March 2017 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Dockerfile for FreeSWITCH
by Brian West 18 Aug '17

18 Aug '17

https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/docker If anyone wishes to submit pull requests to improve these please do so at freeswitch.org/jira I would like to work with the community to improve and document the use cases correct. Thanks, -- *Brian West* brian(a)freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* <https://www.gofundme.com/allison-prompts-for-freeswitch> Wish to schedule a meeting? http://app.timebridge.com/#/meet/freeswitch Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest

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Forward fails with deflect and REFER to null
by Mario G 12 Aug '17

12 Aug '17

What are the bold deflect and refer to lines below telling me? This worked until last year, not sure if it’s related to 1.6, I can’t nail down a start date. A ring group rings. When one phone’s FWD button is pressed prior to a call or during a call the call is forwarded to VM extension 2921, and the other phones stopped ringing. This no longer works, the set phone does not ring, the call is not transferred to VM and the other phones keep ringing. I couldn’t find anything on the wiki (it only described using codes to turn FWD on/off), etc. to explain why this stopped working. Any help to get the phone forwarding working again is greatly appreciated. Thanks! Mario G YEALINK T48G DSS button is set to forward to 2921, pressing again turns it off. All other DSS keys such as call park, etc work fine. <extension name="vmonly"> <condition field="destination_number" expression="^2921$">  <action application="lua" data="/usr/local/FreeSwitch-scripts/known-people.lua CALLIN"/> <action application="answer"/> <action application="playback" data="${lua_vmmsg}"/>  <action application="set" data="skip_greeting=true"/> <action application="voicemail" data="default ${domain_name} 100"/> </condition> </extension> This log shows the only lines that contain 2921 in bold: 2016-11-19 16:27:26.673609 [INFO] mod_dialplan_xml.c:637 Processing Mario G <15051234567>->2921 in context public Dialplan: sofia/internal/15051234567(a)11.123.1.10 parsing [public->unloop] continue=false Dialplan: sofia/internal/15051234567(a)11.123.1.10 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/15051234567(a)11.123.1.10 Regex (PASS) [unloop] ${sip_looped_call}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/15051234567(a)11.123.1.10 Action deflect(${destination_number}) 2016-11-19 16:27:26.673609 [DEBUG] switch_core_state_machine.c:286 (sofia/internal/15051234567(a)11.123.1.10) State Change CS_ROUTING -> CS_EXECUTE 2016-11-19 16:27:26.673609 [DEBUG] switch_core_state_machine.c:643 (sofia/internal/15051234567(a)11.123.1.10) State ROUTING going to sleep 2016-11-19 16:27:26.673609 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/15051234567(a)11.123.1.10) Running State Change CS_EXECUTE 2016-11-19 16:27:26.673609 [DEBUG] switch_core_state_machine.c:650 (sofia/internal/15051234567(a)11.123.1.10) State EXECUTE 2016-11-19 16:27:26.673609 [DEBUG] mod_sofia.c:198 sofia/internal/15051234567(a)11.123.1.10 SOFIA EXECUTE 2016-11-19 16:27:26.673609 [DEBUG] switch_core_state_machine.c:328 sofia/internal/15051234567(a)11.123.1.10 Standard EXECUTE EXECUTE sofia/internal/15051234567(a)11.123.1.10 deflect(2921) 2016-11-19 16:27:26.673609 [DEBUG] sofia.c:8349 Process REFER to [(null)@2921]

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FreeSWITCH Week in Review (Master Branch) December 5th – December 12th
by Ken Rice 15 Jun '17

15 Jun '17

New Post on freeswitch.org from Kathleen King check it out at http://ift.tt/1P8dyo3 FreeSWITCH Week in Review (Master Branch) December 5th – December 12th This week the verto communicator had some new updates to the administrator menu and the core added a new origination_audio_mode variable. Join us Wednesdays at 12:00 CT for some more FreeSWITCH fun! This week we have Italo Rossi and the Evolux call center team! And, head over to freeswitch.com to learn more about FreeSWITCH support. New features that were added: FS-8616 [verto_communicator] A new menu for moderator, added gain buttons, and removed the 3-dot-button, moving its behavior to member-name div FS-8632 [core] Add origination_audio_mode originate variable with options for sendonly, recvonly or sendrecv Improvements in build system, cross platform support, and packaging: FS-8293 [verto] Add quality level 0 to conference (default is 1) and fix some logic in auto bandwidth The following bugs were squashed: FS-8625 [core] Fixed a segfault caused by an external incoming call from Google Voice. FS-8642 [core] Fixed CF_VIDEO_READY being set on non-video calls FS-8603 [verto_communicator] Added device validation to prevent lost microphones after reset FS-8640 [verto_communicator] Don’t clear conference member reservation id on members that don’t have a reservation ID FS-8633 [mod_verto] Fix for the first verto to join a conference does not get “conference-livearray-join” event FS-8621 [mod_av] Fixed H264 HD1080P video quality issues FS-8631 [mod_db] Updated the regex to allow DSN to match the rest of FS code FS-8643 [mod_sofia] Fixed some memory leaks

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Open Source SIP\WebRTC clients compatible with FreeSWITCH
by Shaun Stokes 09 Jun '17

09 Jun '17

Hi All, Does anyone have any recommendations on a good open source SIP\WebRTC client which works on multiple platforms (Windows, Mac, Linux, Mobiles) to provide presence, voice, video, instant messaging, screen sharing and file sharing? This must be capable of integrating with FreeSWITCH for voice and video (presence via FreeSWITCH would be an advantage). Many Thanks, Shaun [http://www.itec-support.co.uk/wp-content/uploads/2016/07/email_logo.jpg] Shaun Stokes - Infrastructure Analyst T : 01453 700713 E : shaun.stokes(a)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. ______________________________________________________________________

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Windows build - mod_lua
by Gregor Nanger 30 May '17

30 May '17

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

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configure playback outbound http proxy
by Anonim Stefan 25 May '17

25 May '17

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. Thank you, Stefan

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More pixelated video canvas on larger number of users
by Chad Phillips 05 May '17

05 May '17

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

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multiple sip profiles
by Rick Jarvis 06 Apr '17

06 Apr '17

Trying to get my head around setting TLS up. I have one internal sip profile, but multiple domains & dialplans. If I want to enable TLS on multiple domains, I think I’m right in saying that I will then need multiple sip profiles, in order to set the TLS up in each? I guess my confusion comes from understanding the difference between domains, profiles and sip profiles. I’m not even sure exactly what question I’m asking here, so apologies!

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a1 password gateway
by Gregor Nanger 05 Apr '17

05 Apr '17

Can I use a1-hash parameter in gateways configuration? Provider request to send hashed password they sent instead. Best regards, Gregor

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BEHAVIOR CHANGE (YES PAY ATTENTION TO THIS IF YOU USE FAXING)
by Brian West 05 Apr '17

05 Apr '17

FreeSWITCHers, In 1.6.14 we did a slight behavior change related to T.38 re-invite acceptance. Before 1.6.14 we would by default accept the re-invite even if t38 wasn't enabled and switch to UDPTL mode. This was an incorrect behavior to do by default. But one caveat in 1.6.14, We by default 488 the fax but switch the media mode to UDPTL which is also a bad behavior fixed in (FS-9943) slated for 1.6.15. If you wish to retain the previous behavior you can set fax_enable_t38=true in your vars.xml, You can also not set that and just set refuse_t38=true if you wish to never do t.38. These default behaviors are fixed in our next release 1.6.15. Thanks, -- *Brian West* brian(a)freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com https://www.gofundme.com/freeswitch_ubuntu Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest

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