December 2016 - 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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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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Codec transcoding PCMA to PMCU
by Ketan Kothari 04 Jan '17

04 Jan '17

Hello, I have configured PCMA codec in my zoipper soft-phone(UAC) and tried to call to 123456789 using our gateway twillo(UAS). Gateway only accept PCMU codec. I have also configured absolute codec =PCMU but still freeswitch sending PCMA to twillo(Provider gateway UAS). I have got error Originate Failed. Cause: SERVICE_NOT_IMPLEMENTED in freeswitch. Once i have added PCMU codec in my zoiper(UAC) it's working. [image: Inline image 1] What am i missing here?

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NGINX + Kamailio + FreeSwitch
by Sergey Safarov 03 Jan '17

03 Jan '17

Hello guys I want configure user frendly WebRTC server based on FreeSwitch and SipML5 client. It can be easy done in FreeSwitch and NGINX is bounded to different IP/ports. But if you wants use one IP and 443 port then you will try configre NGINX to proxy all reuests line "/fs-socket/" to FreeSwitch port 7443. It is works fine but FreeSwitch cannot not see real client IP address like folowing. Captured on Amazon server. freeswitch(a)ip-172-31-29-87.us-west-2.compute.internal> sofia status profile internal reg Registrations: ================================================================================================= Call-ID: f1e8c7ca-8f50-4285-fd1a-148d2f1d1b88 User: 23(a)46.218.201.23 Contact: "23" <sips:23@df7jal23ls0d.invalid ;rtcweb-breaker=no;transport=wss;fs_nat=yes;fs_path=sips%3A23%40172.31.29.87%3A37244%3Brtcweb-breaker%3Dno%3Btransport%3Dwss> Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04 Status: Registered(TLS-NAT)(unknown) EXP(2016-12-24 09:42:43) EXPSECS(230) Ping-Status: Reachable Ping-Time: 0.00 Host: ip-172-31-29-87.us-west-2.compute.internal IP: 172.31.29.87 Port: 37244 Auth-User: 23 Auth-Realm: 46.218.201.23 MWI-Account: 23(a)46.218.201.23 Total items returned: 1 ================================================================================================= Displayed real IP address of NGINX Also when received INVITE then variables like network_addr will cantain real IP of NGINX. Then you can try confgire nginx like ng proxy_set_header X-Real-IP $remote_addr; proxy_set_header X-Forwarded-For $proxy_add_x_forwarded_for; But FreeSwitch wants SIP headers "X-AUTH-IP" and "X-AUTH-PORT" in every SIP message not only connection establishing. NGINX not understand SIP messages and cannot do it. Then you will try cofigure Kamailio between NGINX and FreeSwitch. In this case Kamailio can parce http headers and add requred SIP header. Are you can suggest other way to publish FreeSwitch socket on same port with http server?

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Abandoned members ESL problem
by Kamil Nigmatullin 02 Jan '17

02 Jan '17

I am creating some kind of monitoring for callcenter. Found one thing. Whan a Caller (Memeber) joined queue 1, then member-queue-start event appears. But if caller left the queue, and joined in a few minutes later (discard_abandoned_after = 3600) the same queue, then member-queue-start event is not shown. Maybe it is right but how can I find out if memeber rejoined the queue? -- Kamil Nigmatullin Tel: 77272323748 mob: 7 (707) 2517003 Skype: kamil.nigmatullin

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