January 2017 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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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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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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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Re: [Freeswitch-users] Fwd: hook to event or startup script which will hook to events?
by Mimiko 27 Mar '17

27 Mar '17

On 14.02.2016 02:38, Abdul Hakeem wrote: > Hello Mimiko, > Did you find any resolution to the issue ? > Cheers, > AH No, no answers to my questions. I use start-up script in lua.conf and bind to event in order to avoid segfault. -- Mimiko desu.

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FreeSWITCH blog RSS feed problems
by Daniel Pocock 15 Mar '17

15 Mar '17

Hi, I had been following the FreeSWITCH blog using the RSS feed https://freeswitch.org/feed/ but my feed reader is complaining with error 404 now when accessing that URL. It looks like the web site has been re-arranged and the blog page doesn't have the RSS or Atom icons nor does it have the meta-data for them in the page. What is the new URL for RSS? Maybe it would be a good idea to put a redirect in the web server so you don't lose followers? Regards, Daniel

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Re: [Freeswitch-users] one way speech path after attempting voice path redirecting in new invite
by Bejan, Alex 15 Feb '17

15 Feb '17

Hello, This is my first posting into this list although I monitored activities in the past. Freeswitch behaves great in our environment - thank you all who contribute! We started recently observing some one way speech path issues in the following scenario and I would like to ask for suggestions, please: We have incoming calls from provider to FS that further bridge to different Standard SIP PBX. Call is established to SIP device connected on the PBX. The PBX immediately invites back attempting to redirect rtp path to start recording the call. FS starts sending RTP to the new destination. There are 2 different very similar PBX environments connected to the same FS: - In one the PBX starts sending RTP even before the connection is ACKed. Recording succeeds. - In the second one, PBX starts sending RTP almost one second later. It seems FS reverts back to the original speech path, while the other devices remain unaware, resulting in one way speech path. The rtp-timeout-sec and rtp-hold-timeout-sec are on the default values  <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> Are there any other parameters that could have such an effect? Your suggestions on further troubleshooting or how to make this work even when RTP is not being received will be greatly appreciated! Thank you very much! Regards

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A note from the Lead Developer. FreeSWITCH 1.8 Feature Requests
by Anthony Minessale 14 Feb '17

14 Feb '17

Hi, We are having an annual gathering to decide the roadmap for upcoming FreeSWITCH 1.8 Now is your chance to propose some feature requests in this thread. No guarantees, but this is the best time during our release cycle to daydream about the future. We hope to have a first cut of 1.8 in time for ClueCon 2017 so register early, the price is the lowest it will ever be during the Early Bird pricing offer lasting only a few more weeks. Also, Brian has started a fund to raise money to provide the team with meals and offset travel expenses if possible during the upcoming summit in little over a week from now. Its one really easy way to show your appreciation for another year of open source telephony! https://www.gofundme.com/freeswitch-developer-meeting-2017 -- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>* ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900 https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA

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