>> and money you have to spend to get that to generate revenue ... also most >>>>> of celullar carriers withh put some restrictions on the usage of the >>>>> flat-rate, as all of them have 'fair use' policies. >>>>> At least on Spain, it's ilegal to resell that kind of traffic and I >>>>> asume its the same in most of the world. >>>>> >>>>> 2017-05-10 15:04 GMT+01:00 Colton Conor : >>>>> >>>>>> So how does this GSM stuff work in the USA? Would I got buy a cell >>>>>> phone plan from someone like AT&T or T-Mobile, and then insert the SIM into >>>>>> these devices. The device would then covert the audio from GSM to SIP, and >>>>>> then I could use it like a SIP trunk? >>>>>> >>>>>> What is the advantage of doing this over just buying a pure SIP trunk >>>>>> from an internet provider? >>>>>> >>>>>> >>>>>> >>>>>> On Wed, May 10, 2017 at 4:35 AM, David Villasmil < >>>>>> david.villasmil.work at gmail.com> wrote: >>>>>> >>>>>>> There's another chinese brand that has a 32 port, I've used it, >>>>>>> works pretty well, i think it's "GoIP" or something. >>>>>>> >>>>>>> On Wed, May 10, 2017 at 10:39 AM Bipin Patel >>>>>>> wrote: >>>>>>> >>>>>>>> hi, >>>>>>>> >>>>>>>> we use matrix boxes but 4 port ones, they work quiet well >>>>>>>> >>>>>>>> >>>>>>>> Regards, >>>>>>>> Bipin >>>>>>>> >>>>>>>> >>>>>>>> ------------------------------ >>>>>>>> -------- Original Message -------- >>>>>>>> Subject: Re: [Freeswitch-users] Freeswitch PRI support >>>>>>>> From: Deepika Yadav >>>>>>>> To: FreeSWITCH Users Help >>>>>>>> >>>>>>>> Date: 5/10/2017, 11:02:49 AM >>>>>>>> >>>>>>>> Hi, >>>>>>>> >>>>>>>> I searched some gateways online. The ones available at ebay are >>>>>>>> cheap, the following link shows a gateway supporting 32 SIMS for $859 : >>>>>>>> >>>>>>>> http://www.ebay.com/itm/GoIP-32-32-Port-VoIP-GSM-Gateway-wit >>>>>>>> h-32-External-Antenna-/291811535942 >>>>>>>> >>>>>>>> However, when I contacted a company called "Matrix Telecom >>>>>>>> Solution", they gave the quote of $4150 for similar number of SIM support. >>>>>>>> On asking them to compare the two gateways, they said that the one >>>>>>>> available at ebay is a Chinese gateway about which they are apprehensive >>>>>>>> for the quality, warranty and working. >>>>>>>> >>>>>>>> Regards, >>>>>>>> Deepika >>>>>>>> >>>>>>>> >>>>>>>> On Fri, May 5, 2017 at 4:58 PM, Raymond wrote: >>>>>>>> >>>>>>>>> >>>>>>>>> HI, >>>>>>>>> >>>>>>>>> Regarding this. i have asked some question of module "GSMOpen". >>>>>>>>> >>>>>>>>> >>>>>>>>> In fact ,i want to make Freeswitch box as "GSM VOIP Gateway" by >>>>>>>>> using "gsmopen". But...seems no one help me . :-(( . If it worked , can >>>>>>>>> save more money than hardware gsm voip gateway ( GOIP ). >>>>>>>>> >>>>>>>>> Raymond >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> At 2017-05-04 17:24:13, "Giovanni Maruzzelli" >>>>>>>>> wrote: >>>>>>>>> >>>>>>>>> For PRI you use Sangoma or Patton. >>>>>>>>> >>>>>>>>> But, why don't you use an hardware gateway sip<->gsm? >>>>>>>>> >>>>>>>>> It would save you very big money. >>>>>>>>> >>>>>>>>> Check on ebay and google, there are many of them, you put SIMs >>>>>>>>> inside, and you are good to go. >>>>>>>>> >>>>>>>>> -giovanni >>>>>>>>> >>>>>>>>> >>>>>>>>> sent from mobile >>>>>>>>> cell: +39 347 266 56 18 >>>>>>>>> Giovanni Maruzzelli >>>>>>>>> OpenTelecom.IT >>>>>>>>> >>>>>>>>> Il 04 mag 2017 11:13, "Deepika Yadav" ha >>>>>>>>> scritto: >>>>>>>>> >>>>>>>>>> Hi, >>>>>>>>>> >>>>>>>>>> I am using Freeswitch in an application that initiates conference >>>>>>>>>> calls amongst 20 users in cellular network (mobile phones in GSM network). >>>>>>>>>> Currently, for the VOIP-GSM gateway, we are using the service from a >>>>>>>>>> company called Doorvaani. >>>>>>>>>> >>>>>>>>>> But, since, the cost of call estimates to be high and we cannot >>>>>>>>>> debug the call drops; we are thinking to buy our own PRI card. >>>>>>>>>> >>>>>>>>>> I am seeking recommendation on following points: >>>>>>>>>> >>>>>>>>>> 1. Which card should I buy that is most easily configurable >>>>>>>>>> with the Freeswitch i.e. company and type. >>>>>>>>>> 2. Reference on how should I start to make Freeswitch >>>>>>>>>> configure with the PRI card and start sending and receiving calls. For the >>>>>>>>>> current gateway service in use, I simply put the authentication credentials >>>>>>>>>> for the corresponding VOIP line offered by the company Doorvaani in the >>>>>>>>>> external SIP Profile. In case of PRI, what all changes are needed? >>>>>>>>>> >>>>>>>>>> Regards, >>>>>>>>>> Deepika >>>>>>>>>> >>>>>>>>>> -- >>>>>>>>>> Regards >>>>>>>>>> Deepika >>>>>>>>>> https://www.iiitd.edu.in/~deepikay/ >>>>>>>>>> >>>>>>>>>> ____________________________________________________________ >>>>>>>>>> _____________ >>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>> consulting at freeswitch.org >>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>> >>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>> http://www.freeswitch.org >>>>>>>>>> http://confluence.freeswitch.org >>>>>>>>>> http://www.cluecon.com >>>>>>>>>> >>>>>>>>>> FreeSWITCH-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 >>>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> -- >>>>>>>> Regards >>>>>>>> Deepika >>>>>>>> https://www.iiitd.edu.in/~deepikay/ >>>>>>>> >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> 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/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/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/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://www.iiitd.edu.in/~deepikay/ >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 Sun May 21 05:41:03 2017 From: bipin at xbipin.com (Bipin Patel) Date: Sun, 21 May 2017 09:41:03 +0400 Subject: [Freeswitch-users] Freeswitch PRI support In-Reply-To: References: <27a9d063.ad3b.15bd85e9762.Coremail.xxxman2008@126.com> <6fc1c046-45f9-cdbe-2896-a077dbcc5dad@xbipin.com> Message-ID: hi, this would be interesting as we too faced similar issue with gsm dongles, kindly share the patch upstream so every1 can benefit Regards, Bipin ------------------------------------------------------------------------ -------- Original Message -------- Subject: Re: [Freeswitch-users] Freeswitch PRI support From: Luiz Fernando Softov To: FreeSWITCH Users Help Date: 5/21/2017, 4:02:24 AM > Hello, here I use GsmOpen with some modifications. > > This work very well in my clients, some of then using with 4, 16, 27, > 32 even 64 modens. > > There are different user cases. > > Some of then are using in ISPs making a Point to Point in rural areas > in Brazil. > > There are others clients, using in CallCenter, to call clients. > > Here there is 4 main operators, VIVO, OI, CLARO, TIM, and client can > make portability. > > So, the system has a database of all portability. And recognizes when > a number is from VIVO, but got portability to CLARO, etc. > The system calls using one of CLARO chips, according with the rules of > the routes. > > There is another user case, when a company want to give the clients > more numbers to receive calls, so client doesn't make expensive calls. > > And, there is a lot of user cases more. This is just the main. > > > The main problem is to manage dongles, because sometimes, with a bad > power supply, the modem get down and on (RESET). > So the TTY port can change. > > I tried to use GSMOpen to recognize the tty by IMSI, without success. > > So I made my own daemon, that load all modems and tty, and using > gsmopen reload + XML Curl, I can made changes in FreeSwitch on the fly. > > Here is expensive the cost of GSM Gateways. > Some of then are R$ 4000 with 8 ports. > Or R$ 16000 with 32 ports. > > Using GSMOpen, I can make a system with 32 modems with only R$ 500. > > I have at least 50 companies using this solution. > And ~ 200 companies using my system. > > Search in the list for my name, there is a lot of e-mails talking > about it. > > If you wan't I can give you my patch, there is a lot of corrections. > The main of then are about Thread getting deadlock. > > > > > > > 2017-05-11 11:31 GMT-04:00 Deepika Yadav >: > > This is for the usecase of community heath worker in rural areas > of India who only have access to feature phones i.e. no Internet, > no data plan. > > Regards, > Deepika > > On Thu, May 11, 2017 at 8:05 PM, Colton Conor > > wrote: > > What is the value? Mobile voice plans are expensive (realtive > to sip trunks). Also, call quality is not even G711 in most > cases. > > I could see the use case is you wanted a mobile backup with > GSM for voice in the case your hardline internet were down, > but even then I would think it would be cheaper and more > benefical to get a mobile data backup plan, and continue to > use the SIP trunks over the internet. > > I honestly had now clue these large GSM banks existed. > > Anyone used them in North America? > > On Wed, May 10, 2017 at 10:06 AM, David Villasmil > > wrote: > > I think it's illegal everywhere... but value? there's lots > and lots of values, it only depends on where you are doing > it... :) > ᐧ > > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > > phone: +34669448337 > > On Wed, May 10, 2017 at 4:12 PM, Raúl Alexis Betancor > Santana > > wrote: > > The only real advantage over a regular SIP trunk from > any provider, it's that you could select a plan than > MAYBE allow you some sort of flat-rate and not > per-minute use plans ... but it doesn't worth the > effort and money you have to spend to get that to > generate revenue ... also most of celullar carriers > withh put some restrictions on the usage of the > flat-rate, as all of them have 'fair use' policies. > At least on Spain, it's ilegal to resell that kind of > traffic and I asume its the same in most of the world. > > 2017-05-10 15:04 GMT+01:00 Colton Conor > >: > > So how does this GSM stuff work in the USA? Would > I got buy a cell phone plan from someone like AT&T > or T-Mobile, and then insert the SIM into these > devices. The device would then covert the audio > from GSM to SIP, and then I could use it like a > SIP trunk? > > What is the advantage of doing this over just > buying a pure SIP trunk from an internet provider? > > > > On Wed, May 10, 2017 at 4:35 AM, David Villasmil > > wrote: > > There's another chinese brand that has a 32 > port, I've used it, works pretty well, i think > it's "GoIP" or something. > > On Wed, May 10, 2017 at 10:39 AM Bipin Patel > > > wrote: > > hi, > > we use matrix boxes but 4 port ones, they > work quiet well > > > Regards, > Bipin > > > ------------------------------------------------------------------------ > -------- Original Message -------- > Subject: Re: [Freeswitch-users] Freeswitch > PRI support > From: Deepika Yadav > > To: FreeSWITCH Users Help > > > Date: 5/10/2017, 11:02:49 AM >> Hi, >> >> I searched some gateways online. The ones >> available at ebay are cheap, the >> following link shows a gateway supporting >> 32 SIMS for $859 : >> >> http://www.ebay.com/itm/GoIP-32-32-Port-VoIP-GSM-Gateway-with-32-External-Antenna-/291811535942 >> >> >> However, when I contacted a company >> called "Matrix Telecom Solution", they >> gave the quote of $4150 for similar >> number of SIM support. On asking them to >> compare the two gateways, they said that >> the one available at ebay is a Chinese >> gateway about which they are apprehensive >> for the quality, warranty and working. >> >> Regards, >> Deepika >> >> >> On Fri, May 5, 2017 at 4:58 PM, Raymond >> > > wrote: >> >> >> HI, >> >> Regarding this. i have asked some >> question of module "GSMOpen". >> >> In fact ,i want to make Freeswitch >> box as "GSM VOIP Gateway" by using >> "gsmopen". But...seems no one help >> me . :-(( . If it worked , can save >> more money than hardware gsm voip >> gateway ( GOIP ). >> >> Raymond >> >> >> >> At 2017-05-04 17:24:13, "Giovanni >> Maruzzelli" > > wrote: >> >> For PRI you use Sangoma or Patton. >> >> But, why don't you use an >> hardware gateway sip<->gsm? >> >> It would save you very big money. >> >> Check on ebay and google, there >> are many of them, you put SIMs >> inside, and you are good to go. >> >> -giovanni >> >> >> sent from mobile >> cell: +39 347 266 56 18 >> Giovanni Maruzzelli >> OpenTelecom.IT >> >> Il 04 mag 2017 11:13, "Deepika >> Yadav" > > ha >> scritto: >> >> Hi, >> >> I am using Freeswitch in an >> application that initiates >> conference calls amongst 20 >> users in cellular network >> (mobile phones in GSM >> network). Currently, for the >> VOIP-GSM gateway, we are >> using the service from a >> company called Doorvaani. >> >> But, since, the cost of call >> estimates to be high and we >> cannot debug the call drops; >> we are thinking to buy our >> own PRI card. >> >> I am seeking recommendation >> on following points: >> >> 1. Which card should I buy >> that is most easily >> configurable with the >> Freeswitch i.e. company >> and type. >> 2. Reference on how should I >> start to make Freeswitch >> configure with the PRI >> card and start sending >> and receiving calls. For >> the current gateway >> service in use, I simply >> put the authentication >> credentials for the >> corresponding VOIP line >> offered by the company >> Doorvaani in the external >> SIP Profile. In case of >> PRI, what all changes are >> needed? >> >> Regards, >> Deepika >> >> -- >> Regards >> Deepika >> https://www.iiitd.edu.in/~deepikay/ >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH >> Consulting Services: >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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://www.iiitd.edu.in/~deepikay/ >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 > > > > > -- > Regards > Deepika > https://www.iiitd.edu.in/~deepikay/ > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 petedao at gmail.com Sun May 21 10:42:35 2017 From: petedao at gmail.com (Pete Kay) Date: Sun, 21 May 2017 18:42:35 +0800 Subject: [Freeswitch-users] Questions about api_hangup_hook Message-ID: Hi I tried to pass the session variable to a restful interface but much of the variable are giving . Does anyone know how I can pass channel variable in the curl? If we can't pass all variables, can we past at least the hangup-cause? I tried the following and it is giving empty: 2017-05-21 12:11:21.619542 [DEBUG] switch_core_state_machine.c:783 Hangup Command with Session curl( http://198.130.149.164:8000/test_takt/Records?callerid=&answer_time=&hangup=1000&hangup2=0&sip=EARLY MEDIA&state=&originate_disposition=&hangup_cause=&sip_term_status=&proto_specific_hangup_cause=&variable_sip_term_status=&variable_sip_hangup_disposition=&endpoint_disposition=EARLY MEDIA&variable_endpoint_disposition=&sip_hangup_disposition=&answer_stamp=&answer_epoch=&end=): The other question i have is that I tried to execute two api in the api_hangup_hook, it does not call the second api. Does anyone know how to do this or is it possible to do? -------------- next part -------------- An HTML attachment was scrubbed... URL: From bipin at xbipin.com Sun May 21 12:32:55 2017 From: bipin at xbipin.com (Bipin Patel) Date: Sun, 21 May 2017 16:32:55 +0400 Subject: [Freeswitch-users] return to shell only when call from originate is complete? In-Reply-To: <4112C9C4-5BAA-408F-9731-8B11F3CCDAD8@stratusvideo.com> References: <69660425-2e51-1943-4fba-fc1ff5ea7094@xbipin.com> <4E8BF635-E985-4056-B301-A1EEFF702B09@jerris.com> <8eb5928b-d77f-eb70-fa0b-84f9d21f7d5b@xbipin.com> <4112C9C4-5BAA-408F-9731-8B11F3CCDAD8@stratusvideo.com> Message-ID: <0f62f615-0556-55aa-c52b-21955f56e1db@xbipin.com> hi, i converted the bash script to python as below: ret = subprocess.Popen('sudo fs_cli -x \"show channels\" | grep -c \"'+UUID+'\"', shell=True, stdout=subprocess.PIPE, stderr=subprocess.PIPE) sout, serr = ret.communicate() print sout the reason i use -c instead of -q is because fs_cli always returns a 0 exit code so it doesnt work through python when combined with grep so i tried to exit loop when the counter is 0 meaning the call is completed, but the above returns some weird numbers in sout, when the call is running it shows 5 and when done shows 3 but same if i run directly through console it shows accurately. any idea how to fix this? Regards, Bipin ------------------------------------------------------------------------ -------- Original Message -------- Subject: Re: [Freeswitch-users] return to shell only when call from originate is complete? From: William Simon To: FreeSWITCH Users Help Date: 5/20/2017, 1:29:05 AM > If you are using fs_cli then a simple shell script can monitor the > call and return when the call completes: > > #!/bin/bash > > RESULT=$( fs_cli -x "originate leg1 leg2" ) > > if [[ $RESULT =~ "OK" ]]; then > UUID=$( echo "$RESULT" | cut -f 2 -d ' ' ) > while fs_cli -x "show channels" | grep -q $UUID ; > do > sleep 5; > done > echo "Call completed." > fi > > > > >> On May 19, 2017, at 12:31 PM, Bipin Patel > > wrote: >> >> hi, >> >> any other way this can be done >> >> >> Regards, >> Bipin >> >> >> ------------------------------------------------------------------------ >> -------- Original Message -------- >> Subject: Re: [Freeswitch-users] return to shell only when call from >> originate is complete? >> From: Michael Jerris >> To: Bipin Patel >> Date: 5/19/2017, 7:48:22 PM >>> No, thats not what originate does. >>> >>>> On May 19, 2017, at 11:43 AM, Bipin Patel >>> > wrote: >>>> >>>> hi, >>>> >>>> does it happen right after answer? >>>> is there any way to lock it so it returns only after failure or >>>> after hangup? yes im using ignore_early_media >>>> >>>> >>>> Regards, >>>> Bipin >>>> >>>> >>>> ------------------------------------------------------------------------ >>>> -------- Original Message -------- >>>> Subject: Re: [Freeswitch-users] return to shell only when call from >>>> originate is complete? >>>> From: Michael Jerris >>>> To: FreeSWITCH Users Help >>>> Date: 5/19/2017, 7:34:35 PM >>>>> originate returns when we get call progress, or in the case of >>>>> ignore_early_media, on answer or failure. >>>>> >>>>>> On May 19, 2017, at 10:36 AM, Bipin Patel >>>>> > wrote: >>>>>> >>>>>> hi, >>>>>> >>>>>> i was wondering if its possible to to release handle from an >>>>>> originate command only once the call is completed, i mean i send >>>>>> a originate command from shell using fs_cli but only return back >>>>>> to prompt once the call is completed rather than as soon as a >>>>>> bridge is created or a UUID generated >>>>>> >>>>> >>> >> >> >> >> “The information transmitted is intended only for the person or >> entity to which it is addressed and may contain proprietary, >> business-confidential and/or privileged material. If you are not the >> intended recipient of this message you are hereby notified that any >> use, review, retransmission, dissemination, distribution, >> reproduction or any action taken in reliance upon this message is >> prohibited. If you received this in error, please contact the sender >> and delete the material from any computer.” >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > “The information transmitted is intended only for the person or entity > to which it is addressed and may contain proprietary, > business-confidential and/or privileged material. If you are not the > intended recipient of this message you are hereby notified that any > use, review, retransmission, dissemination, distribution, reproduction > or any action taken in reliance upon this message is prohibited. If > you received this in error, please contact the sender and delete the > material from any computer.” > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 Sun May 21 15:25:31 2017 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Sun, 21 May 2017 17:25:31 +0200 Subject: [Freeswitch-users] return to shell only when call from originate is complete? In-Reply-To: <0f62f615-0556-55aa-c52b-21955f56e1db@xbipin.com> References: <69660425-2e51-1943-4fba-fc1ff5ea7094@xbipin.com> <4E8BF635-E985-4056-B301-A1EEFF702B09@jerris.com> <8eb5928b-d77f-eb70-fa0b-84f9d21f7d5b@xbipin.com> <4112C9C4-5BAA-408F-9731-8B11F3CCDAD8@stratusvideo.com> <0f62f615-0556-55aa-c52b-21955f56e1db@xbipin.com> Message-ID: i still think you are better consume events with python... its easy and straightforward #!/usr/bin/env python import string import sys from ESL import * con = ESLconnection("127.0.0.1","8021","ClueCon") #are we connected? callDirection = "unknown" if con.connected: print "we connected \n" con.events("plain", "all"); while 1: #my $e = $con->recvEventTimed(100); e = con.recvEvent() if e: #print e.serialize() en = e.getHeader("Event-Name") print "Name =>" ,en if en == "CHANNEL_OUTGOING": callDirection = "OUT" print "Direction =", callDirection elif en == "CHANNEL_ORIGINATE": print "Originate call - direction =", callDirection elif en == "CHANNEL_CALLSTATE": print "Call State =>", e.getHeader("Answer-State") elif en == "CHANNEL_ANSWER": print "Call START - direction =", callDirection elif en == "CHANNEL_HANGUP": print "Call END - direction =", callDirection elif en == "CHANNEL_PARK": callDirection = "IN" print "Call PARK - direction =", callDirection you can work it out from here pretty everything you want. T. On 21 May 2017 at 14:32, Bipin Patel wrote: > hi, > > i converted the bash script to python as below: > > ret = subprocess.Popen('sudo fs_cli -x \"show channels\" | grep -c > \"'+UUID+'\"', shell=True, stdout=subprocess.PIPE, stderr=subprocess.PIPE) > sout, serr = ret.communicate() > print sout > > the reason i use -c instead of -q is because fs_cli always returns a 0 > exit code so it doesnt work through python when combined with grep so i > tried to exit loop when the counter is 0 meaning the call is completed, but > the above returns some weird numbers in sout, when the call is running it > shows 5 and when done shows 3 but same if i run directly through console it > shows accurately. > > any idea how to fix this? > > > Regards, > Bipin > > > ------------------------------ > -------- Original Message -------- > Subject: Re: [Freeswitch-users] return to shell only when call from > originate is complete? > From: William Simon > To: FreeSWITCH Users Help > > Date: 5/20/2017, 1:29:05 AM > > If you are using fs_cli then a simple shell script can monitor the call > and return when the call completes: > > #!/bin/bash > > RESULT=$( fs_cli -x "originate leg1 leg2" ) > > if [[ $RESULT =~ "OK" ]]; then > UUID=$( echo "$RESULT" | cut -f 2 -d ' ' ) > while fs_cli -x "show channels" | grep -q $UUID ; > do > sleep 5; > done > echo "Call completed." > fi > > > > > On May 19, 2017, at 12:31 PM, Bipin Patel wrote: > > hi, > > any other way this can be done > > > Regards, > Bipin > > > ------------------------------ > -------- Original Message -------- > Subject: Re: [Freeswitch-users] return to shell only when call from > originate is complete? > From: Michael Jerris > To: Bipin Patel > Date: 5/19/2017, 7:48:22 PM > > No, thats not what originate does. > > On May 19, 2017, at 11:43 AM, Bipin Patel wrote: > > hi, > > does it happen right after answer? > is there any way to lock it so it returns only after failure or after > hangup? yes im using ignore_early_media > > > Regards, > Bipin > > > ------------------------------ > -------- Original Message -------- > Subject: Re: [Freeswitch-users] return to shell only when call from > originate is complete? > From: Michael Jerris > To: FreeSWITCH Users Help > > Date: 5/19/2017, 7:34:35 PM > > originate returns when we get call progress, or in the case of > ignore_early_media, on answer or failure. > > On May 19, 2017, at 10:36 AM, Bipin Patel wrote: > > hi, > > i was wondering if its possible to to release handle from an originate > command only once the call is completed, i mean i send a originate command > from shell using fs_cli but only return back to prompt once the call is > completed rather than as soon as a bridge is created or a UUID generated > > > > > > > “The information transmitted is intended only for the person or entity to > which it is addressed and may contain proprietary, business-confidential > and/or privileged material. If you are not the intended recipient of this > message you are hereby notified that any use, review, retransmission, > dissemination, distribution, reproduction or any action taken in reliance > upon this message is prohibited. If you received this in error, please > contact the sender and delete the material from any computer.” > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > “The information transmitted is intended only for the person or entity to > which it is addressed and may contain proprietary, business-confidential > and/or privileged material. If you are not the intended recipient of this > message you are hereby notified that any use, review, retransmission, > dissemination, distribution, reproduction or any action taken in reliance > upon this message is prohibited. If you received this in error, please > contact the sender and delete the material from any computer.” > > _________________________________________________________________________ > 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 Sun May 21 15:42:14 2017 From: bipin at xbipin.com (Bipin Patel) Date: Sun, 21 May 2017 19:42:14 +0400 Subject: [Freeswitch-users] return to shell only when call from originate is complete? In-Reply-To: References: <69660425-2e51-1943-4fba-fc1ff5ea7094@xbipin.com> <4E8BF635-E985-4056-B301-A1EEFF702B09@jerris.com> <8eb5928b-d77f-eb70-fa0b-84f9d21f7d5b@xbipin.com> <4112C9C4-5BAA-408F-9731-8B11F3CCDAD8@stratusvideo.com> <0f62f615-0556-55aa-c52b-21955f56e1db@xbipin.com> Message-ID: hi, im trying to avoid going the esl route, for now i solved this by calling a bash script from python which monitors the call once the UUID is passed to it so that holds the python command till call is over and then it proceeds further Regards, Bipin ------------------------------------------------------------------------ -------- Original Message -------- Subject: Re: [Freeswitch-users] return to shell only when call from originate is complete? From: Tihomir Culjaga To: FreeSWITCH Users Help Date: 5/21/2017, 7:25:31 PM > i still think you are better consume events with python... > its easy and straightforward > > > #!/usr/bin/env python > > import string > import sys > > from ESL import * > > con = ESLconnection("127.0.0.1","8021","ClueCon") > #are we connected? > callDirection = "unknown" > > > if con.connected: > print "we connected \n" > con.events("plain", "all"); > > while 1: > #my $e = $con->recvEventTimed(100); > e = con.recvEvent() > > if e: > #print e.serialize() > en = e.getHeader("Event-Name") > print "Name =>" ,en > > if en == "CHANNEL_OUTGOING": > callDirection = "OUT" > print "Direction =", callDirection > > elif en == "CHANNEL_ORIGINATE": > print "Originate call - direction =", callDirection > > elif en == "CHANNEL_CALLSTATE": > print "Call State =>", e.getHeader("Answer-State") > > elif en == "CHANNEL_ANSWER": > print "Call START - direction =", callDirection > > elif en == "CHANNEL_HANGUP": > print "Call END - direction =", callDirection > > elif en == "CHANNEL_PARK": > callDirection = "IN" > print "Call PARK - direction =", callDirection > > > > you can work it out from here pretty everything you want. > > > T. > > On 21 May 2017 at 14:32, Bipin Patel > wrote: > > hi, > > i converted the bash script to python as below: > > ret = subprocess.Popen('sudo fs_cli -x \"show channels\" | grep -c > \"'+UUID+'\"', shell=True, stdout=subprocess.PIPE, > stderr=subprocess.PIPE) > sout, serr = ret.communicate() > print sout > > the reason i use -c instead of -q is because fs_cli always returns > a 0 exit code so it doesnt work through python when combined with > grep so i tried to exit loop when the counter is 0 meaning the > call is completed, but the above returns some weird numbers in > sout, when the call is running it shows 5 and when done shows 3 > but same if i run directly through console it shows accurately. > > any idea how to fix this? > > > Regards, > Bipin > > > ------------------------------------------------------------------------ > -------- Original Message -------- > Subject: Re: [Freeswitch-users] return to shell only when call > from originate is complete? > From: William Simon > > To: FreeSWITCH Users Help > > Date: 5/20/2017, 1:29:05 AM >> If you are using fs_cli then a simple shell script can monitor >> the call and return when the call completes: >> >> #!/bin/bash >> >> RESULT=$( fs_cli -x "originate leg1 leg2" ) >> >> if [[ $RESULT =~ "OK" ]]; then >> UUID=$( echo "$RESULT" | cut -f 2 -d ' ' ) >> while fs_cli -x "show channels" | grep -q $UUID ; >> do >> sleep 5; >> done >> echo "Call completed." >> fi >> >> >> >> >>> On May 19, 2017, at 12:31 PM, Bipin Patel >> > wrote: >>> >>> hi, >>> >>> any other way this can be done >>> >>> >>> Regards, >>> Bipin >>> >>> >>> ------------------------------------------------------------------------ >>> -------- Original Message -------- >>> Subject: Re: [Freeswitch-users] return to shell only when call >>> from originate is complete? >>> From: Michael Jerris >>> To: Bipin Patel >>> Date: 5/19/2017, 7:48:22 PM >>>> No, thats not what originate does. >>>> >>>>> On May 19, 2017, at 11:43 AM, Bipin Patel >>>> > wrote: >>>>> >>>>> hi, >>>>> >>>>> does it happen right after answer? >>>>> is there any way to lock it so it returns only after failure >>>>> or after hangup? yes im using ignore_early_media >>>>> >>>>> >>>>> Regards, >>>>> Bipin >>>>> >>>>> >>>>> ------------------------------------------------------------------------ >>>>> -------- Original Message -------- >>>>> Subject: Re: [Freeswitch-users] return to shell only when call >>>>> from originate is complete? >>>>> From: Michael Jerris >>>>> To: FreeSWITCH Users >>>>> Help >>>>> >>>>> Date: 5/19/2017, 7:34:35 PM >>>>>> originate returns when we get call progress, or in the case >>>>>> of ignore_early_media, on answer or failure. >>>>>> >>>>>>> On May 19, 2017, at 10:36 AM, Bipin Patel >>>>>> > wrote: >>>>>>> >>>>>>> hi, >>>>>>> >>>>>>> i was wondering if its possible to to release handle from an >>>>>>> originate command only once the call is completed, i mean i >>>>>>> send a originate command from shell using fs_cli but only >>>>>>> return back to prompt once the call is completed rather than >>>>>>> as soon as a bridge is created or a UUID generated >>>>>>> >>>>>> >>>> >>> >>> >>> >>> “The information transmitted is intended only for the person or >>> entity to which it is addressed and may contain proprietary, >>> business-confidential and/or privileged material. If you are not >>> the intended recipient of this message you are hereby notified >>> that any use, review, retransmission, dissemination, >>> distribution, reproduction or any action taken in reliance upon >>> this message is prohibited. If you received this in error, >>> please contact the sender and delete the material from any >>> computer.” >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >>> http://www.freeswitch.org >> >> >> >> “The information transmitted is intended only for the person or >> entity to which it is addressed and may contain proprietary, >> business-confidential and/or privileged material. If you are not >> the intended recipient of this message you are hereby notified >> that any use, review, retransmission, dissemination, >> distribution, reproduction or any action taken in reliance upon >> this message is prohibited. If you received this in error, please >> contact the sender and delete the material from any computer.” >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 mike at jerris.com Mon May 22 04:25:03 2017 From: mike at jerris.com (Michael Jerris) Date: Mon, 22 May 2017 04:25:03 +0000 Subject: [Freeswitch-users] return to shell only when call from originate is complete? In-Reply-To: References: <69660425-2e51-1943-4fba-fc1ff5ea7094@xbipin.com> <4E8BF635-E985-4056-B301-A1EEFF702B09@jerris.com> <8eb5928b-d77f-eb70-fa0b-84f9d21f7d5b@xbipin.com> <4112C9C4-5BAA-408F-9731-8B11F3CCDAD8@stratusvideo.com> <0f62f615-0556-55aa-c52b-21955f56e1db@xbipin.com> Message-ID: fs_cli IS the esl route. Cleaner to just use esl and call the api command you want directly instead of hacky stuff like this. On Sun, May 21, 2017 at 11:44 AM Bipin Patel wrote: > hi, > > im trying to avoid going the esl route, for now i solved this by calling a > bash script from python which monitors the call once the UUID is passed to > it so that holds the python command till call is over and then it proceeds > further > > > Regards, > Bipin > > > ------------------------------ > -------- Original Message -------- > Subject: Re: [Freeswitch-users] return to shell only when call from > originate is complete? > From: Tihomir Culjaga > To: FreeSWITCH Users Help > > Date: 5/21/2017, 7:25:31 PM > > i still think you are better consume events with python... > its easy and straightforward > > > #!/usr/bin/env python > > import string > import sys > > from ESL import * > > con = ESLconnection("127.0.0.1","8021","ClueCon") > #are we connected? > callDirection = "unknown" > > > if con.connected: > print "we connected \n" > con.events("plain", "all"); > > while 1: > #my $e = $con->recvEventTimed(100); > e = con.recvEvent() > > if e: > #print e.serialize() > en = e.getHeader("Event-Name") > print "Name =>" ,en > > if en == "CHANNEL_OUTGOING": > callDirection = "OUT" > print "Direction =", callDirection > > elif en == "CHANNEL_ORIGINATE": > print "Originate call - direction =", callDirection > > elif en == "CHANNEL_CALLSTATE": > print "Call State =>", e.getHeader("Answer-State") > > elif en == "CHANNEL_ANSWER": > print "Call START - direction =", callDirection > > elif en == "CHANNEL_HANGUP": > print "Call END - direction =", callDirection > > elif en == "CHANNEL_PARK": > callDirection = "IN" > print "Call PARK - direction =", callDirection > > > > you can work it out from here pretty everything you want. > > > T. > > On 21 May 2017 at 14:32, Bipin Patel wrote: > >> hi, >> >> i converted the bash script to python as below: >> >> ret = subprocess.Popen('sudo fs_cli -x \"show channels\" | grep -c >> \"'+UUID+'\"', shell=True, stdout=subprocess.PIPE, stderr=subprocess.PIPE) >> sout, serr = ret.communicate() >> print sout >> >> the reason i use -c instead of -q is because fs_cli always returns a 0 >> exit code so it doesnt work through python when combined with grep so i >> tried to exit loop when the counter is 0 meaning the call is completed, but >> the above returns some weird numbers in sout, when the call is running it >> shows 5 and when done shows 3 but same if i run directly through console it >> shows accurately. >> >> any idea how to fix this? >> >> >> Regards, >> Bipin >> >> >> ------------------------------ >> -------- Original Message -------- >> Subject: Re: [Freeswitch-users] return to shell only when call from >> originate is complete? >> From: William Simon >> To: FreeSWITCH Users Help >> >> Date: 5/20/2017, 1:29:05 AM >> >> If you are using fs_cli then a simple shell script can monitor the call >> and return when the call completes: >> >> #!/bin/bash >> >> RESULT=$( fs_cli -x "originate leg1 leg2" ) >> >> if [[ $RESULT =~ "OK" ]]; then >> UUID=$( echo "$RESULT" | cut -f 2 -d ' ' ) >> while fs_cli -x "show channels" | grep -q $UUID ; >> do >> sleep 5; >> done >> echo "Call completed." >> fi >> >> >> >> >> On May 19, 2017, at 12:31 PM, Bipin Patel wrote: >> >> hi, >> >> any other way this can be done >> >> >> Regards, >> Bipin >> >> >> ------------------------------ >> -------- Original Message -------- >> Subject: Re: [Freeswitch-users] return to shell only when call from >> originate is complete? >> From: Michael Jerris >> To: Bipin Patel >> Date: 5/19/2017, 7:48:22 PM >> >> No, thats not what originate does. >> >> On May 19, 2017, at 11:43 AM, Bipin Patel wrote: >> >> hi, >> >> does it happen right after answer? >> is there any way to lock it so it returns only after failure or after >> hangup? yes im using ignore_early_media >> >> >> Regards, >> Bipin >> >> >> ------------------------------ >> -------- Original Message -------- >> Subject: Re: [Freeswitch-users] return to shell only when call from >> originate is complete? >> From: Michael Jerris >> To: FreeSWITCH Users Help >> >> Date: 5/19/2017, 7:34:35 PM >> >> originate returns when we get call progress, or in the case of >> ignore_early_media, on answer or failure. >> >> On May 19, 2017, at 10:36 AM, Bipin Patel wrote: >> >> hi, >> >> i was wondering if its possible to to release handle from an originate >> command only once the call is completed, i mean i send a originate command >> from shell using fs_cli but only return back to prompt once the call is >> completed rather than as soon as a bridge is created or a UUID generated >> >> >> >> >> >> >> “The information transmitted is intended only for the person or entity to >> which it is addressed and may contain proprietary, business-confidential >> and/or privileged material. If you are not the intended recipient of this >> message you are hereby notified that any use, review, retransmission, >> dissemination, distribution, reproduction or any action taken in reliance >> upon this message is prohibited. If you received this in error, please >> contact the sender and delete the material from any computer.” >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> “The information transmitted is intended only for the person or entity to >> which it is addressed and may contain proprietary, business-confidential >> and/or privileged material. If you are not the intended recipient of this >> message you are hereby notified that any use, review, retransmission, >> dissemination, distribution, reproduction or any action taken in reliance >> upon this message is prohibited. If you received this in error, please >> contact the sender and delete the material from any computer.” >> >> _________________________________________________________________________ >> 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... URL: From anthony.minessale at gmail.com Mon May 22 04:38:13 2017 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 22 May 2017 04:38:13 +0000 Subject: [Freeswitch-users] return to shell only when call from originate is complete? In-Reply-To: References: <69660425-2e51-1943-4fba-fc1ff5ea7094@xbipin.com> <4E8BF635-E985-4056-B301-A1EEFF702B09@jerris.com> <8eb5928b-d77f-eb70-fa0b-84f9d21f7d5b@xbipin.com> <4112C9C4-5BAA-408F-9731-8B11F3CCDAD8@stratusvideo.com> <0f62f615-0556-55aa-c52b-21955f56e1db@xbipin.com> Message-ID: I am trying to catch a fish can you help me? Get a pole and use some fishing line with a hook and add a worm. I dont really want to use a pole, I am just going to throw a grenade in the water and a fish will fall into my boat. On Sun, May 21, 2017 at 10:43 AM Bipin Patel wrote: > hi, > > im trying to avoid going the esl route, for now i solved this by calling a > bash script from python which monitors the call once the UUID is passed to > it so that holds the python command till call is over and then it proceeds > further > > > Regards, > Bipin > > > ------------------------------ > -------- Original Message -------- > Subject: Re: [Freeswitch-users] return to shell only when call from > originate is complete? > From: Tihomir Culjaga > To: FreeSWITCH Users Help > > Date: 5/21/2017, 7:25:31 PM > > i still think you are better consume events with python... > its easy and straightforward > > > #!/usr/bin/env python > > import string > import sys > > from ESL import * > > con = ESLconnection("127.0.0.1","8021","ClueCon") > #are we connected? > callDirection = "unknown" > > > if con.connected: > print "we connected \n" > con.events("plain", "all"); > > while 1: > #my $e = $con->recvEventTimed(100); > e = con.recvEvent() > > if e: > #print e.serialize() > en = e.getHeader("Event-Name") > print "Name =>" ,en > > if en == "CHANNEL_OUTGOING": > callDirection = "OUT" > print "Direction =", callDirection > > elif en == "CHANNEL_ORIGINATE": > print "Originate call - direction =", callDirection > > elif en == "CHANNEL_CALLSTATE": > print "Call State =>", e.getHeader("Answer-State") > > elif en == "CHANNEL_ANSWER": > print "Call START - direction =", callDirection > > elif en == "CHANNEL_HANGUP": > print "Call END - direction =", callDirection > > elif en == "CHANNEL_PARK": > callDirection = "IN" > print "Call PARK - direction =", callDirection > > > > you can work it out from here pretty everything you want. > > > T. > > On 21 May 2017 at 14:32, Bipin Patel wrote: > >> hi, >> >> i converted the bash script to python as below: >> >> ret = subprocess.Popen('sudo fs_cli -x \"show channels\" | grep -c >> \"'+UUID+'\"', shell=True, stdout=subprocess.PIPE, stderr=subprocess.PIPE) >> sout, serr = ret.communicate() >> print sout >> >> the reason i use -c instead of -q is because fs_cli always returns a 0 >> exit code so it doesnt work through python when combined with grep so i >> tried to exit loop when the counter is 0 meaning the call is completed, but >> the above returns some weird numbers in sout, when the call is running it >> shows 5 and when done shows 3 but same if i run directly through console it >> shows accurately. >> >> any idea how to fix this? >> >> >> Regards, >> Bipin >> >> >> ------------------------------ >> -------- Original Message -------- >> Subject: Re: [Freeswitch-users] return to shell only when call from >> originate is complete? >> From: William Simon >> To: FreeSWITCH Users Help >> >> Date: 5/20/2017, 1:29:05 AM >> >> If you are using fs_cli then a simple shell script can monitor the call >> and return when the call completes: >> >> #!/bin/bash >> >> RESULT=$( fs_cli -x "originate leg1 leg2" ) >> >> if [[ $RESULT =~ "OK" ]]; then >> UUID=$( echo "$RESULT" | cut -f 2 -d ' ' ) >> while fs_cli -x "show channels" | grep -q $UUID ; >> do >> sleep 5; >> done >> echo "Call completed." >> fi >> >> >> >> >> On May 19, 2017, at 12:31 PM, Bipin Patel wrote: >> >> hi, >> >> any other way this can be done >> >> >> Regards, >> Bipin >> >> >> ------------------------------ >> -------- Original Message -------- >> Subject: Re: [Freeswitch-users] return to shell only when call from >> originate is complete? >> From: Michael Jerris >> To: Bipin Patel >> Date: 5/19/2017, 7:48:22 PM >> >> No, thats not what originate does. >> >> On May 19, 2017, at 11:43 AM, Bipin Patel wrote: >> >> hi, >> >> does it happen right after answer? >> is there any way to lock it so it returns only after failure or after >> hangup? yes im using ignore_early_media >> >> >> Regards, >> Bipin >> >> >> ------------------------------ >> -------- Original Message -------- >> Subject: Re: [Freeswitch-users] return to shell only when call from >> originate is complete? >> From: Michael Jerris >> To: FreeSWITCH Users Help >> >> Date: 5/19/2017, 7:34:35 PM >> >> originate returns when we get call progress, or in the case of >> ignore_early_media, on answer or failure. >> >> On May 19, 2017, at 10:36 AM, Bipin Patel wrote: >> >> hi, >> >> i was wondering if its possible to to release handle from an originate >> command only once the call is completed, i mean i send a originate command >> from shell using fs_cli but only return back to prompt once the call is >> completed rather than as soon as a bridge is created or a UUID generated >> >> >> >> >> >> >> “The information transmitted is intended only for the person or entity to >> which it is addressed and may contain proprietary, business-confidential >> and/or privileged material. If you are not the intended recipient of this >> message you are hereby notified that any use, review, retransmission, >> dissemination, distribution, reproduction or any action taken in reliance >> upon this message is prohibited. If you received this in error, please >> contact the sender and delete the material from any computer.” >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> “The information transmitted is intended only for the person or entity to >> which it is addressed and may contain proprietary, business-confidential >> and/or privileged material. If you are not the intended recipient of this >> message you are hereby notified that any use, review, retransmission, >> dissemination, distribution, reproduction or any action taken in reliance >> upon this message is prohibited. If you received this in error, please >> contact the sender and delete the material from any computer.” >> >> _________________________________________________________________________ >> 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 -- 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=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: From bipin at xbipin.com Mon May 22 06:10:17 2017 From: bipin at xbipin.com (Bipin Patel) Date: Mon, 22 May 2017 10:10:17 +0400 Subject: [Freeswitch-users] return to shell only when call from originate is complete? In-Reply-To: References: <69660425-2e51-1943-4fba-fc1ff5ea7094@xbipin.com> <4E8BF635-E985-4056-B301-A1EEFF702B09@jerris.com> <8eb5928b-d77f-eb70-fa0b-84f9d21f7d5b@xbipin.com> <4112C9C4-5BAA-408F-9731-8B11F3CCDAD8@stratusvideo.com> <0f62f615-0556-55aa-c52b-21955f56e1db@xbipin.com> Message-ID: hi, that was a good one, the reason im not using esl directly from python is because all i want to do is send originate command and get UUID and thats about it and stop the python script as soon as its done, ill try it once the requirements go higher with the client :) Regards, Bipin ------------------------------------------------------------------------ -------- Original Message -------- Subject: Re: [Freeswitch-users] return to shell only when call from originate is complete? From: Anthony Minessale To: FreeSWITCH Users Help Date: 5/22/2017, 8:38:13 AM > I am trying to catch a fish can you help me? > > Get a pole and use some fishing line with a hook and add a worm. > > I dont really want to use a pole, I am just going to throw a grenade > in the water and a fish will fall into my boat. > > > > > > > On Sun, May 21, 2017 at 10:43 AM Bipin Patel > wrote: > > hi, > > im trying to avoid going the esl route, for now i solved this by > calling a bash script from python which monitors the call once the > UUID is passed to it so that holds the python command till call is > over and then it proceeds further > > > Regards, > Bipin > > > ------------------------------------------------------------------------ > -------- Original Message -------- > Subject: Re: [Freeswitch-users] return to shell only when call > from originate is complete? > From: Tihomir Culjaga > To: FreeSWITCH Users Help > > Date: 5/21/2017, 7:25:31 PM >> i still think you are better consume events with python... >> its easy and straightforward >> >> >> #!/usr/bin/env python >> >> import string >> import sys >> >> from ESL import * >> >> con = ESLconnection("127.0.0.1","8021","ClueCon") >> #are we connected? >> callDirection = "unknown" >> >> >> if con.connected: >> print "we connected \n" >> con.events("plain", "all"); >> >> while 1: >> #my $e = $con->recvEventTimed(100); >> e = con.recvEvent() >> >> if e: >> #print e.serialize() >> en = e.getHeader("Event-Name") >> print "Name =>" ,en >> >> if en == "CHANNEL_OUTGOING": >> callDirection = "OUT" >> print "Direction =", callDirection >> >> elif en == "CHANNEL_ORIGINATE": >> print "Originate call - direction =", callDirection >> >> elif en == "CHANNEL_CALLSTATE": >> print "Call State =>", e.getHeader("Answer-State") >> >> elif en == "CHANNEL_ANSWER": >> print "Call START - direction =", callDirection >> >> elif en == "CHANNEL_HANGUP": >> print "Call END - direction =", callDirection >> >> elif en == "CHANNEL_PARK": >> callDirection = "IN" >> print "Call PARK - direction =", callDirection >> >> >> >> you can work it out from here pretty everything you want. >> >> >> T. >> >> On 21 May 2017 at 14:32, Bipin Patel > > wrote: >> >> hi, >> >> i converted the bash script to python as below: >> >> ret = subprocess.Popen('sudo fs_cli -x \"show channels\" | >> grep -c \"'+UUID+'\"', shell=True, stdout=subprocess.PIPE, >> stderr=subprocess.PIPE) >> sout, serr = ret.communicate() >> print sout >> >> the reason i use -c instead of -q is because fs_cli always >> returns a 0 exit code so it doesnt work through python when >> combined with grep so i tried to exit loop when the counter >> is 0 meaning the call is completed, but the above returns >> some weird numbers in sout, when the call is running it shows >> 5 and when done shows 3 but same if i run directly through >> console it shows accurately. >> >> any idea how to fix this? >> >> >> Regards, >> Bipin >> >> >> ------------------------------------------------------------------------ >> -------- Original Message -------- >> Subject: Re: [Freeswitch-users] return to shell only when >> call from originate is complete? >> From: William Simon >> >> To: FreeSWITCH Users Help >> >> >> Date: 5/20/2017, 1:29:05 AM >>> If you are using fs_cli then a simple shell script can >>> monitor the call and return when the call completes: >>> >>> #!/bin/bash >>> >>> RESULT=$( fs_cli -x "originate leg1 leg2" ) >>> >>> if [[ $RESULT =~ "OK" ]]; then >>> UUID=$( echo "$RESULT" | cut -f 2 -d ' ' ) >>> while fs_cli -x "show channels" | grep -q $UUID ; >>> do >>> sleep 5; >>> done >>> echo "Call completed." >>> fi >>> >>> >>> >>> >>>> On May 19, 2017, at 12:31 PM, Bipin Patel >>> > wrote: >>>> >>>> hi, >>>> >>>> any other way this can be done >>>> >>>> >>>> Regards, >>>> Bipin >>>> >>>> >>>> ------------------------------------------------------------------------ >>>> -------- Original Message -------- >>>> Subject: Re: [Freeswitch-users] return to shell only when >>>> call from originate is complete? >>>> From: Michael Jerris >>>> To: Bipin Patel >>>> Date: 5/19/2017, 7:48:22 PM >>>>> No, thats not what originate does. >>>>> >>>>>> On May 19, 2017, at 11:43 AM, Bipin Patel >>>>>> > wrote: >>>>>> >>>>>> hi, >>>>>> >>>>>> does it happen right after answer? >>>>>> is there any way to lock it so it returns only after >>>>>> failure or after hangup? yes im using ignore_early_media >>>>>> >>>>>> >>>>>> Regards, >>>>>> Bipin >>>>>> >>>>>> >>>>>> ------------------------------------------------------------------------ >>>>>> -------- Original Message -------- >>>>>> Subject: Re: [Freeswitch-users] return to shell only when >>>>>> call from originate is complete? >>>>>> From: Michael Jerris >>>>>> >>>>>> To: FreeSWITCH Users >>>>>> Help >>>>>> >>>>>> Date: 5/19/2017, 7:34:35 PM >>>>>>> originate returns when we get call progress, or in the >>>>>>> case of ignore_early_media, on answer or failure. >>>>>>> >>>>>>>> On May 19, 2017, at 10:36 AM, Bipin Patel >>>>>>>> > wrote: >>>>>>>> >>>>>>>> hi, >>>>>>>> >>>>>>>> i was wondering if its possible to to release handle >>>>>>>> from an originate command only once the call is >>>>>>>> completed, i mean i send a originate command from shell >>>>>>>> using fs_cli but only return back to prompt once the >>>>>>>> call is completed rather than as soon as a bridge is >>>>>>>> created or a UUID generated >>>>>>>> >>>>>>> >>>>> >>>> >>>> >>>> >>>> “The information transmitted is intended only for the >>>> person or entity to which it is addressed and may contain >>>> proprietary, business-confidential and/or privileged >>>> material. If you are not the intended recipient of this >>>> message you are hereby notified that any use, review, >>>> retransmission, dissemination, distribution, reproduction >>>> or any action taken in reliance upon this message is >>>> prohibited. If you received this in error, please contact >>>> the sender and delete the material from any computer.” >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> >>> “The information transmitted is intended only for the person >>> or entity to which it is addressed and may contain >>> proprietary, business-confidential and/or privileged >>> material. If you are not the intended recipient of this >>> message you are hereby notified that any use, review, >>> retransmission, dissemination, distribution, reproduction or >>> any action taken in reliance upon this message is >>> prohibited. If you received this in error, please contact >>> the sender and delete the material from any computer.” >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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=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 Mon May 22 06:33:53 2017 From: bipin at xbipin.com (Bipin Patel) Date: Mon, 22 May 2017 10:33:53 +0400 Subject: [Freeswitch-users] return to shell only when call from originate is complete? In-Reply-To: References: <69660425-2e51-1943-4fba-fc1ff5ea7094@xbipin.com> <4E8BF635-E985-4056-B301-A1EEFF702B09@jerris.com> <8eb5928b-d77f-eb70-fa0b-84f9d21f7d5b@xbipin.com> <4112C9C4-5BAA-408F-9731-8B11F3CCDAD8@stratusvideo.com> <0f62f615-0556-55aa-c52b-21955f56e1db@xbipin.com> Message-ID: <1edfad2b-dbe1-837b-408b-1bcf8e6df6c7@xbipin.com> hi, btw is it possible to use python with esl on windows, i cant seem to find any guide to it Regards, Bipin ------------------------------------------------------------------------ -------- Original Message -------- Subject: Re: [Freeswitch-users] return to shell only when call from originate is complete? From: Bipin Patel To: FreeSWITCH Users Help Date: 5/22/2017, 10:10:17 AM > hi, > > that was a good one, the reason im not using esl directly from python > is because all i want to do is send originate command and get UUID and > thats about it and stop the python script as soon as its done, ill try > it once the requirements go higher with the client :) > > > Regards, > Bipin > > > ------------------------------------------------------------------------ > -------- Original Message -------- > Subject: Re: [Freeswitch-users] return to shell only when call from > originate is complete? > From: Anthony Minessale > To: FreeSWITCH Users Help > Date: 5/22/2017, 8:38:13 AM >> I am trying to catch a fish can you help me? >> >> Get a pole and use some fishing line with a hook and add a worm. >> >> I dont really want to use a pole, I am just going to throw a grenade >> in the water and a fish will fall into my boat. >> >> >> >> >> >> >> On Sun, May 21, 2017 at 10:43 AM Bipin Patel > > wrote: >> >> hi, >> >> im trying to avoid going the esl route, for now i solved this by >> calling a bash script from python which monitors the call once >> the UUID is passed to it so that holds the python command till >> call is over and then it proceeds further >> >> >> Regards, >> Bipin >> >> >> ------------------------------------------------------------------------ >> -------- Original Message -------- >> Subject: Re: [Freeswitch-users] return to shell only when call >> from originate is complete? >> From: Tihomir Culjaga >> >> To: FreeSWITCH Users Help >> >> Date: 5/21/2017, 7:25:31 PM >>> i still think you are better consume events with python... >>> its easy and straightforward >>> >>> >>> #!/usr/bin/env python >>> >>> import string >>> import sys >>> >>> from ESL import * >>> >>> con = ESLconnection("127.0.0.1","8021","ClueCon") >>> #are we connected? >>> callDirection = "unknown" >>> >>> >>> if con.connected: >>> print "we connected \n" >>> con.events("plain", "all"); >>> >>> while 1: >>> #my $e = $con->recvEventTimed(100); >>> e = con.recvEvent() >>> >>> if e: >>> #print e.serialize() >>> en = e.getHeader("Event-Name") >>> print "Name =>" ,en >>> >>> if en == "CHANNEL_OUTGOING": >>> callDirection = "OUT" >>> print "Direction =", callDirection >>> >>> elif en == "CHANNEL_ORIGINATE": >>> print "Originate call - direction =", callDirection >>> >>> elif en == "CHANNEL_CALLSTATE": >>> print "Call State =>", e.getHeader("Answer-State") >>> >>> elif en == "CHANNEL_ANSWER": >>> print "Call START - direction =", callDirection >>> >>> elif en == "CHANNEL_HANGUP": >>> print "Call END - direction =", callDirection >>> >>> elif en == "CHANNEL_PARK": >>> callDirection = "IN" >>> print "Call PARK - direction =", callDirection >>> >>> >>> >>> you can work it out from here pretty everything you want. >>> >>> >>> T. >>> >>> On 21 May 2017 at 14:32, Bipin Patel >> > wrote: >>> >>> hi, >>> >>> i converted the bash script to python as below: >>> >>> ret = subprocess.Popen('sudo fs_cli -x \"show channels\" | >>> grep -c \"'+UUID+'\"', shell=True, stdout=subprocess.PIPE, >>> stderr=subprocess.PIPE) >>> sout, serr = ret.communicate() >>> print sout >>> >>> the reason i use -c instead of -q is because fs_cli always >>> returns a 0 exit code so it doesnt work through python when >>> combined with grep so i tried to exit loop when the counter >>> is 0 meaning the call is completed, but the above returns >>> some weird numbers in sout, when the call is running it >>> shows 5 and when done shows 3 but same if i run directly >>> through console it shows accurately. >>> >>> any idea how to fix this? >>> >>> >>> Regards, >>> Bipin >>> >>> >>> ------------------------------------------------------------------------ >>> -------- Original Message -------- >>> Subject: Re: [Freeswitch-users] return to shell only when >>> call from originate is complete? >>> From: William Simon >>> >>> To: FreeSWITCH Users Help >>> >>> >>> Date: 5/20/2017, 1:29:05 AM >>>> If you are using fs_cli then a simple shell script can >>>> monitor the call and return when the call completes: >>>> >>>> #!/bin/bash >>>> >>>> RESULT=$( fs_cli -x "originate leg1 leg2" ) >>>> >>>> if [[ $RESULT =~ "OK" ]]; then >>>> UUID=$( echo "$RESULT" | cut -f 2 -d ' ' ) >>>> while fs_cli -x "show channels" | grep -q $UUID ; >>>> do >>>> sleep 5; >>>> done >>>> echo "Call completed." >>>> fi >>>> >>>> >>>> >>>> >>>>> On May 19, 2017, at 12:31 PM, Bipin Patel >>>>> > wrote: >>>>> >>>>> hi, >>>>> >>>>> any other way this can be done >>>>> >>>>> >>>>> Regards, >>>>> Bipin >>>>> >>>>> >>>>> ------------------------------------------------------------------------ >>>>> -------- Original Message -------- >>>>> Subject: Re: [Freeswitch-users] return to shell only when >>>>> call from originate is complete? >>>>> From: Michael Jerris >>>>> To: Bipin Patel >>>>> Date: 5/19/2017, 7:48:22 PM >>>>>> No, thats not what originate does. >>>>>> >>>>>>> On May 19, 2017, at 11:43 AM, Bipin Patel >>>>>>> > wrote: >>>>>>> >>>>>>> hi, >>>>>>> >>>>>>> does it happen right after answer? >>>>>>> is there any way to lock it so it returns only after >>>>>>> failure or after hangup? yes im using ignore_early_media >>>>>>> >>>>>>> >>>>>>> Regards, >>>>>>> Bipin >>>>>>> >>>>>>> >>>>>>> ------------------------------------------------------------------------ >>>>>>> -------- Original Message -------- >>>>>>> Subject: Re: [Freeswitch-users] return to shell only >>>>>>> when call from originate is complete? >>>>>>> From: Michael Jerris >>>>>>> >>>>>>> To: FreeSWITCH Users >>>>>>> Help >>>>>>> >>>>>>> Date: 5/19/2017, 7:34:35 PM >>>>>>>> originate returns when we get call progress, or in the >>>>>>>> case of ignore_early_media, on answer or failure. >>>>>>>> >>>>>>>>> On May 19, 2017, at 10:36 AM, Bipin Patel >>>>>>>>> > wrote: >>>>>>>>> >>>>>>>>> hi, >>>>>>>>> >>>>>>>>> i was wondering if its possible to to release handle >>>>>>>>> from an originate command only once the call is >>>>>>>>> completed, i mean i send a originate command from >>>>>>>>> shell using fs_cli but only return back to prompt once >>>>>>>>> the call is completed rather than as soon as a bridge >>>>>>>>> is created or a UUID generated >>>>>>>>> >>>>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> “The information transmitted is intended only for the >>>>> person or entity to which it is addressed and may contain >>>>> proprietary, business-confidential and/or privileged >>>>> material. If you are not the intended recipient of this >>>>> message you are hereby notified that any use, review, >>>>> retransmission, dissemination, distribution, reproduction >>>>> or any action taken in reliance upon this message is >>>>> prohibited. If you received this in error, please contact >>>>> the sender and delete the material from any computer.” >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> “The information transmitted is intended only for the >>>> person or entity to which it is addressed and may contain >>>> proprietary, business-confidential and/or privileged >>>> material. If you are not the intended recipient of this >>>> message you are hereby notified that any use, review, >>>> retransmission, dissemination, distribution, reproduction >>>> or any action taken in reliance upon this message is >>>> prohibited. If you received this in error, please contact >>>> the sender and delete the material from any computer.” >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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=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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 Mon May 22 08:12:48 2017 From: khorsmann at gmail.com (Karsten Horsmann) Date: Mon, 22 May 2017 10:12:48 +0200 Subject: [Freeswitch-users] freeswitch segfault at sp error 4 in libmysqlclient.so.18.0.0 Message-ID: Hello, i am a littlebit confused about an error that my FreeSWITCH 1.6.17 Testing Box has. Its an vanilla rpm installation for CentOS 7.3.11. After a while it stops working. The only thing mysql-based i do is an lua-script that use lua-dbi for mysql. Maybe this script kills the prozess. How can i debug that? Should i fill an JIRA? FreeSWITCH Version 1.6.17 In the logfile i see this error messages: May 20 17:40:50 fs16dev systemd: Cannot add dependency job for unit microcode.service, ignoring: Unit is not loaded properly: Invalid argument. May 20 17:40:50 fs16dev systemd: Started Check_MK (172.20.160.60:41436). May 20 17:40:50 fs16dev systemd: Starting Check_MK (172.20.160.60:41436)... May 20 17:41:18 fs16dev kernel: freeswitch[5603]: segfault at 0 ip 00007f8946fb2e5f sp 00007f894773f240 error 4 in libmysqlclient.so.18.0.0[7f8946f8b000+2de000] May 20 17:41:18 fs16dev systemd: freeswitch.service: main process exited, code=killed, status=11/SEGV May 20 17:41:18 fs16dev systemd: Unit freeswitch.service entered failed state. May 20 17:41:18 fs16dev systemd: freeswitch.service failed. May 20 17:41:18 fs16dev systemd: freeswitch.service holdoff time over, scheduling restart. May 20 17:41:18 fs16dev systemd: Cannot add dependency job for unit microcode.service, ignoring: Unit is not loaded properly: Invalid argument. May 20 17:41:18 fs16dev systemd: Starting freeswitch... May 20 17:41:18 fs16dev freeswitch: 5610 Backgrounding. May 20 17:41:20 fs16dev freeswitch: FreeSWITCH[5609] Waiting for background process pid:5610 to be ready..... May 20 17:41:20 fs16dev freeswitch: FreeSWITCH[5609] System Ready pid:5610 May 20 17:41:20 fs16dev systemd: Started freeswitch. BTW - the rpm systemd service didnt restart the server if it breaks. I used an debian style version of the freeswitch stash. Kind Regards -- Mit freundlichen Grüßen *Karsten Horsmann* -------------- next part -------------- An HTML attachment was scrubbed... URL: From bipin at xbipin.com Mon May 22 10:21:24 2017 From: bipin at xbipin.com (Bipin Patel) Date: Mon, 22 May 2017 14:21:24 +0400 Subject: [Freeswitch-users] return to shell only when call from originate is complete? In-Reply-To: <1edfad2b-dbe1-837b-408b-1bcf8e6df6c7@xbipin.com> References: <69660425-2e51-1943-4fba-fc1ff5ea7094@xbipin.com> <4E8BF635-E985-4056-B301-A1EEFF702B09@jerris.com> <8eb5928b-d77f-eb70-fa0b-84f9d21f7d5b@xbipin.com> <4112C9C4-5BAA-408F-9731-8B11F3CCDAD8@stratusvideo.com> <0f62f615-0556-55aa-c52b-21955f56e1db@xbipin.com> <1edfad2b-dbe1-837b-408b-1bcf8e6df6c7@xbipin.com> Message-ID: <322d2b9b-e2e1-8ee2-cfa8-c8b825d832a9@xbipin.com> hi, i tried to install the python-ESL module using pip but it wont compile even with swig, complaining about sys/time.h Regards, Bipin ------------------------------------------------------------------------ -------- Original Message -------- Subject: Re: [Freeswitch-users] return to shell only when call from originate is complete? From: Bipin Patel To: FreeSWITCH Users Help Date: 5/22/2017, 10:33:53 AM > hi, > > btw is it possible to use python with esl on windows, i cant seem to > find any guide to it > > > Regards, > Bipin > > > ------------------------------------------------------------------------ > -------- Original Message -------- > Subject: Re: [Freeswitch-users] return to shell only when call from > originate is complete? > From: Bipin Patel > To: FreeSWITCH Users Help > Date: 5/22/2017, 10:10:17 AM >> hi, >> >> that was a good one, the reason im not using esl directly from python >> is because all i want to do is send originate command and get UUID >> and thats about it and stop the python script as soon as its done, >> ill try it once the requirements go higher with the client :) >> >> >> Regards, >> Bipin >> >> >> ------------------------------------------------------------------------ >> -------- Original Message -------- >> Subject: Re: [Freeswitch-users] return to shell only when call from >> originate is complete? >> From: Anthony Minessale >> To: FreeSWITCH Users Help >> Date: 5/22/2017, 8:38:13 AM >>> I am trying to catch a fish can you help me? >>> >>> Get a pole and use some fishing line with a hook and add a worm. >>> >>> I dont really want to use a pole, I am just going to throw a grenade >>> in the water and a fish will fall into my boat. >>> >>> >>> >>> >>> >>> >>> On Sun, May 21, 2017 at 10:43 AM Bipin Patel >> > wrote: >>> >>> hi, >>> >>> im trying to avoid going the esl route, for now i solved this by >>> calling a bash script from python which monitors the call once >>> the UUID is passed to it so that holds the python command till >>> call is over and then it proceeds further >>> >>> >>> Regards, >>> Bipin >>> >>> >>> ------------------------------------------------------------------------ >>> -------- Original Message -------- >>> Subject: Re: [Freeswitch-users] return to shell only when call >>> from originate is complete? >>> From: Tihomir Culjaga >>> >>> To: FreeSWITCH Users Help >>> >>> >>> Date: 5/21/2017, 7:25:31 PM >>>> i still think you are better consume events with python... >>>> its easy and straightforward >>>> >>>> >>>> #!/usr/bin/env python >>>> >>>> import string >>>> import sys >>>> >>>> from ESL import * >>>> >>>> con = ESLconnection("127.0.0.1","8021","ClueCon") >>>> #are we connected? >>>> callDirection = "unknown" >>>> >>>> >>>> if con.connected: >>>> print "we connected \n" >>>> con.events("plain", "all"); >>>> >>>> while 1: >>>> #my $e = $con->recvEventTimed(100); >>>> e = con.recvEvent() >>>> >>>> if e: >>>> #print e.serialize() >>>> en = e.getHeader("Event-Name") >>>> print "Name =>" ,en >>>> >>>> if en == "CHANNEL_OUTGOING": >>>> callDirection = "OUT" >>>> print "Direction =", callDirection >>>> >>>> elif en == "CHANNEL_ORIGINATE": >>>> print "Originate call - direction =", callDirection >>>> >>>> elif en == "CHANNEL_CALLSTATE": >>>> print "Call State =>", e.getHeader("Answer-State") >>>> >>>> elif en == "CHANNEL_ANSWER": >>>> print "Call START - direction =", callDirection >>>> >>>> elif en == "CHANNEL_HANGUP": >>>> print "Call END - direction =", callDirection >>>> >>>> elif en == "CHANNEL_PARK": >>>> callDirection = "IN" >>>> print "Call PARK - direction =", callDirection >>>> >>>> >>>> >>>> you can work it out from here pretty everything you want. >>>> >>>> >>>> T. >>>> >>>> On 21 May 2017 at 14:32, Bipin Patel >>> > wrote: >>>> >>>> hi, >>>> >>>> i converted the bash script to python as below: >>>> >>>> ret = subprocess.Popen('sudo fs_cli -x \"show channels\" | >>>> grep -c \"'+UUID+'\"', shell=True, stdout=subprocess.PIPE, >>>> stderr=subprocess.PIPE) >>>> sout, serr = ret.communicate() >>>> print sout >>>> >>>> the reason i use -c instead of -q is because fs_cli always >>>> returns a 0 exit code so it doesnt work through python when >>>> combined with grep so i tried to exit loop when the counter >>>> is 0 meaning the call is completed, but the above returns >>>> some weird numbers in sout, when the call is running it >>>> shows 5 and when done shows 3 but same if i run directly >>>> through console it shows accurately. >>>> >>>> any idea how to fix this? >>>> >>>> >>>> Regards, >>>> Bipin >>>> >>>> >>>> ------------------------------------------------------------------------ >>>> -------- Original Message -------- >>>> Subject: Re: [Freeswitch-users] return to shell only when >>>> call from originate is complete? >>>> From: William Simon >>>> >>>> To: FreeSWITCH Users Help >>>> >>>> >>>> Date: 5/20/2017, 1:29:05 AM >>>>> If you are using fs_cli then a simple shell script can >>>>> monitor the call and return when the call completes: >>>>> >>>>> #!/bin/bash >>>>> >>>>> RESULT=$( fs_cli -x "originate leg1 leg2" ) >>>>> >>>>> if [[ $RESULT =~ "OK" ]]; then >>>>> UUID=$( echo "$RESULT" | cut -f 2 -d ' ' ) >>>>> while fs_cli -x "show channels" | grep -q $UUID ; >>>>> do >>>>> sleep 5; >>>>> done >>>>> echo "Call completed." >>>>> fi >>>>> >>>>> >>>>> >>>>> >>>>>> On May 19, 2017, at 12:31 PM, Bipin Patel >>>>>> > wrote: >>>>>> >>>>>> hi, >>>>>> >>>>>> any other way this can be done >>>>>> >>>>>> >>>>>> Regards, >>>>>> Bipin >>>>>> >>>>>> >>>>>> ------------------------------------------------------------------------ >>>>>> -------- Original Message -------- >>>>>> Subject: Re: [Freeswitch-users] return to shell only when >>>>>> call from originate is complete? >>>>>> From: Michael Jerris >>>>>> >>>>>> To: Bipin Patel >>>>>> Date: 5/19/2017, 7:48:22 PM >>>>>>> No, thats not what originate does. >>>>>>> >>>>>>>> On May 19, 2017, at 11:43 AM, Bipin Patel >>>>>>>> > wrote: >>>>>>>> >>>>>>>> hi, >>>>>>>> >>>>>>>> does it happen right after answer? >>>>>>>> is there any way to lock it so it returns only after >>>>>>>> failure or after hangup? yes im using ignore_early_media >>>>>>>> >>>>>>>> >>>>>>>> Regards, >>>>>>>> Bipin >>>>>>>> >>>>>>>> >>>>>>>> ------------------------------------------------------------------------ >>>>>>>> -------- Original Message -------- >>>>>>>> Subject: Re: [Freeswitch-users] return to shell only >>>>>>>> when call from originate is complete? >>>>>>>> From: Michael Jerris >>>>>>>> >>>>>>>> To: FreeSWITCH Users >>>>>>>> Help >>>>>>>> >>>>>>>> Date: 5/19/2017, 7:34:35 PM >>>>>>>>> originate returns when we get call progress, or in the >>>>>>>>> case of ignore_early_media, on answer or failure. >>>>>>>>> >>>>>>>>>> On May 19, 2017, at 10:36 AM, Bipin Patel >>>>>>>>>> > wrote: >>>>>>>>>> >>>>>>>>>> hi, >>>>>>>>>> >>>>>>>>>> i was wondering if its possible to to release handle >>>>>>>>>> from an originate command only once the call is >>>>>>>>>> completed, i mean i send a originate command from >>>>>>>>>> shell using fs_cli but only return back to prompt >>>>>>>>>> once the call is completed rather than as soon as a >>>>>>>>>> bridge is created or a UUID generated >>>>>>>>>> >>>>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> >>>>>> “The information transmitted is intended only for the >>>>>> person or entity to which it is addressed and may contain >>>>>> proprietary, business-confidential and/or privileged >>>>>> material. If you are not the intended recipient of this >>>>>> message you are hereby notified that any use, review, >>>>>> retransmission, dissemination, distribution, reproduction >>>>>> or any action taken in reliance upon this message is >>>>>> prohibited. If you received this in error, please contact >>>>>> the sender and delete the material from any computer.” >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> “The information transmitted is intended only for the >>>>> person or entity to which it is addressed and may contain >>>>> proprietary, business-confidential and/or privileged >>>>> material. If you are not the intended recipient of this >>>>> message you are hereby notified that any use, review, >>>>> retransmission, dissemination, distribution, reproduction >>>>> or any action taken in reliance upon this message is >>>>> prohibited. If you received this in error, please contact >>>>> the sender and delete the material from any computer.” >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at 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=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 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 loi.dangthanh at gmail.com Mon May 22 10:49:33 2017 From: loi.dangthanh at gmail.com (=?UTF-8?B?TOG7o2kgxJDhurduZw==?=) Date: Mon, 22 May 2017 10:49:33 +0000 Subject: [Freeswitch-users] About proxy-hold codec re-negotiation behavior. Message-ID: Hi list, I'm using FS with *proxy-hold*. Call flow is simple A -> FreeSWITCH -> B Assume FS, A and B support PCMA and PCMU, FS prefer PCMA in codec-prefs profile configuration. In affects of my configuration for *greedy early negotiation* and *disable transcoding*, PCMA is negotiated in both legs for initial INVITE, that's good, and expected. But then I remove PCMA from A and compose a re-INVITE for holding with PCMU only, the a leg is re-negotiated with PCMU, but the b leg have re-INVITE with PCMA due to *proxy-hold* variable, that causes transcoding happened after hold. My desire is to have FS to re-negotiate with b leg too, not so similar but as what I observed when I was using it with media proxy option ( A-A before hold, U-U after hold). I tried late negotiation in this case, but no luck. So is that not able for FS to re-negotiate codec on b leg, in using proxy-hold? Any advise is appreciated. rgds, Loi Dang -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Mon May 22 15:30:12 2017 From: mike at jerris.com (Michael Jerris) Date: Mon, 22 May 2017 11:30:12 -0400 Subject: [Freeswitch-users] freeswitch segfault at sp error 4 in libmysqlclient.so.18.0.0 In-Reply-To: References: Message-ID: <44FC56B9-1094-4F21-BD1A-760D820EFE7C@jerris.com> you would have to look at a backtrace of the crash. If the issue is from lua mysql, you should try to reproduce that standalone with lua and mysql, and report the issue to your distro. > On May 22, 2017, at 4:12 AM, Karsten Horsmann wrote: > > Hello, > > i am a littlebit confused about an error that my FreeSWITCH 1.6.17 Testing Box has. > Its an vanilla rpm installation for CentOS 7.3.11. After a while it stops working. > > The only thing mysql-based i do is an lua-script that use lua-dbi for mysql. > Maybe this script kills the prozess. How can i debug that? Should i fill an JIRA? > > FreeSWITCH Version 1.6.17 > > In the logfile i see this error messages: > > > May 20 17:40:50 fs16dev systemd: Cannot add dependency job for unit microcode.service, ignoring: Unit is not loaded properly: Invalid argument. > May 20 17:40:50 fs16dev systemd: Started Check_MK (172.20.160.60:41436 ). > May 20 17:40:50 fs16dev systemd: Starting Check_MK (172.20.160.60:41436)... > May 20 17:41:18 fs16dev kernel: freeswitch[5603]: segfault at 0 ip 00007f8946fb2e5f sp 00007f894773f240 error 4 in libmysqlclient.so.18.0.0[7f8946f8b000+2de000] > May 20 17:41:18 fs16dev systemd: freeswitch.service: main process exited, code=killed, status=11/SEGV > May 20 17:41:18 fs16dev systemd: Unit freeswitch.service entered failed state. > May 20 17:41:18 fs16dev systemd: freeswitch.service failed. > May 20 17:41:18 fs16dev systemd: freeswitch.service holdoff time over, scheduling restart. > May 20 17:41:18 fs16dev systemd: Cannot add dependency job for unit microcode.service, ignoring: Unit is not loaded properly: Invalid argument. > May 20 17:41:18 fs16dev systemd: Starting freeswitch... > May 20 17:41:18 fs16dev freeswitch: 5610 Backgrounding. > May 20 17:41:20 fs16dev freeswitch: FreeSWITCH[5609] Waiting for background process pid:5610 to be ready..... > May 20 17:41:20 fs16dev freeswitch: FreeSWITCH[5609] System Ready pid:5610 > May 20 17:41:20 fs16dev systemd: Started freeswitch. > > BTW - the rpm systemd service didnt restart the server if it breaks. > I used an debian style version of the freeswitch stash. > > > Kind Regards > -- > Mit freundlichen Grüßen > *Karsten Horsmann* -------------- next part -------------- An HTML attachment was scrubbed... URL: From babak.freeswitch at gmail.com Tue May 23 06:17:11 2017 From: babak.freeswitch at gmail.com (Babak Yakhchali) Date: Tue, 23 May 2017 10:47:11 +0430 Subject: [Freeswitch-users] Call setup failes on outbound calls In-Reply-To: References: <51AAF734-6B60-4553-82FA-6718F28F6FED@vallimamod.org> <27AFE1E9-58FE-4F2B-B416-CB77F85B840A@vallimamod.org> <1F8AE859-2FA3-4613-BB2A-783F2224E449@vallimamod.org> Message-ID: Hi After hours of testing I found that the problem is related to codecs! I set the codec to only PCMA and everything works :) I donno why. I hope this is useful for someone else. On Mon, Apr 10, 2017 at 9:36 AM, Babak Yakhchali wrote: > I switched to Freeswitch version1.2.24+git~20150504T151740Z~da04da87df~64bit > and now everything is working!!!! > . I think the problem is with this line in SDP: > *a=rtpmap:0 PCMU/8000* > > This is the working invite > > INVITE sip:09127006837 at 10.10.10.3 SIP/2.0 > Via: SIP/2.0/UDP 10.10.10.4;rport;branch=z9hG4bKUDaHF6g2ZcSZm > Max-Forwards: 69 > From: "100" ;tag=FF2ryU4pDQ7ac > To: > Call-ID: 788f5327-98ad-1235-9fa7-00505698455d > CSeq: 105584769 INVITE > Contact: > User-Agent: FreeSWITCH > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, > REGISTER, REFER, NOTIFY > Supported: timer, path, replaces > Allow-Events: talk, hold, conference, refer > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 201 > X-FS-Support: update_display,send_info > Remote-Party-ID: "100" ;party= > calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 1491809308 1491809309 IN IP4 10.10.10.4 > s=FreeSWITCH > c=IN IP4 10.10.10.4 > t=0 0 > m=audio 32358 RTP/AVP 0 8 101 13 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > > > On Sat, Apr 8, 2017 at 4:40 PM, Vallimamod Abdullah > wrote: > >> Then you should definitely check on the SoftX3000 side. >> >> Best Regards, >> -- >> Vallimamod Abdullah >> SIP Solutions >> vma at sipsolutions.fr >> . >> >> > On 8 Apr 2017, at 13:47, Babak Yakhchali >> wrote: >> > >> > tried it.... but it seems the problem is not related to silence >> suppression >> > >> > I donno why this is hapening ... >> > >> > Warning: 399 - "SoftX3000 R601-CCU Rel POS:[3103] Release from CR" >> > Reason: Q.850;cause=98;text="Message not compatible with call state >> or message type non-existent or not implemented" >> > >> > >> > On Sat, Apr 8, 2017 at 3:34 PM, Vallimamod Abdullah >> wrote: >> > Hi, >> > >> > After a look in the source code, it seems that it is either CN or >> silenceSupp:off, no other choice as the latter is pretty standard. >> > If you are compiling from sources, you can make a quick patch by >> commenting out the 2 lines with "a=silenceSupp:off" in switch_core_media.c >> (not tested.) >> > >> > Best Regards, >> > -- >> > Vallimamod Abdullah >> > SIP Solutions >> > vma at sipsolutions.fr >> > . >> > >> > > On 8 Apr 2017, at 12:26, Babak Yakhchali >> wrote: >> > > >> > > thank >> > > doing so rsults in : >> > > a=fmtp:101 0-16 >> > > a=silenceSupp:off - - - - >> > > a=ptime:20 >> > > it seems SoftX3000 has problems with any silence related config.. I >> need a way to remove any silence related parameter in sdp >> > > >> > > >> > > On Sat, Apr 8, 2017 at 2:48 PM, Vallimamod Abdullah < >> vma at vallimamod.org> wrote: >> > > Hi, >> > > >> > > It is for comfort noise generation. You can set suppress_cng channel >> var or suppress-cng profile param to true to disable it. >> > > >> > > Best Regards, >> > > -- >> > > Vallimamod Abdullah >> > > SIP Solutions >> > > vma at sipsolutions.fr >> > > . >> > > >> > > > On 8 Apr 2017, at 12:08, Babak Yakhchali < >> babak.freeswitch at gmail.com> wrote: >> > > > >> > > > As I searched more it seems the problem is with this line: >> > > > a=rtpmap:13 CN/8000 >> > > > how can I remove this? >> > > > >> > > > On Sat, Apr 8, 2017 at 1:48 PM, Babak Yakhchali < >> babak.freeswitch at gmail.com> wrote: >> > > > Hi >> > > > I'm trying to send calls through a provider with ip 10.10.10.3 but >> call fails with result "Message not compatible with call state or message >> type non-existent or not implemented". how can I debug this problem? >> > > > this is siptrace from the failed call: >> > > > >> > > > send 1061 bytes to udp/[10.10.10.3]:5060 at 13:35:27.382378: >> > > > ------------------------------------------------------------ >> ------------ >> > > > INVITE sip:12345678 at 10.10.10.3 SIP/2.0 >> > > > Via: SIP/2.0/UDP 10.10.10.4;rport;branch=z9hG4bKeN5gp5S5KrZce >> > > > Max-Forwards: 69 >> > > > From: "100" ;tag=N69FS63aHU7ta >> > > > To: >> > > > Call-ID: 596dba35-96dd-1235-59bc-00505698455d >> > > > CSeq: 105485099 INVITE >> > > > Contact: > 0;transport=udp;gw=trunk_1> >> > > > User-Agent: FreeSWITCH-mod_sofia/1.6.15+gi >> t~20170303T233525Z~02c0860de6~64bit >> > > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, >> REGISTER, REFER, NOTIFY >> > > > Supported: timer, path, replaces >> > > > Allow-Events: talk, hold, conference, refer >> > > > Content-Type: application/sdp >> > > > Content-Disposition: session >> > > > Content-Length: 242 >> > > > X-FS-Support: update_display,send_info >> > > > Remote-Party-ID: "100" ;party=cal >> ling;screen=yes;privacy=off >> > > > >> > > > v=0 >> > > > o=FreeSWITCH 1491655976 1491655977 IN IP4 10.10.10.4 >> > > > s=FreeSWITCH >> > > > c=IN IP4 10.10.10.4 >> > > > t=0 0 >> > > > m=audio 27714 RTP/AVP 0 101 13 >> > > > a=rtpmap:0 PCMU/8000 >> > > > a=rtpmap:101 telephone-event/8000 >> > > > a=fmtp:101 0-16 >> > > > a=rtpmap:13 CN/8000 >> > > > a=ptime:20 >> > > > ------------------------------------------------------------ >> ------------ >> > > > recv 274 bytes from udp/[10.10.10.3]:5060 at 13:35:27.398407: >> > > > ------------------------------------------------------------ >> ------------ >> > > > SIP/2.0 100 Trying >> > > > Via: SIP/2.0/UDP 10.10.10.4;branch=z9hG4bKeN5gp >> 5S5KrZce;rport=5060 >> > > > Call-ID: 596dba35-96dd-1235-59bc-00505698455d >> > > > From: "100";tag=N69FS63aHU7ta >> > > > To: >> > > > CSeq: 105485099 INVITE >> > > > Content-Length: 0 >> > > > >> > > > ------------------------------------------------------------ >> ------------ >> > > > recv 488 bytes from udp/[10.10.10.3]:5060 at 13:35:27.448335: >> > > > ------------------------------------------------------------ >> ------------ >> > > > SIP/2.0 500 Server Internal Error >> > > > Via: SIP/2.0/UDP 10.10.10.4;branch=z9hG4bKeN5gp >> 5S5KrZce;rport=5060 >> > > > Call-ID: 596dba35-96dd-1235-59bc-00505698455d >> > > > From: "100";tag=N69FS63aHU7ta >> > > > To: ;tag=juiky7nh >> > > > CSeq: 105485099 INVITE >> > > > Warning: 399 - "SoftX3000 R601-CCU Rel POS:[3103] Release from >> CR" >> > > > Reason: Q.850;cause=98;text="Message not compatible with call >> state or message type non-existent or not implemented" >> > > > Content-Length: 0 >> > > > >> > > > ------------------------------------------------------------ >> ------------ >> > > > send 321 bytes to udp/[10.10.10.3]:5060 at 13:35:27.448500: >> > > > ------------------------------------------------------------ >> ------------ >> > > > ACK sip:12345678 at 10.10.10.3 SIP/2.0 >> > > > Via: SIP/2.0/UDP 10.10.10.4;rport;branch=z9hG4bKeN5gp5S5KrZce >> > > > Max-Forwards: 69 >> > > > From: "100" ;tag=N69FS63aHU7ta >> > > > To: ;tag=juiky7nh >> > > > Call-ID: 596dba35-96dd-1235-59bc-00505698455d >> > > > CSeq: 105485099 ACK >> > > > Content-Length: 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 >> > >> > ____________________________________________________________ >> _____________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://confluence.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at 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 v.zakhozhai at gmail.com Tue May 23 09:23:57 2017 From: v.zakhozhai at gmail.com (Vladyslav Zakhozhai) Date: Tue, 23 May 2017 12:23:57 +0300 Subject: [Freeswitch-users] mod_xml_curl performance questions In-Reply-To: References: <1494961415.1484944.978717552.3EE447BC@webmail.messagingengine.com> Message-ID: Hi, Michael, I see no big problems here. I agree with Giovanni that xml_curl does not affect call quality but call initiation duration and success. You can decrease load on your webserver by cacheing xml responses and with backup webserver. And I think it would work just fine. 2017-05-18 19:27 GMT+03:00 Giovanni Maruzzelli : > On 16 May 2017 at 21:03, Michael Avers wrote: > >> >> I'm planning on moving all our local lua-generated XML to use >> mod_xml_curl over HTTPS. However, the primary http server is not in the >> same data center as our FS instances. Ping time is about 50ms. Is this >> going to be an issue? Is it typical to run xml_curl hosted elsewhere? The >> web app itself is written in Go and is very fast. Just wondering about >> network latency aspect of it all. >> > > usually you run the webserver in same datacenter. > > That said, maybe in your case this is not a problem, and your application > can bear the round trips. Take into account this do not affect call quality > or call latency, it only affect how fast FreeSWITCH can get data (eg, how > fast it takes to get auth info for a caller, how fast get the dialplan > snippet to be executed, etc.) > Once the call is established, rtp packets runs independently from > database. > > > > > >> >> 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 >> > > > > -- > > Sincerely, > > Giovanni Maruzzelli > OpenTelecom.IT > cell: +39 347 266 56 18 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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, Владислав Захожай -------------- next part -------------- An HTML attachment was scrubbed... URL: From luis.daniel.lucio at gmail.com Mon May 22 14:33:58 2017 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Mon, 22 May 2017 10:33:58 -0400 Subject: [Freeswitch-users] freeswitch segfault at sp error 4 in libmysqlclient.so.18.0.0 In-Reply-To: References: Message-ID: Just a guess, don't mix MySQL libs with mariadb 10's Some libraries are not backwards compatible Le 22 mai 2017 4:13 AM, "Karsten Horsmann" a écrit : > Hello, > > i am a littlebit confused about an error that my FreeSWITCH 1.6.17 Testing > Box has. > Its an vanilla rpm installation for CentOS 7.3.11. After a while it stops > working. > > The only thing mysql-based i do is an lua-script that use lua-dbi for > mysql. > Maybe this script kills the prozess. How can i debug that? Should i fill > an JIRA? > > FreeSWITCH Version 1.6.17 > > In the logfile i see this error messages: > > > May 20 17:40:50 fs16dev systemd: Cannot add dependency job for unit > microcode.service, ignoring: Unit is not loaded properly: Invalid argument. > May 20 17:40:50 fs16dev systemd: Started Check_MK (172.20.160.60:41436). > May 20 17:40:50 fs16dev systemd: Starting Check_MK (172.20.160.60:41436 > )... > May 20 17:41:18 fs16dev kernel: freeswitch[5603]: segfault at 0 ip > 00007f8946fb2e5f sp 00007f894773f240 error 4 in libmysqlclient.so.18.0.0[ > 7f8946f8b000+2de000] > May 20 17:41:18 fs16dev systemd: freeswitch.service: main process exited, > code=killed, status=11/SEGV > May 20 17:41:18 fs16dev systemd: Unit freeswitch.service entered failed > state. > May 20 17:41:18 fs16dev systemd: freeswitch.service failed. > May 20 17:41:18 fs16dev systemd: freeswitch.service holdoff time over, > scheduling restart. > May 20 17:41:18 fs16dev systemd: Cannot add dependency job for unit > microcode.service, ignoring: Unit is not loaded properly: Invalid argument. > May 20 17:41:18 fs16dev systemd: Starting freeswitch... > May 20 17:41:18 fs16dev freeswitch: 5610 Backgrounding. > May 20 17:41:20 fs16dev freeswitch: FreeSWITCH[5609] Waiting for > background process pid:5610 to be ready..... > May 20 17:41:20 fs16dev freeswitch: FreeSWITCH[5609] System Ready pid:5610 > May 20 17:41:20 fs16dev systemd: Started freeswitch. > > BTW - the rpm systemd service didnt restart the server if it breaks. > I used an debian style version of the freeswitch stash. > > > Kind Regards > -- > Mit freundlichen Grüßen > *Karsten Horsmann* > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 wsimon at stratusvideo.com Mon May 22 14:19:16 2017 From: wsimon at stratusvideo.com (William Simon) Date: Mon, 22 May 2017 14:19:16 +0000 Subject: [Freeswitch-users] We've been Busy! In-Reply-To: <060c01d2d0df$81512970$83f37c50$@freeswitch.org> References: <04ad01d2d0da$ba3ddcb0$2eb99610$@freeswitch.org> <060c01d2d0df$81512970$83f37c50$@freeswitch.org> Message-ID: Is Mr. West still attacking the data center? We are not able to access the repos or the Hipchat, among other things, this morning. On May 19, 2017, at 4:35 PM, Ken Rice > wrote: This is both a test message and a user notification. (if you saw this twice its because I sent it again to make sure everyone sees it!) We’ve been busy and are going to be all weekend. So today I’ve done a pile of upgrades to the list server here. And tomorrow we’ll have a team lead by Mr West attacking the data center! So don’t be surprised if you see services offline a bit tomorrow. This will include everything at the FreeSWITC.org datacenter as we replace some critical infrastructure. What does this mean for you, if you need to do an install from our package repos or do a git clone etc, do it before about 0900 US Eastern in the AM or plan on doing it much later in the day. This will be an extended maintenance window. If you want to help us avoid these outages in the future please visit https://freeswitch.org/ and use the donate link in the menu. Contributions from the FreeSWITCH userbase is what allows us to keep the hardware up and running and where possible deploy redundant hardware so we can fail over and not miss a beat. Thanks! Ken “The information transmitted is intended only for the person or entity to which it is addressed and may contain proprietary, business-confidential and/or privileged material. If you are not the intended recipient of this message you are hereby notified that any use, review, retransmission, dissemination, distribution, reproduction or any action taken in reliance upon this message is prohibited. If you received this in error, please contact the sender and delete the material from any computer.” _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org “The information transmitted is intended only for the person or entity to which it is addressed and may contain proprietary, business-confidential and/or privileged material. If you are not the intended recipient of this message you are hereby notified that any use, review, retransmission, dissemination, distribution, reproduction or any action taken in reliance upon this message is prohibited. If you received this in error, please contact the sender and delete the material from any computer.” -------------- next part -------------- An HTML attachment was scrubbed... URL: From bipin at xbipin.com Wed May 24 08:29:12 2017 From: bipin at xbipin.com (Bipin Patel) Date: Wed, 24 May 2017 12:29:12 +0400 Subject: [Freeswitch-users] python ESL module Message-ID: hi, is any1 using the ESL python module to create python scripts for FS ESL, if so can some1 kindly let me know how to get it to install as i keep getting the below error when i try Collecting python-ESL Using cached python-ESL-1.4.18.tar.gz Installing collected packages: python-ESL Running setup.py install for python-ESL ... error Complete output from command c:\python27\python.exe -u -c "import setuptools, tokenize;__file__='c:\\users\\bipin\\appdata\\local\\temp\\pip-build-snoqec\\python-ESL\\setup.py';f=getattr(tokenize, 'open', open)(__file__);code=f.read().replace('\r\n', '\n');f.close();exec(compile(code, __file__, 'exec'))" install --record c:\users\bipin\appdata\local\temp\pip-4yo5fb-record\install-record.txt --single-version-externally-managed --compile: running install running build running build_py creating build creating build\lib.win32-2.7 copying ESL.py -> build\lib.win32-2.7 running build_ext building '_ESL' extension swigging ESL.i to ESL_wrap.cpp D:\swigwin-3.0.12\swig.exe -python -classic -c++ -DMULTIPLICITY -threads -I. -o ESL_wrap.cpp ESL.i creating build\temp.win32-2.7 creating build\temp.win32-2.7\Release C:\Users\Bipin\AppData\Local\Programs\Common\Microsoft\Visual C++ for Python\9.0\VC\Bin\cl.exe /c /nologo /Ox /MD /W3 /GS- /DNDEBUG -Ic:\python27\include -Ic:\python27\PC /Tcesl.c /Fobuild\temp.win32-2.7\Release\esl.obj -I. esl.c .\esl.h(186) : fatal error C1083: Cannot open include file: 'sys/time.h': No such file or directory error: command 'C:\\Users\\Bipin\\AppData\\Local\\Programs\\Common\\Microsoft\\Visual C++ for Python\\9.0\\VC\\Bin\\cl.exe' failed with exit status 2 ---------------------------------------- Command "c:\python27\python.exe -u -c "import setuptools, tokenize;__file__='c:\\users\\bipin\\appdata\\local\\temp\\pip-build-snoqec\\python-ESL\\setup.py';f=getattr(tokenize, 'open', open)(__file__);code=f.read().replace('\r\n', '\n');f.close();exec(compile(code, __file__, 'exec'))" install --record c:\users\bipin\appdata\local\temp\pip-4yo5fb-record\install-record.txt --single-version-externally-managed --compile" failed with error code 1 in c:\users\bipin\appdata\local\temp\pip-build-snoqec\python-ESL\ -- Regards, Bipin ------------------------------------------------------------------------ -------------- next part -------------- An HTML attachment was scrubbed... URL: From jose.lopes at itcenter.com.pt Wed May 24 08:49:37 2017 From: jose.lopes at itcenter.com.pt (=?UTF-8?Q?Jos=C3=A9_Lopes?=) Date: Wed, 24 May 2017 09:49:37 +0100 Subject: [Freeswitch-users] Conference with flags wait-mod|nomoh plays MOH Message-ID: Hello, I have configured a audio conference with flags wait-mod and nomoh. I make a call to the conference as the first member and I listen moh, instead of silence. I am using FreeSwitch 1.6.17 If I configured the audio conference without the flag wait-mod, when i enter the conference I have silence and it is ok. This is a bug or this is the right behavior ? There is any option that i can configure so when i configure the conference with flags wait-mod and nomoh, when the first members enters, it get silence? Thanks in advance. Best regards, Jose Lopes -------------- next part -------------- An HTML attachment was scrubbed... URL: From s.safarov at gmail.com Wed May 24 11:05:42 2017 From: s.safarov at gmail.com (Sergey Safarov) Date: Wed, 24 May 2017 11:05:42 +0000 Subject: [Freeswitch-users] timer issue Message-ID: I found strange timestamp in FreeSwitch logs https://pastebin.freeswitch.org/view/2933090b message received at send 513 bytes to udp/[31.24.110.61]:5060 at 11:34: 20.608499: system time but call processing have time stamp 12:14:13.279724 Please help me understand why this take place. How it can be fixed? Used kazoo config -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.org Wed May 24 15:59:11 2017 From: brian at freeswitch.org (Brian West) Date: Wed, 24 May 2017 10:59:11 -0500 Subject: [Freeswitch-users] Conference with flags wait-mod|nomoh plays MOH In-Reply-To: References: Message-ID: nomoh is a member flag, not a conference flag. On Wed, May 24, 2017 at 3:49 AM, José Lopes wrote: > Hello, > > I have configured a audio conference with flags wait-mod and nomoh. > > I make a call to the conference as the first member and I listen moh, > instead of silence. > I am using FreeSwitch 1.6.17 > > If I configured the audio conference without the flag wait-mod, when i > enter the conference I have silence and it is ok. > > > This is a bug or this is the right behavior ? > There is any option that i can configure so when i configure the > conference with flags wait-mod and nomoh, when the first members enters, it > get silence? > > Thanks in advance. > > Best regards, > Jose Lopes > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Book a phone call (CST) Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.org Wed May 24 19:11:59 2017 From: brian at freeswitch.org (Brian West) Date: Wed, 24 May 2017 14:11:59 -0500 Subject: [Freeswitch-users] timer issue In-Reply-To: References: Message-ID: Did you happen to change your clock or timezone after FreeSWITCH was started? and have long has this system been running? 'fsctl sync_clock' will fix the clock sync, is this VM? On Wed, May 24, 2017 at 6:05 AM, Sergey Safarov wrote: > I found strange timestamp in FreeSwitch logs https://pastebin. > freeswitch.org/view/2933090b > message received at send 513 bytes to udp/[31.24.110.61]:5060 at 11:34: > 20.608499: system time > but call processing have time stamp 12:14:13.279724 > Please help me understand why this take place. How it can be fixed? > > Used kazoo config > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Book a phone call (CST) Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: From thesipguy at gmail.com Thu May 25 09:09:37 2017 From: thesipguy at gmail.com (Schneur Rosenberg) Date: Thu, 25 May 2017 12:09:37 +0300 Subject: [Freeswitch-users] Freeswitch PRI support In-Reply-To: References: <27a9d063.ad3b.15bd85e9762.Coremail.xxxman2008@126.com> <6fc1c046-45f9-cdbe-2896-a077dbcc5dad@xbipin.com> Message-ID: I once built a system for a cellphone rental company in Israel, at the time termination to Israel was very expensive, they wanted to provide a local USA DID to ring on the Israeli mobile phone, so they bought a 32 channel 2N 3g gateway, they had a deal with the mobile carrier that they got a couple of thousand minutes per SIM to call any other SIM in the account (they called it VPN minutes) then the customers would call the USA DID which terminated the call through the 3G gateway, they set limits on each SIM how many outgoing minutes it can call, the 3G gateway distributed the calls in round robin until a sim used its allowed minutes. So they were able to offer a USA DID to ring to the mobile phone at a very low rate, but the market has changed significantly and its a lot cheaper to terminate the calls via sip trunks, and by doing so the called party also got caller ID. On Wed, May 10, 2017 at 5:04 PM, Colton Conor wrote: > So how does this GSM stuff work in the USA? Would I got buy a cell phone > plan from someone like AT&T or T-Mobile, and then insert the SIM into these > devices. The device would then covert the audio from GSM to SIP, and then I > could use it like a SIP trunk? > > What is the advantage of doing this over just buying a pure SIP trunk from > an internet provider? > > > > On Wed, May 10, 2017 at 4:35 AM, David Villasmil < > david.villasmil.work at gmail.com> wrote: > >> There's another chinese brand that has a 32 port, I've used it, works >> pretty well, i think it's "GoIP" or something. >> >> On Wed, May 10, 2017 at 10:39 AM Bipin Patel wrote: >> >>> hi, >>> >>> we use matrix boxes but 4 port ones, they work quiet well >>> >>> >>> Regards, >>> Bipin >>> >>> >>> ------------------------------ >>> -------- Original Message -------- >>> Subject: Re: [Freeswitch-users] Freeswitch PRI support >>> From: Deepika Yadav >>> To: FreeSWITCH Users Help >>> >>> Date: 5/10/2017, 11:02:49 AM >>> >>> Hi, >>> >>> I searched some gateways online. The ones available at ebay are cheap, >>> the following link shows a gateway supporting 32 SIMS for $859 : >>> >>> http://www.ebay.com/itm/GoIP-32-32-Port-VoIP-GSM-Gateway-wit >>> h-32-External-Antenna-/291811535942 >>> >>> However, when I contacted a company called "Matrix Telecom Solution", >>> they gave the quote of $4150 for similar number of SIM support. On asking >>> them to compare the two gateways, they said that the one available at ebay >>> is a Chinese gateway about which they are apprehensive for the quality, >>> warranty and working. >>> >>> Regards, >>> Deepika >>> >>> >>> On Fri, May 5, 2017 at 4:58 PM, Raymond wrote: >>> >>>> >>>> HI, >>>> >>>> Regarding this. i have asked some question of module "GSMOpen". >>>> >>>> In fact ,i want to make Freeswitch box as "GSM VOIP Gateway" by using >>>> "gsmopen". But...seems no one help me . :-(( . If it worked , can save >>>> more money than hardware gsm voip gateway ( GOIP ). >>>> >>>> Raymond >>>> >>>> >>>> >>>> At 2017-05-04 17:24:13, "Giovanni Maruzzelli" >>>> wrote: >>>> >>>> For PRI you use Sangoma or Patton. >>>> >>>> But, why don't you use an hardware gateway sip<->gsm? >>>> >>>> It would save you very big money. >>>> >>>> Check on ebay and google, there are many of them, you put SIMs inside, >>>> and you are good to go. >>>> >>>> -giovanni >>>> >>>> >>>> sent from mobile >>>> cell: +39 347 266 56 18 >>>> Giovanni Maruzzelli >>>> OpenTelecom.IT >>>> >>>> Il 04 mag 2017 11:13, "Deepika Yadav" ha >>>> scritto: >>>> >>>>> Hi, >>>>> >>>>> I am using Freeswitch in an application that initiates conference >>>>> calls amongst 20 users in cellular network (mobile phones in GSM network). >>>>> Currently, for the VOIP-GSM gateway, we are using the service from a >>>>> company called Doorvaani. >>>>> >>>>> But, since, the cost of call estimates to be high and we cannot debug >>>>> the call drops; we are thinking to buy our own PRI card. >>>>> >>>>> I am seeking recommendation on following points: >>>>> >>>>> 1. Which card should I buy that is most easily configurable with >>>>> the Freeswitch i.e. company and type. >>>>> 2. Reference on how should I start to make Freeswitch configure >>>>> with the PRI card and start sending and receiving calls. For the current >>>>> gateway service in use, I simply put the authentication credentials for the >>>>> corresponding VOIP line offered by the company Doorvaani in the external >>>>> SIP Profile. In case of PRI, what all changes are needed? >>>>> >>>>> Regards, >>>>> Deepika >>>>> >>>>> -- >>>>> Regards >>>>> Deepika >>>>> https://www.iiitd.edu.in/~deepikay/ >>>>> >>>>> ____________________________________________________________ >>>>> _____________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at 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://www.iiitd.edu.in/~deepikay/ >>> >>> >>> _________________________________________________________________________ >>> 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 fanx07 at gmail.com Thu May 25 11:18:58 2017 From: fanx07 at gmail.com (Anonim Stefan) Date: Thu, 25 May 2017 14:18:58 +0300 Subject: [Freeswitch-users] configure playback outbound http proxy In-Reply-To: References: <5074F5F6-942A-4A43-888F-ECF91C950DAB@jerris.com> Message-ID: Hi, What about: 1. setting an http_proxy environment variable in /etc/sysconfig/freeswitch (similar to 'PYTHONPATH' or 'SWIFT_HOME') ? or 2. setting an http_proxy like here: https://askubuntu.com/questions/228530/updating-http-proxy-environment-variable ? Would Freeswitch consider that http_proxy for mod_shout? Thank you, Stefan On Mon, Feb 20, 2017 at 10:26 AM, Anonim Stefan wrote: > Thank you for answer! > > I couldn't find something similar either, hence the question. > > Stefan > > On Tue, Feb 14, 2017 at 6:49 PM, Michael Jerris wrote: > >> I do not believe mod_shout supports a proxy param. >> >> >> > On Feb 14, 2017, at 9:58 AM, Anonim Stefan wrote: >> > >> > Hi, >> > >> > Does Freeswitch provide a parameter to configure an outbound http >> proxy, when playback a http stream? >> > >> > What I am trying to avoid here is to add specific routes for each IP I >> am playing back from. >> > >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 ira at connectmevoice.com Thu May 25 19:54:02 2017 From: ira at connectmevoice.com (Ira Tessler) Date: Thu, 25 May 2017 15:54:02 -0400 Subject: [Freeswitch-users] Error Deleting from sip_dialogs Message-ID: Hi, I saw this error in my freeswitch.log. Anyone know why this would occur or how to fix it? 2017-05-23 13:03:47.553338 [ERR] switch_odbc.c:522 ERR: [delete from sip_dialogs where uuid='3fc23dbe-4762-4d58-9a49-156d99001fef'] [STATE: HY000 CODE 1213 ERROR: [MySQL][ODBC 5.1 Driver][mysqld-5.6.23-log]Deadlock found when trying to get lock; try restarting transaction ] 2017-05-23 13:03:47.553338 [ERR] switch_core_sqldb.c:587 ODBC SQL ERR [STATE: HY000 CODE 1213 ERROR: [MySQL][ODBC 5.1 Driver][mysqld-5.6.23-log]Deadlock found when trying to get lock; try restarting transaction ] delete from sip_dialogs where uuid='3fc23dbe-4762-4d58-9a49-156d99001fef' Thanks, Ira Tessler Lead Software Engineer ConnectMe (732) 490-9007 x2 ira at connectmevoice.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From abaci64 at gmail.com Fri May 26 01:29:32 2017 From: abaci64 at gmail.com (Abaci B) Date: Thu, 25 May 2017 21:29:32 -0400 Subject: [Freeswitch-users] Windows build - mod_lua In-Reply-To: References: Message-ID: I see that there is no mod_lua in the windows 1.6.17 build, is this official or a bug? Thanks On Tue, Nov 8, 2016 at 5:34 PM, Gregor Nanger wrote: > I've compiled FS with visual studio and there is no mod_lua.dll generated. > Hence I get error when FS starts that it can not find dll. > > Windows build went fine (with excluded modv8) > > Anyone also find this problem? > > Best regards, 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: From steveayre at gmail.com Fri May 26 12:23:03 2017 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 26 May 2017 13:23:03 +0100 Subject: [Freeswitch-users] return to shell only when call from originate is complete? In-Reply-To: <322d2b9b-e2e1-8ee2-cfa8-c8b825d832a9@xbipin.com> References: <69660425-2e51-1943-4fba-fc1ff5ea7094@xbipin.com> <4E8BF635-E985-4056-B301-A1EEFF702B09@jerris.com> <8eb5928b-d77f-eb70-fa0b-84f9d21f7d5b@xbipin.com> <4112C9C4-5BAA-408F-9731-8B11F3CCDAD8@stratusvideo.com> <0f62f615-0556-55aa-c52b-21955f56e1db@xbipin.com> <1edfad2b-dbe1-837b-408b-1bcf8e6df6c7@xbipin.com> <322d2b9b-e2e1-8ee2-cfa8-c8b825d832a9@xbipin.com> Message-ID: Try https://github.com/sjthomason/PyESL - pure python implementation using the same API as the official one. I'd generate the UUID yourself and set origination_uuid in the dialstring, which means you can subscribe to events on that call before the origination. CHANNEL_HANGUP and CHANNEL_DESTROY might be good ones to watch for to know when it's complete. On 22 May 2017 at 11:21, Bipin Patel wrote: > hi, > > i tried to install the python-ESL module using pip but it wont compile > even with swig, complaining about sys/time.h > > > Regards, > Bipin > > > ------------------------------ > -------- Original Message -------- > Subject: Re: [Freeswitch-users] return to shell only when call from > originate is complete? > From: Bipin Patel > To: FreeSWITCH Users Help > > Date: 5/22/2017, 10:33:53 AM > > hi, > > btw is it possible to use python with esl on windows, i cant seem to find > any guide to it > > > Regards, > Bipin > > > ------------------------------ > -------- Original Message -------- > Subject: Re: [Freeswitch-users] return to shell only when call from > originate is complete? > From: Bipin Patel > To: FreeSWITCH Users Help > > Date: 5/22/2017, 10:10:17 AM > > hi, > > that was a good one, the reason im not using esl directly from python is > because all i want to do is send originate command and get UUID and thats > about it and stop the python script as soon as its done, ill try it once > the requirements go higher with the client :) > > > Regards, > Bipin > > > ------------------------------ > -------- Original Message -------- > Subject: Re: [Freeswitch-users] return to shell only when call from > originate is complete? > From: Anthony Minessale > > To: FreeSWITCH Users Help > > Date: 5/22/2017, 8:38:13 AM > > I am trying to catch a fish can you help me? > > Get a pole and use some fishing line with a hook and add a worm. > > I dont really want to use a pole, I am just going to throw a grenade in > the water and a fish will fall into my boat. > > > > > > > On Sun, May 21, 2017 at 10:43 AM Bipin Patel wrote: > >> hi, >> >> im trying to avoid going the esl route, for now i solved this by calling >> a bash script from python which monitors the call once the UUID is passed >> to it so that holds the python command till call is over and then it >> proceeds further >> >> >> Regards, >> Bipin >> >> >> ------------------------------ >> -------- Original Message -------- >> Subject: Re: [Freeswitch-users] return to shell only when call from >> originate is complete? >> From: Tihomir Culjaga >> To: FreeSWITCH Users Help >> >> Date: 5/21/2017, 7:25:31 PM >> >> i still think you are better consume events with python... >> its easy and straightforward >> >> >> #!/usr/bin/env python >> >> import string >> import sys >> >> from ESL import * >> >> con = ESLconnection("127.0.0.1","8021","ClueCon") >> #are we connected? >> callDirection = "unknown" >> >> >> if con.connected: >> print "we connected \n" >> con.events("plain", "all"); >> >> while 1: >> #my $e = $con->recvEventTimed(100); >> e = con.recvEvent() >> >> if e: >> #print e.serialize() >> en = e.getHeader("Event-Name") >> print "Name =>" ,en >> >> if en == "CHANNEL_OUTGOING": >> callDirection = "OUT" >> print "Direction =", callDirection >> >> elif en == "CHANNEL_ORIGINATE": >> print "Originate call - direction =", callDirection >> >> elif en == "CHANNEL_CALLSTATE": >> print "Call State =>", e.getHeader("Answer-State") >> >> elif en == "CHANNEL_ANSWER": >> print "Call START - direction =", callDirection >> >> elif en == "CHANNEL_HANGUP": >> print "Call END - direction =", callDirection >> >> elif en == "CHANNEL_PARK": >> callDirection = "IN" >> print "Call PARK - direction =", callDirection >> >> >> >> you can work it out from here pretty everything you want. >> >> >> T. >> >> On 21 May 2017 at 14:32, Bipin Patel wrote: >> >>> hi, >>> >>> i converted the bash script to python as below: >>> >>> ret = subprocess.Popen('sudo fs_cli -x \"show channels\" | grep -c >>> \"'+UUID+'\"', shell=True, stdout=subprocess.PIPE, stderr=subprocess.PIPE) >>> sout, serr = ret.communicate() >>> print sout >>> >>> the reason i use -c instead of -q is because fs_cli always returns a 0 >>> exit code so it doesnt work through python when combined with grep so i >>> tried to exit loop when the counter is 0 meaning the call is completed, but >>> the above returns some weird numbers in sout, when the call is running it >>> shows 5 and when done shows 3 but same if i run directly through console it >>> shows accurately. >>> >>> any idea how to fix this? >>> >>> >>> Regards, >>> Bipin >>> >>> >>> ------------------------------ >>> -------- Original Message -------- >>> Subject: Re: [Freeswitch-users] return to shell only when call from >>> originate is complete? >>> From: William Simon >>> To: FreeSWITCH Users Help >>> >>> Date: 5/20/2017, 1:29:05 AM >>> >>> If you are using fs_cli then a simple shell script can monitor the call >>> and return when the call completes: >>> >>> #!/bin/bash >>> >>> RESULT=$( fs_cli -x "originate leg1 leg2" ) >>> >>> if [[ $RESULT =~ "OK" ]]; then >>> UUID=$( echo "$RESULT" | cut -f 2 -d ' ' ) >>> while fs_cli -x "show channels" | grep -q $UUID ; >>> do >>> sleep 5; >>> done >>> echo "Call completed." >>> fi >>> >>> >>> >>> >>> On May 19, 2017, at 12:31 PM, Bipin Patel wrote: >>> >>> hi, >>> >>> any other way this can be done >>> >>> >>> Regards, >>> Bipin >>> >>> >>> ------------------------------ >>> -------- Original Message -------- >>> Subject: Re: [Freeswitch-users] return to shell only when call from >>> originate is complete? >>> From: Michael Jerris >>> To: Bipin Patel >>> Date: 5/19/2017, 7:48:22 PM >>> >>> No, thats not what originate does. >>> >>> On May 19, 2017, at 11:43 AM, Bipin Patel wrote: >>> >>> hi, >>> >>> does it happen right after answer? >>> is there any way to lock it so it returns only after failure or after >>> hangup? yes im using ignore_early_media >>> >>> >>> Regards, >>> Bipin >>> >>> >>> ------------------------------ >>> -------- Original Message -------- >>> Subject: Re: [Freeswitch-users] return to shell only when call from >>> originate is complete? >>> From: Michael Jerris >>> To: FreeSWITCH Users Help >>> >>> Date: 5/19/2017, 7:34:35 PM >>> >>> originate returns when we get call progress, or in the case of >>> ignore_early_media, on answer or failure. >>> >>> On May 19, 2017, at 10:36 AM, Bipin Patel wrote: >>> >>> hi, >>> >>> i was wondering if its possible to to release handle from an originate >>> command only once the call is completed, i mean i send a originate command >>> from shell using fs_cli but only return back to prompt once the call is >>> completed rather than as soon as a bridge is created or a UUID generated >>> >>> >>> >>> >>> >>> >>> “The information transmitted is intended only for the person or entity >>> to which it is addressed and may contain proprietary, business-confidential >>> and/or privileged material. If you are not the intended recipient of this >>> message you are hereby notified that any use, review, retransmission, >>> dissemination, distribution, reproduction or any action taken in reliance >>> upon this message is prohibited. If you received this in error, please >>> contact the sender and delete the material from any computer.” >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> >>> “The information transmitted is intended only for the person or entity >>> to which it is addressed and may contain proprietary, business-confidential >>> and/or privileged material. If you are not the intended recipient of this >>> message you are hereby notified that any use, review, retransmission, >>> dissemination, distribution, reproduction or any action taken in reliance >>> upon this message is prohibited. If you received this in error, please >>> contact the sender and delete the material from any computer.” >>> >>> _________________________________________________________________________ >>> 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 > > -- > 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 <(919)%20386-9900> > > https://www.youtube.com/watch?v=9XXgW34t40s > https://www.youtube.com/watch?v=NLaDpGQuZDA > > > _________________________________________________________________________ > 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.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.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 rahul.ultimate at gmail.com Fri May 26 12:23:33 2017 From: rahul.ultimate at gmail.com (Rahul MathuR) Date: Fri, 26 May 2017 17:53:33 +0530 Subject: [Freeswitch-users] Proper syntax of sched_hangup from webapi Message-ID: Hi guys, I am trying to implement a callback functionality using mod_xml_rpc, my URL looks like - http://192.168.2.54:8080/webapi/originate?{origination_caller_id_name=8888888888,origination_caller_id_number=8888888888}sofia/gateway/OBG1/3333333333%20set(execute_on_answer=sched_hangup%20%2B10%20normal_clearing%20bleg)%20&bridge(sofia/gateway/OBG1/4444444444) But FS sends out INVITE to 3333333333, receives 183 and then even before it could send out an INVITE to 4444444444, it feels it has reached to the last statement of execution and hence sends CANCEL and drops the call. When I put sched_hangup at the last, it just doesn't play any part in call duration control. I feel that somewhere I am making some mistake but I need another set of eyes to locate it. Could you kindly help me resolve this issue. -- Warm Regds. MathuR -------------- next part -------------- An HTML attachment was scrubbed... URL: From steveayre at gmail.com Fri May 26 12:24:33 2017 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 26 May 2017 13:24:33 +0100 Subject: [Freeswitch-users] return to shell only when call from originate is complete? In-Reply-To: References: <69660425-2e51-1943-4fba-fc1ff5ea7094@xbipin.com> <4E8BF635-E985-4056-B301-A1EEFF702B09@jerris.com> <8eb5928b-d77f-eb70-fa0b-84f9d21f7d5b@xbipin.com> <4112C9C4-5BAA-408F-9731-8B11F3CCDAD8@stratusvideo.com> <0f62f615-0556-55aa-c52b-21955f56e1db@xbipin.com> <1edfad2b-dbe1-837b-408b-1bcf8e6df6c7@xbipin.com> <322d2b9b-e2e1-8ee2-cfa8-c8b825d832a9@xbipin.com> Message-ID: This will automatically use whichever one you have available (preferring the FreeSWITCH one) try: import ESL except ImportError: import PyESL as ESL On 26 May 2017 at 13:23, Steven Ayre wrote: > Try https://github.com/sjthomason/PyESL - pure python implementation > using the same API as the official one. > > I'd generate the UUID yourself and set origination_uuid in the dialstring, > which means you can subscribe to events on that call before the > origination. CHANNEL_HANGUP and CHANNEL_DESTROY might be good ones to watch > for to know when it's complete. > > On 22 May 2017 at 11:21, Bipin Patel wrote: > >> hi, >> >> i tried to install the python-ESL module using pip but it wont compile >> even with swig, complaining about sys/time.h >> >> >> Regards, >> Bipin >> >> >> ------------------------------ >> -------- Original Message -------- >> Subject: Re: [Freeswitch-users] return to shell only when call from >> originate is complete? >> From: Bipin Patel >> To: FreeSWITCH Users Help >> >> Date: 5/22/2017, 10:33:53 AM >> >> hi, >> >> btw is it possible to use python with esl on windows, i cant seem to find >> any guide to it >> >> >> Regards, >> Bipin >> >> >> ------------------------------ >> -------- Original Message -------- >> Subject: Re: [Freeswitch-users] return to shell only when call from >> originate is complete? >> From: Bipin Patel >> To: FreeSWITCH Users Help >> >> Date: 5/22/2017, 10:10:17 AM >> >> hi, >> >> that was a good one, the reason im not using esl directly from python is >> because all i want to do is send originate command and get UUID and thats >> about it and stop the python script as soon as its done, ill try it once >> the requirements go higher with the client :) >> >> >> Regards, >> Bipin >> >> >> ------------------------------ >> -------- Original Message -------- >> Subject: Re: [Freeswitch-users] return to shell only when call from >> originate is complete? >> From: Anthony Minessale >> >> To: FreeSWITCH Users Help >> >> Date: 5/22/2017, 8:38:13 AM >> >> I am trying to catch a fish can you help me? >> >> Get a pole and use some fishing line with a hook and add a worm. >> >> I dont really want to use a pole, I am just going to throw a grenade in >> the water and a fish will fall into my boat. >> >> >> >> >> >> >> On Sun, May 21, 2017 at 10:43 AM Bipin Patel wrote: >> >>> hi, >>> >>> im trying to avoid going the esl route, for now i solved this by calling >>> a bash script from python which monitors the call once the UUID is passed >>> to it so that holds the python command till call is over and then it >>> proceeds further >>> >>> >>> Regards, >>> Bipin >>> >>> >>> ------------------------------ >>> -------- Original Message -------- >>> Subject: Re: [Freeswitch-users] return to shell only when call from >>> originate is complete? >>> From: Tihomir Culjaga >>> To: FreeSWITCH Users Help >>> >>> Date: 5/21/2017, 7:25:31 PM >>> >>> i still think you are better consume events with python... >>> its easy and straightforward >>> >>> >>> #!/usr/bin/env python >>> >>> import string >>> import sys >>> >>> from ESL import * >>> >>> con = ESLconnection("127.0.0.1","8021","ClueCon") >>> #are we connected? >>> callDirection = "unknown" >>> >>> >>> if con.connected: >>> print "we connected \n" >>> con.events("plain", "all"); >>> >>> while 1: >>> #my $e = $con->recvEventTimed(100); >>> e = con.recvEvent() >>> >>> if e: >>> #print e.serialize() >>> en = e.getHeader("Event-Name") >>> print "Name =>" ,en >>> >>> if en == "CHANNEL_OUTGOING": >>> callDirection = "OUT" >>> print "Direction =", callDirection >>> >>> elif en == "CHANNEL_ORIGINATE": >>> print "Originate call - direction =", callDirection >>> >>> elif en == "CHANNEL_CALLSTATE": >>> print "Call State =>", e.getHeader("Answer-State") >>> >>> elif en == "CHANNEL_ANSWER": >>> print "Call START - direction =", callDirection >>> >>> elif en == "CHANNEL_HANGUP": >>> print "Call END - direction =", callDirection >>> >>> elif en == "CHANNEL_PARK": >>> callDirection = "IN" >>> print "Call PARK - direction =", callDirection >>> >>> >>> >>> you can work it out from here pretty everything you want. >>> >>> >>> T. >>> >>> On 21 May 2017 at 14:32, Bipin Patel wrote: >>> >>>> hi, >>>> >>>> i converted the bash script to python as below: >>>> >>>> ret = subprocess.Popen('sudo fs_cli -x \"show channels\" | grep -c >>>> \"'+UUID+'\"', shell=True, stdout=subprocess.PIPE, stderr=subprocess.PIPE) >>>> sout, serr = ret.communicate() >>>> print sout >>>> >>>> the reason i use -c instead of -q is because fs_cli always returns a 0 >>>> exit code so it doesnt work through python when combined with grep so i >>>> tried to exit loop when the counter is 0 meaning the call is completed, but >>>> the above returns some weird numbers in sout, when the call is running it >>>> shows 5 and when done shows 3 but same if i run directly through console it >>>> shows accurately. >>>> >>>> any idea how to fix this? >>>> >>>> >>>> Regards, >>>> Bipin >>>> >>>> >>>> ------------------------------ >>>> -------- Original Message -------- >>>> Subject: Re: [Freeswitch-users] return to shell only when call from >>>> originate is complete? >>>> From: William Simon >>>> To: FreeSWITCH Users Help >>>> >>>> Date: 5/20/2017, 1:29:05 AM >>>> >>>> If you are using fs_cli then a simple shell script can monitor the call >>>> and return when the call completes: >>>> >>>> #!/bin/bash >>>> >>>> RESULT=$( fs_cli -x "originate leg1 leg2" ) >>>> >>>> if [[ $RESULT =~ "OK" ]]; then >>>> UUID=$( echo "$RESULT" | cut -f 2 -d ' ' ) >>>> while fs_cli -x "show channels" | grep -q $UUID ; >>>> do >>>> sleep 5; >>>> done >>>> echo "Call completed." >>>> fi >>>> >>>> >>>> >>>> >>>> On May 19, 2017, at 12:31 PM, Bipin Patel wrote: >>>> >>>> hi, >>>> >>>> any other way this can be done >>>> >>>> >>>> Regards, >>>> Bipin >>>> >>>> >>>> ------------------------------ >>>> -------- Original Message -------- >>>> Subject: Re: [Freeswitch-users] return to shell only when call from >>>> originate is complete? >>>> From: Michael Jerris >>>> To: Bipin Patel >>>> Date: 5/19/2017, 7:48:22 PM >>>> >>>> No, thats not what originate does. >>>> >>>> On May 19, 2017, at 11:43 AM, Bipin Patel wrote: >>>> >>>> hi, >>>> >>>> does it happen right after answer? >>>> is there any way to lock it so it returns only after failure or after >>>> hangup? yes im using ignore_early_media >>>> >>>> >>>> Regards, >>>> Bipin >>>> >>>> >>>> ------------------------------ >>>> -------- Original Message -------- >>>> Subject: Re: [Freeswitch-users] return to shell only when call from >>>> originate is complete? >>>> From: Michael Jerris >>>> To: FreeSWITCH Users Help >>>> >>>> Date: 5/19/2017, 7:34:35 PM >>>> >>>> originate returns when we get call progress, or in the case of >>>> ignore_early_media, on answer or failure. >>>> >>>> On May 19, 2017, at 10:36 AM, Bipin Patel wrote: >>>> >>>> hi, >>>> >>>> i was wondering if its possible to to release handle from an originate >>>> command only once the call is completed, i mean i send a originate command >>>> from shell using fs_cli but only return back to prompt once the call is >>>> completed rather than as soon as a bridge is created or a UUID generated >>>> >>>> >>>> >>>> >>>> >>>> >>>> “The information transmitted is intended only for the person or entity >>>> to which it is addressed and may contain proprietary, business-confidential >>>> and/or privileged material. If you are not the intended recipient of this >>>> message you are hereby notified that any use, review, retransmission, >>>> dissemination, distribution, reproduction or any action taken in reliance >>>> upon this message is prohibited. If you received this in error, please >>>> contact the sender and delete the material from any computer.” >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ >>>> freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> >>>> “The information transmitted is intended only for the person or entity >>>> to which it is addressed and may contain proprietary, business-confidential >>>> and/or privileged material. If you are not the intended recipient of this >>>> message you are hereby notified that any use, review, retransmission, >>>> dissemination, distribution, reproduction or any action taken in reliance >>>> upon this message is prohibited. If you received this in error, please >>>> contact the sender and delete the material from any computer.” >>>> >>>> _________________________________________________________________________ >>>> 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 >> >> -- >> 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 <(919)%20386-9900> >> >> https://www.youtube.com/watch?v=9XXgW34t40s >> https://www.youtube.com/watch?v=NLaDpGQuZDA >> >> >> _________________________________________________________________________ >> 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.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.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 karl at xtronics.com Sat May 27 04:59:23 2017 From: karl at xtronics.com (Karl Schmidt) Date: Fri, 26 May 2017 23:59:23 -0500 Subject: [Freeswitch-users] Debian Stretch In-Reply-To: References: <000201d2ca8e$a53caeb0$efb60c10$@accra.ca> Message-ID: <1af2ce9c-9dcb-e72a-59bc-d71d6c9ed193@xtronics.com> The key is the wrong type - DSA1024 rather than rsa4096 FS works on stretch - but I'm not doing any encryption. -------------------------------------------------------------------------------- Karl Schmidt EMail Karl at xtronics.com 3209 West 9th Street Ph (785) 841-3089 Lawrence, KS 66049 Nothing defines humans better than their willingness to do irrational things in the pursuit of phenomenally unlikely payoffs. This is the principle behind lotteries, dating, and religion. --Scott Adams -------------------------------------------------------------------------------- From krice at freeswitch.org Sat May 27 16:29:46 2017 From: krice at freeswitch.org (Ken Rice) Date: Sat, 27 May 2017 11:29:46 -0500 Subject: [Freeswitch-users] Maintenance at the FreeSWITCH DataCenter Message-ID: Hey FreeSWITCHers, over the extended holiday weekend, we'll be doing some maint at the DC you may see some services down as we finish realigning things to fully utilize the new hardware we installed last weekend. You can can help us with the data center expenses by hitting the donate button on the homepage @ https://freeswitch.org Thanks Ken Sent from my iPhone From findmeinwland at gmail.com Mon May 29 05:58:02 2017 From: findmeinwland at gmail.com (Artur Mega) Date: Mon, 29 May 2017 10:58:02 +0500 Subject: [Freeswitch-users] Running conference from api and xml-dialplan Message-ID: Hello community, I have a problem with conferences. When I start a conference from xml dialplan like this - everythiing works (plays audio, I'm asked to enter PIN): ``` ``` But when I use fs_cli to call to my mobile number, then joining to conference - I don't hear audio (woman's voice "You are only one person in the conference"), and what is more important - I'm not asked to enter PIN code. Whyy???? ``` originate {origination_caller_id_number=8080000000}sofia/gateway/mygateway/my-mobile-number &conference(test at default-en+1234+flags{endconf|moderator}) ``` I need to use API to join a mobile phone number to conference​, and ask him to enter a PIN number (becouse sometimes phone number can be offline, but owner can turn on a voicemail service, and voicemail will record conversation untill it became full (up to 1 hour)). I don't want to call to voicemail for 1 hour becouse of money. ​ Thanks. -- ​С уважением, ​ Артур ​Regards, Arthur​ -------------- next part -------------- An HTML attachment was scrubbed... URL: From Alexander.Haugg at c4b.de Mon May 29 16:25:03 2017 From: Alexander.Haugg at c4b.de (Alexander Haugg) Date: Mon, 29 May 2017 16:25:03 +0000 Subject: [Freeswitch-users] Turn configuration mod_sofia Message-ID: Hi all, the stun configuration for the mod_sofia profile is very easy "" but what is to do, if i need a relay candidate in the sdp? How can i set the turn address and the login credentials? Thanks a lot -------------- next part -------------- An HTML attachment was scrubbed... URL: From tg-maillistings at level5.de Mon May 29 17:07:46 2017 From: tg-maillistings at level5.de (=?UTF-8?Q?Thorsten_G=c3=b6llner?=) Date: Mon, 29 May 2017 19:07:46 +0200 Subject: [Freeswitch-users] Multi Tenant Installation / problem with call transfer (att_xfer) Message-ID: <22019889-7cf2-e5b0-0ca9-7fea6c76b62c@level5.de> Hi, I am using Freeswitch 1.6.14 and set it up with a multi tenant installation. Some users are in company-a and some users are in company-b. Both companies have their own dialplan. Works fine. Now I want to make an attanded transfer (att_xfer). I make a call from device 1 to device 2. Then I want to initiate a tranfer. So I press *3 followed by the wanted target (say *312 to call 12). Yes, I bound the meta app key in the dialplan. But now I get the following message/error in the log: EXECUTE sofia/internal/12 at my.domain.com att_xfer(user/14 at my_public_ip) The transfer fails. The user 12 is registered in domain my.domain.com but the wanted user cannot be reached because att_xfer tries to reach the user in the "domain" set in $${domain} in vars.xml. In my vars.xml you can find: And in my features.xml you can find: So att_xfer tries to call the user in $${domain}. But it is set to my public ip. I followd the instruction on https://wiki.freeswitch.org/wiki/Multiple_Companies and in the last chapter wou will find the hint to set the domain to any other domain but not to one of the productive domains. So my IP should be fine. I also tried to set the domain variable before in the dialplan via but this does not take any effect. Do I missing something? Can you give me a hint, what the correct way is to make it? Thanks in advance, Thorsten From abaci64 at gmail.com Tue May 30 00:59:38 2017 From: abaci64 at gmail.com (Abaci B) Date: Mon, 29 May 2017 20:59:38 -0400 Subject: [Freeswitch-users] Windows build - mod_lua In-Reply-To: References: Message-ID: I see now that the 32 bit version does have lua but it's version 5.1, would appreciate any help on where I can find a windows build that has Lua 5.2 I also updated https://freeswitch.org/jira/browse/FS-9717 Thanks On Thu, May 25, 2017 at 9:29 PM, Abaci B wrote: > I see that there is no mod_lua in the windows 1.6.17 build, is this > official or a bug? > Thanks > > > On Tue, Nov 8, 2016 at 5:34 PM, Gregor Nanger wrote: > >> I've compiled FS with visual studio and there is no mod_lua.dll >> generated. Hence I get error when FS starts that it can not find dll. >> >> Windows build went fine (with excluded modv8) >> >> Anyone also find this problem? >> >> Best regards, 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 >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From hardikitpl at gmail.com Tue May 30 04:49:32 2017 From: hardikitpl at gmail.com (Hardik Patel) Date: Tue, 30 May 2017 10:19:32 +0530 Subject: [Freeswitch-users] Fax receive issue with t30 codec Message-ID: Hello, I am using opensips as entry point using dispatcher. opensips( 127.0.0.1), i am routing call to freeswitch server (127.0.0.3). Now I am trying to receive fax, my issue is when i try to send fax in softphone(Zoiper) from the log i am seeing that it is sending fax using t30 codec. and i am not receiving the fax at destination, is it because of codec, should it only work with t38 codec? if that is the issue than how am i be able to send the fax using t38 from zoiper? Here i am attaching the fs log with loglevel 9 and sip trace is also enabled. 127.0.0.2 => carrier/provider IP 123456789 => Fax number test at gamil.com => Email Address 127.0.0.4 =>UI IP Pastebin link:https://pastebin.freeswitch.org/view/9ec52715 -- Hardik Patel iNextrix Technologies Pvt Ltd -------------- next part -------------- An HTML attachment was scrubbed... URL: From tculjaga at gmail.com Tue May 30 12:40:56 2017 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Tue, 30 May 2017 14:40:56 +0200 Subject: [Freeswitch-users] originated_legs Message-ID: Hello, There is something that is bugging me pretty hard and i need to understand how variable_originated_legs in CHANNEL_HANGUP gets populated ? example: variable_originated_legs: ARRAY::faf71922-4526-11e7-b0f4-9fb65fdc7341;Outbound Call;888|:faf71922-4526-11e7-b0f4-9fb65fdc7341;Outbound Call;888|:faf76c56-4526-11e7-b0fd-9fb65fdc7341;Outbound Call;0916331550", the reason :=) i have a scenario like this: 1. Incoming calls (IN_CALL) are sent to PARK application 2. On CHANNEL_PARK i originate a call to an extension say EXT1 e.g. originate {ogirination_uuid=}user/1002 & park() ... and eventually i do uuid_bridge 3. EXT1 makes a call transfer to EXT2 (using att_xfer) and hangs up the call 4. on CHANNEL_HANGUP from EXT1 i learn EXT2 uuid and.,,, IN_CALL is talking to EXT2 and we are happy :=) ... this is all good if the transfer destination is a single extension (i get enough info from signal_bond etc...). But if i have multiple destination behind EXT2 e.g. user/1002,user/1003,sofia/gateway/gw1/012345 the thing gets complicated. I was hoping i could exploit variable_originated_leg in CHANNEL_HANGUP from EXT1 to learn all origination legs uuid after an att_xfer to a multiple destination. is that feasible ? I mean, will this variable contain all UUIDs att_xfer generated on a multi-destination transfer ? please i would appreciate any heads up here :=) thanks, Tihomir. -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.org Tue May 30 13:48:59 2017 From: brian at freeswitch.org (Brian West) Date: Tue, 30 May 2017 08:48:59 -0500 Subject: [Freeswitch-users] Fax receive issue with t30 codec In-Reply-To: References: Message-ID: You'll want to use 1.6.17 I fixed a bit of issues since 1.6.8 in relation to faxing. /b On Mon, May 29, 2017 at 11:49 PM, Hardik Patel wrote: > Hello, > > I am using opensips as entry point using dispatcher. opensips( 127.0.0.1), > i am routing call to freeswitch server (127.0.0.3). > > Now I am trying to receive fax, my issue is when i try to send fax in > softphone(Zoiper) from the log i am seeing that it is sending fax using t30 > codec. and i am not receiving the fax at destination, is it because of > codec, should it only work with t38 codec? if that is the issue than how am > i be able to send the fax using t38 from zoiper? > > Here i am attaching the fs log with loglevel 9 and sip trace is also > enabled. > > 127.0.0.2 => carrier/provider IP > 123456789 => Fax number > test at gamil.com => Email Address > 127.0.0.4 =>UI IP > > Pastebin link:https://pastebin.freeswitch.org/view/9ec52715 > > -- > Hardik Patel > iNextrix Technologies Pvt Ltd > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Book a phone call (CST) Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.org Tue May 30 13:50:32 2017 From: brian at freeswitch.org (Brian West) Date: Tue, 30 May 2017 08:50:32 -0500 Subject: [Freeswitch-users] Turn configuration mod_sofia In-Reply-To: References: Message-ID: There is little reason to use TURN when speaking to FreeSWITCH, What issue are you trying to solve? On Mon, May 29, 2017 at 11:25 AM, Alexander Haugg wrote: > Hi all, > > > > the stun configuration for the mod_sofia profile is very easy „ name="ext-rtp-ip" value=" stun:stun.freeswitch.org"/>“ > > but what is to do, if i need a relay candidate in the sdp? How can i set > the turn address and the login credentials? > > > > 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 > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Book a phone call (CST) Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.org Tue May 30 15:44:34 2017 From: brian at freeswitch.org (Brian West) Date: Tue, 30 May 2017 10:44:34 -0500 Subject: [Freeswitch-users] Testing Message-ID: Oh and don't forget to register for ClueCon :) -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Book a phone call (CST) Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: From hunterj91 at hotmail.com Tue May 30 19:27:13 2017 From: hunterj91 at hotmail.com (Jonathan Hunter) Date: Tue, 30 May 2017 19:27:13 +0000 Subject: [Freeswitch-users] mod_voicemail greeting_path file permissions Message-ID: Hi Guys, Hope all are well? With mod_voicemail, can the greeting_path be set as a variable? And if so can the file permissions also be defined? I see there was a patch for file_path sometime ago but just wondered if the same can be set for greeting_path, as I need to allow both apache and freeswitch to access the same directory and fileset and permissions seems to be playing a part. thanks jon -------------- next part -------------- An HTML attachment was scrubbed... URL: From lte at lte-net.de Wed May 31 06:50:43 2017 From: lte at lte-net.de (Fred Schulz) Date: Wed, 31 May 2017 08:50:43 +0200 (CEST) Subject: [Freeswitch-users] Codec list after restarting freeswitch Message-ID: <296804635.106943.1496213443218@ox.hosteurope.de> An HTML attachment was scrubbed... URL: From matt at supportedbusiness.com Wed May 31 07:36:01 2017 From: matt at supportedbusiness.com (Matt Broad) Date: Wed, 31 May 2017 08:36:01 +0100 Subject: [Freeswitch-users] test Message-ID: -------------- next part -------------- An HTML attachment was scrubbed... URL: From deepikay at iiitd.ac.in Wed May 31 08:34:22 2017 From: deepikay at iiitd.ac.in (Deepika Yadav) Date: Wed, 31 May 2017 14:04:22 +0530 Subject: [Freeswitch-users] Subscription to different conferences at a time Message-ID: Hi, I have a scenario where mutliple conferences are hosted through Freeswitch. I programmed a python thread that subscribes to Freeswitch conference events using following filters: freeswitch_connection.events('plain', 'all') if event_name == "CHANNEL_ANSWER": ---do something...... if event_name == "CUSTOM": action = conference_event.getHeader('Event-Subclass') if action == 'conference::maintenance': conf_action = conference_event.getHeader('Action') if conf_action == 'conference-destroy': elif (conf_action == 'unmute-member'): This works perfect only when there is one conference going on. I am worried when mutiple conferences will be there then this subscription to events of other conferences also. Is there any way for subscribing to conferences based on their IDs or uuids? Regards Deepika https://www.iiitd.edu.in/~deepikay/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From rick at magicmail.mooo.com Wed May 31 13:57:38 2017 From: rick at magicmail.mooo.com (Rick Jarvis) Date: Wed, 31 May 2017 14:57:38 +0100 Subject: [Freeswitch-users] FS to FS Message-ID: <5BD3EEFB-4F3C-4E14-B80C-1CED60244C5E@magicmail.mooo.com> Sorry if this is a dumb question, but it’s something I’ve always struggled with that I’m convinced should be quite straightforward. Getting one FS to register with another as an extension (I know there are other ways of getting FS to talk together, but this is the situation I’m faced with). Box one (the host) is not behind NAT. Box two (the client) IS behind NAT. In my mind it should be good enough to have a directory entry on box one, and a SIP profile on box two, then something like a catchall dial plan entry on box 2 to accept the calls made to the ‘extension’ it’s registered as and handle the call accordingly. But I don’t think it’s that simple? I’ve looked through the soft phone configuration, which suggests using a directory entry with the register flag set to true, but this doesn’t seem to help me. Thanks R From mike at jerris.com Wed May 31 16:03:47 2017 From: mike at jerris.com (Michael Jerris) Date: Wed, 31 May 2017 12:03:47 -0400 Subject: [Freeswitch-users] Mailing list issues. Message-ID: <9D2D63A8-D6CF-49CC-A0E9-620334870B75@jerris.com> It looks like the mailing list was having issues stemming from our recent server maintenance. It should be corrected now. Mike From ar at cyberfonica.com Wed May 31 16:28:45 2017 From: ar at cyberfonica.com (Alejandro Recarey) Date: Wed, 31 May 2017 18:28:45 +0200 Subject: [Freeswitch-users] mod_event_socket 'linger timeout, closing socket' Message-ID: Hi all, thanks for taking the time to read, I'm using an outbound socket server (plivo open source) to control my FS calls. The problem I'm having is that when using the 'record' command sometimes the outbound socket app is not receiving the events that signal the end of the recording. This is a somewhat random event. Sometimes it works perfectly, and sometimes it doesnt. In the cases where it doesnt work, I get the following message in the FS console: 2017-05-30 16:55:53.013021 [DEBUG] mod_event_socket.c:1431 linger timeout, closing socket I added a little bit of extra logging to mod_event_socket.c and recompiled and I got this info: 2017-05-31 12:46:47.009854 [DEBUG] mod_event_socket.c:1431 Extra Debug: Linger timeout: 1496234806 - Epoch Time Now: 1496234807 2017-05-31 12:46:47.009854 [DEBUG] mod_event_socket.c:1432 linger timeout, closing socket I wanted to print those two variables, because they're used in mod_event_socket in the if that determines whether to throw the timeout error or not: if (switch_test_flag(listener, LFLAG_HANDLE_DISCO) && listener->linger_timeout != (time_t) -1 && switch_epoch_time_now(NULL) > listener->linger_timeout) { switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(listener->session), SWITCH_LOG_DEBUG, "linger timeout, closing socket\n"); status = SWITCH_STATUS_FALSE; break; } Notice that if the epoch has advanced just 1 second (hence epoch > linger_timeout) the socket is dropped. I would hope that the socket would linger for longer than one second. Is there any way to raise this timeout? Thanks again, Alex -------------- next part -------------- An HTML attachment was scrubbed... URL: From ar at cyberfonica.com Wed May 31 16:52:49 2017 From: ar at cyberfonica.com (Alejandro Recarey) Date: Wed, 31 May 2017 18:52:49 +0200 Subject: [Freeswitch-users] mod_event_socket 'linger timeout, closing socket' In-Reply-To: References: Message-ID: My bad, I just figured out that linger can take arguments. I was sending some corrupted data (a space and the null character) and FS took it as a 0, killing the linger on the first epoch change. I set it to -1 to get unlimited linger. Quickly fixed and working, thanks again! On Wed, May 31, 2017 at 6:28 PM, Alejandro Recarey wrote: > Hi all, thanks for taking the time to read, > > I'm using an outbound socket server (plivo open source) to control my FS > calls. The problem I'm having is that when using the 'record' command > sometimes the outbound socket app is not receiving the events that signal > the end of the recording. This is a somewhat random event. Sometimes it > works perfectly, and sometimes it doesnt. > > In the cases where it doesnt work, I get the following message in the FS > console: > > 2017-05-30 16:55:53.013021 [DEBUG] mod_event_socket.c:1431 linger timeout, > closing socket > > I added a little bit of extra logging to mod_event_socket.c and recompiled > and I got this info: > > 2017-05-31 12:46:47.009854 [DEBUG] mod_event_socket.c:1431 Extra Debug: > Linger timeout: 1496234806 - Epoch Time Now: 1496234807 > 2017-05-31 12:46:47.009854 [DEBUG] mod_event_socket.c:1432 linger timeout, > closing socket > > I wanted to print those two variables, because they're used in > mod_event_socket in the if that determines whether to throw the timeout > error or not: > > if (switch_test_flag(listener, LFLAG_HANDLE_DISCO) && > listener->linger_timeout != (time_t) -1 && switch_epoch_time_now(NULL) > > listener->linger_timeout) { > switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(listener->session), > SWITCH_LOG_DEBUG, "linger timeout, closing socket\n"); > status = SWITCH_STATUS_FALSE; > break; > } > > Notice that if the epoch has advanced just 1 second (hence epoch > > linger_timeout) the socket is dropped. > > I would hope that the socket would linger for longer than one second. Is > there any way to raise this timeout? > > > Thanks again, > > Alex > -------------- next part -------------- An HTML attachment was scrubbed... URL: From hunterj91 at hotmail.com Wed May 31 19:24:27 2017 From: hunterj91 at hotmail.com (Jonathan Hunter) Date: Wed, 31 May 2017 19:24:27 +0000 Subject: [Freeswitch-users] Monitoring call quality using variable_rtp_audio_xxx Message-ID: Hi Guys, Sorry for the noise, we are looking to poll calls in progress to grab the rtp_audio variables when a call is in progress, and we want to understand if this is a good approach and to that end what values/variables we should consider and what ranges should we be working with? I understand mos and quality percentage but what other values are a good indicator? Again Im clear on jitter and its meaning just want to understand whats what as not everything understandably is not documented. Many thanks Jon "variable_rtp_audio_recv_pt": "8", "variable_rtp_audio_in_raw_bytes": "374960", "variable_rtp_audio_in_media_bytes": "374616", "variable_rtp_audio_in_packet_count": "2180", "variable_rtp_audio_in_media_packet_count": "2178", "variable_rtp_audio_in_skip_packet_count": "92", "variable_rtp_audio_in_jitter_packet_count": "0", "variable_rtp_audio_in_dtmf_packet_count": "0", "variable_rtp_audio_in_cng_packet_count": "0", "variable_rtp_audio_in_flush_packet_count": "2", "variable_rtp_audio_in_largest_jb_size": "0", "variable_rtp_audio_in_jitter_min_variance": "28.57", "variable_rtp_audio_in_jitter_max_variance": "116.33", "variable_rtp_audio_in_jitter_loss_rate": "0.02", "variable_rtp_audio_in_jitter_burst_rate": "0.98", "variable_rtp_audio_in_mean_interval": "20.39", "variable_rtp_audio_in_flaw_total": "43", "variable_rtp_audio_in_quality_percentage": "97.00", "variable_rtp_audio_in_mos": "4.47", "variable_rtp_audio_out_raw_bytes": "210012", "variable_rtp_audio_out_media_bytes": "210012", "variable_rtp_audio_out_packet_count": "1221", "variable_rtp_audio_out_media_packet_count": "1221", "variable_rtp_audio_out_skip_packet_count": "0", "variable_rtp_audio_out_dtmf_packet_count": "0", "variable_rtp_audio_out_cng_packet_count": "0", "variable_rtp_audio_rtcp_packet_count": "0", "variable_rtp_audio_rtcp_octet_count": "0", -------------- next part -------------- An HTML attachment was scrubbed... URL: From krice at freeswitch.org Wed May 31 20:22:46 2017 From: krice at freeswitch.org (Ken Rice) Date: Wed, 31 May 2017 15:22:46 -0500 Subject: [Freeswitch-users] testing 1234 Message-ID: <1aa001d2da4b$ab19c1c0$014d4540$@freeswitch.org> This is just a test -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.org Wed May 31 20:25:33 2017 From: brian at freeswitch.org (Brian West) Date: Wed, 31 May 2017 15:25:33 -0500 Subject: [Freeswitch-users] testing 1234 In-Reply-To: <1aa001d2da4b$ab19c1c0$014d4540$@freeswitch.org> References: <1aa001d2da4b$ab19c1c0$014d4540$@freeswitch.org> Message-ID: I got this. /b On Wed, May 31, 2017 at 3:22 PM, Ken Rice wrote: > This is just a test > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Book a phone call (CST) Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: From krice at freeswitch.org Wed May 31 20:30:32 2017 From: krice at freeswitch.org (Ken Rice) Date: Wed, 31 May 2017 15:30:32 -0500 Subject: [Freeswitch-users] testing 1234 In-Reply-To: References: <1aa001d2da4b$ab19c1c0$014d4540$@freeswitch.org> Message-ID: <1aaa01d2da4c$c09e3890$41daa9b0$@freeswitch.org> Really now From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Wednesday, May 31, 2017 3:26 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] testing 1234 I got this. /b On Wed, May 31, 2017 at 3:22 PM, Ken Rice > wrote: This is just a test _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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.org Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com Book a phone call (CST) Allison prompts for FreeSWITCH: https://www.gofundme.com/allison-prompts-for-freeswitch Got Bugs? Report them here! | Reddit: /r/freeswitch T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) Skype:briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: From krice at freeswitch.org Wed May 31 21:27:52 2017 From: krice at freeswitch.org (Ken Rice) Date: Wed, 31 May 2017 16:27:52 -0500 Subject: [Freeswitch-users] testing 1234 In-Reply-To: References: <1aa001d2da4b$ab19c1c0$014d4540$@freeswitch.org> Message-ID: <1abd01d2da54$c34b84a0$49e28de0$@freeswitch.org> You got this? From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Wednesday, May 31, 2017 3:26 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] testing 1234 I got this. /b On Wed, May 31, 2017 at 3:22 PM, Ken Rice > wrote: This is just a test _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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.org Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com Book a phone call (CST) Allison prompts for FreeSWITCH: https://www.gofundme.com/allison-prompts-for-freeswitch Got Bugs? Report them here! | Reddit: /r/freeswitch T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) Skype:briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: From admin at tedssupply.com Wed May 31 22:01:02 2017 From: admin at tedssupply.com (admin) Date: Wed, 31 May 2017 18:01:02 -0400 Subject: [Freeswitch-users] testing 1234 In-Reply-To: <1abd01d2da54$c34b84a0$49e28de0$@freeswitch.org> References: <1aa001d2da4b$ab19c1c0$014d4540$@freeswitch.org> <1abd01d2da54$c34b84a0$49e28de0$@freeswitch.org> Message-ID: <592F04DE020000310000A092@mail.tedssupply.com> Loud and clear! >>> "Ken Rice" 05/31/17 5:33 PM >>> You got this? From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Wednesday, May 31, 2017 3:26 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] testing 1234 I got this. /b On Wed, May 31, 2017 at 3:22 PM, Ken Rice wrote: This is just a test _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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.org Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com Book a phone call (CST) Allison prompts for FreeSWITCH: https://www.gofundme.com/allison-prompts-for-freeswitch Got Bugs? Report them here! | Reddit: /r/freeswitch T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) Skype:briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: From john at mobex.biz Wed May 31 22:15:59 2017 From: john at mobex.biz (John Dalrymple) Date: Wed, 31 May 2017 18:15:59 -0400 Subject: [Freeswitch-users] testing 1234 In-Reply-To: <592F04DE020000310000A092@mail.tedssupply.com> References: <1aa001d2da4b$ab19c1c0$014d4540$@freeswitch.org> <1abd01d2da54$c34b84a0$49e28de0$@freeswitch.org> <592F04DE020000310000A092@mail.tedssupply.com> Message-ID: unsubscribe (please) John Dalrymple President 813-990-0996 call or text On Wed, May 31, 2017 at 6:01 PM, admin wrote: > Loud and clear! > > > >>> "Ken Rice" 05/31/17 5:33 PM >>> > > You got this? > > > > *From:* FreeSWITCH-users [mailto:freeswitch-users- > bounces at lists.freeswitch.org] *On Behalf Of *Brian West > *Sent:* Wednesday, May 31, 2017 3:26 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] testing 1234 > > > > I got this. > > > > /b > > > > > > On Wed, May 31, 2017 at 3:22 PM, Ken Rice wrote: > > This is just a test > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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.org > > *Twitter: @FreeSWITCH , @briankwest* > > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > Book a phone call (CST) > > Allison prompts for FreeSWITCH: > > *https://www.gofundme.com/allison-prompts-for-freeswitch* > > > Got Bugs? Report them here ! | Reddit: > /r/freeswitch > > *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002> > | *M:*+1918424WEST (9378) > *Skype:*briankwest > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 matt at supportedbusiness.com Wed May 31 22:20:22 2017 From: matt at supportedbusiness.com (Matt Broad) Date: Wed, 31 May 2017 23:20:22 +0100 Subject: [Freeswitch-users] testing 1234 In-Reply-To: References: <1aa001d2da4b$ab19c1c0$014d4540$@freeswitch.org> <1abd01d2da54$c34b84a0$49e28de0$@freeswitch.org> <592F04DE020000310000A092@mail.tedssupply.com> Message-ID: roger roger Matt Broad Tel: +44 (0)203 011 1313 <+44%2020%203011%201313> Web: www.supportedbusiness.com On 31 May 2017 at 23:15, John Dalrymple wrote: > unsubscribe (please) > > John Dalrymple > President > 813-990-0996 <(813)%20990-0996> call or text > > > > > > On Wed, May 31, 2017 at 6:01 PM, admin wrote: > >> Loud and clear! >> >> >> >>> "Ken Rice" 05/31/17 5:33 PM >>> >> >> You got this? >> >> >> >> *From:* FreeSWITCH-users [mailto:freeswitch-users-bounc >> es at lists.freeswitch.org] *On Behalf Of *Brian West >> *Sent:* Wednesday, May 31, 2017 3:26 PM >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] testing 1234 >> >> >> >> I got this. >> >> >> >> /b >> >> >> >> >> >> On Wed, May 31, 2017 at 3:22 PM, Ken Rice wrote: >> >> This is just a test >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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.org >> >> *Twitter: @FreeSWITCH , @briankwest* >> >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> Book a phone call (CST) >> >> Allison prompts for FreeSWITCH: >> >> *https://www.gofundme.com/allison-prompts-for-freeswitch* >> >> >> Got Bugs? Report them here ! | Reddit: >> /r/freeswitch >> >> *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002> >> | *M:*+1918424WEST (9378) >> *Skype:*briankwest >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at gmail.com Wed May 31 22:20:44 2017 From: david.villasmil at gmail.com (David Villasmil Govea) Date: Wed, 31 May 2017 18:20:44 -0400 Subject: [Freeswitch-users] testing 1234 In-Reply-To: References: <1aa001d2da4b$ab19c1c0$014d4540$@freeswitch.org> <1abd01d2da54$c34b84a0$49e28de0$@freeswitch.org> <592F04DE020000310000A092@mail.tedssupply.com> Message-ID: yep ᐧ 2017-05-31 18:15 GMT-04:00 John Dalrymple : > unsubscribe (please) > > John Dalrymple > President > 813-990-0996 call or text > > > > > > On Wed, May 31, 2017 at 6:01 PM, admin wrote: > >> Loud and clear! >> >> >> >>> "Ken Rice" 05/31/17 5:33 PM >>> >> >> You got this? >> >> >> >> *From:* FreeSWITCH-users [mailto:freeswitch-users-bounc >> es at lists.freeswitch.org] *On Behalf Of *Brian West >> *Sent:* Wednesday, May 31, 2017 3:26 PM >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] testing 1234 >> >> >> >> I got this. >> >> >> >> /b >> >> >> >> >> >> On Wed, May 31, 2017 at 3:22 PM, Ken Rice wrote: >> >> This is just a test >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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.org >> >> *Twitter: @FreeSWITCH , @briankwest* >> >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> Book a phone call (CST) >> >> Allison prompts for FreeSWITCH: >> >> *https://www.gofundme.com/allison-prompts-for-freeswitch* >> >> >> Got Bugs? Report them here ! | Reddit: >> /r/freeswitch >> >> *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002> >> | *M:*+1918424WEST (9378) >> *Skype:*briankwest >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 > -- DVG -- Imagination is more important than knowledge Albert Einstein -------------- next part -------------- An HTML attachment was scrubbed... URL: From denys.pozniak at crazycall.com Mon May 29 13:43:49 2017 From: denys.pozniak at crazycall.com (Denys Pozniak) Date: Mon, 29 May 2017 15:43:49 +0200 Subject: [Freeswitch-users] question about HA solution Message-ID: Hello! I built FS HA solution based on keepalived and mysql master-master. It works ok generally, but as I understand FS after restarting cleaning own database. So when node1 fails calls jump to node2, after script restarts node1 it is not possible to move calls back. Tried options in switch.conf.xml, but no luck: Is there is a way to solve this? BR, Denys -------------- next part -------------- An HTML attachment was scrubbed... URL: From krice at freeswitch.org Wed May 31 22:26:52 2017 From: krice at freeswitch.org (Ken Rice) Date: Wed, 31 May 2017 17:26:52 -0500 Subject: [Freeswitch-users] testing 1234 In-Reply-To: References: <1aa001d2da4b$ab19c1c0$014d4540$@freeswitch.org> <1abd01d2da54$c34b84a0$49e28de0$@freeswitch.org> <592F04DE020000310000A092@mail.tedssupply.com> Message-ID: <1b0c01d2da5d$011a7310$034f5930$@freeswitch.org> Thanks for replying everyone. And just a reminder for those that are still on the list and want to unsubscribe, or just want to manage their list membership, there is a link at the bottom of every email that comes across the list that you can click. It will allow you to unsubscribe, or even change you from individual emails on the list to the digest where you get 1 email a day with all the other emails aggregated up. Thanks Guys! Ken From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Matt Broad Sent: Wednesday, May 31, 2017 5:20 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] testing 1234 roger roger Matt Broad Tel: +44 (0)203 011 1313 Web: www.supportedbusiness.com On 31 May 2017 at 23:15, John Dalrymple > wrote: unsubscribe (please) John Dalrymple President 813-990-0996 call or text On Wed, May 31, 2017 at 6:01 PM, admin > wrote: Loud and clear! >>> "Ken Rice" > 05/31/17 5:33 PM >>> You got this? From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org ] On Behalf Of Brian West Sent: Wednesday, May 31, 2017 3:26 PM To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] testing 1234 I got this. /b On Wed, May 31, 2017 at 3:22 PM, Ken Rice > wrote: This is just a test _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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.org Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com Book a phone call (CST) Allison prompts for FreeSWITCH: https://www.gofundme.com/allison-prompts-for-freeswitch Got Bugs? Report them here! | Reddit: /r/freeswitch T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) Skype:briankwest _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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: