April 2017 - FreeSWITCH-users - FreeSWITCH Mailing Lists

FreeSWITCH on Raspberry board
by youssef elouam 20 Sep '17

20 Sep '17

hi all ; I am trying install freeSWITCH in raspberry board (Raspbian OS). but when I run "make && make install" I get as result : make: *** No targets. Stop. could you please help fix this issue. thanks in advance Bestregards --------------------------Youssef ELOUAM.

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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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from Asterisk sip.conf to FS
by Valter Nogueira 06 Jul '17

06 Jul '17

I have an app that runs in Asterisk and I am trying port it to FS. Below, I pasted a small sip.conf that sums up all my needs. 1. We register out in a VOIP_PROVIDER 2. An external asterisk register in our asterisk as CLIENT01 3. Calls from CLIENT01 route to VOIP_PROVIDER 4. Yealink's, with ACL restrictions, also uses VOIP_PROVIDER to external calls 5. VOIP PROVIDER and Yealinks use one NIC (eth0) 6. CLIENT01 uses another NIC (eth1) In Asterisk all configs are keep together in sip.conf. However, in FS it seems I should spread out things in different configs parts. So: Should Yealink's accounts go into /usr/local/freeswitch/conf/directory/default? Should I put VOIP_PROVIDER in sofia's external profile? Should I create an additional external profile to CLIENT01? Where should I define CLIENT01 account to allow it register in? In /usr/local/freeswitch/conf/directory/default? Thanks for any help. Valter === SIP.CONF === [general] disallow=all context=default ; Default context for incoming calls allowoverlap=no ; Disable overlap dialing support. (Default is yes) bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls alwaysauthreject=yes allowguest=no ;voip provider register => 185999:1010@xxx.xxx.xxx.xxx:5060 [VOIP_PROVIDER] type=peer context=FDTRONCO_94 host=xxx.xxx.xxx.xxx port=5060 username=185999 authuser=185999 authname=185999 secret=XXXXXXXXXXXXXX fromdomain=xxx.xxx.xxx.xxx fromuser=185999 insecure=port,invite canreinvite=no nat=no disallow=all allow=ulaw:30 dtmfmode=rfc2833 ignoreregexpire=yes language=pt_BR call-limit=9999 ;EXTERNAL ASTERISK ACCOUNT - CALLS GOES THRU VOIP_PROVIDER [CLIENT01] type=peer context=FDTRONCO_60 host=dynamic port=5060 username=CLIENT01 authuser=CLIENT01 authname=CLIENT01 secret=XXXXXXX insecure=port,invite canreinvite=no nat=yes disallow=all allow=g729 dtmfmode=rfc2833 ignoreregexpire=yes language=pt_BR call-limit=440 ;yealinks ip phones [1000] type=friend secret=XXXX host=dynamic deny=0.0.0.0/0.0.0.0 permit=192.168.0.0/16 username=1000 context=FASTDIALER_RAMAIS callerid=testefone <1000> requirecalltoken=no nat=no canreinvite=no qualify=yes disallow=all allow=alaw call-limit=2 dtmfmode=RFC2833 language=pt_BR Callgroup=1 pickupgroup=1 [2012] type=friend secret=XXXX host=dynamic deny=0.0.0.0/0.0.0.0 permit=192.168.0.0/255.255.0.0 username=2012 context=FASTDIALER_RAMAIS callerid=Telefone sem fio <2012> requirecalltoken=no nat=no canreinvite=no qualify=no disallow=all allow=alaw call-limit=2 dtmfmode=RFC2833 language=pt_BR Callgroup=1 pickupgroup=1

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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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Call setup failes on outbound calls
by Babak Yakhchali 23 May '17

23 May '17

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@10.10.10.3 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.4;rport;branch=z9hG4bKeN5gp5S5KrZce Max-Forwards: 69 From: "100" <sip:74413@10.10.10.3>;tag=N69FS63aHU7ta To: <sip:12345678@10.10.10.3> Call-ID: 596dba35-96dd-1235-59bc-00505698455d CSeq: 105485099 INVITE Contact: <sip:gw+trunk_1@10.10.10.4:5060;transport=udp;gw=trunk_1> User-Agent: FreeSWITCH-mod_sofia/1.6.15+git~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" <sip:100@10.10.10.3 >;party=calling;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=z9hG4bKeN5gp5S5KrZce;rport=5060 Call-ID: 596dba35-96dd-1235-59bc-00505698455d From: "100"<sip:74413@10.10.10.3>;tag=N69FS63aHU7ta To: <sip:12345678@10.10.10.3> 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=z9hG4bKeN5gp5S5KrZce;rport=5060 Call-ID: 596dba35-96dd-1235-59bc-00505698455d From: "100"<sip:74413@10.10.10.3>;tag=N69FS63aHU7ta To: <sip:12345678@10.10.10.3>;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@10.10.10.3 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.4;rport;branch=z9hG4bKeN5gp5S5KrZce Max-Forwards: 69 From: "100" <sip:74413@10.10.10.3>;tag=N69FS63aHU7ta To: <sip:12345678@10.10.10.3>;tag=juiky7nh Call-ID: 596dba35-96dd-1235-59bc-00505698455d CSeq: 105485099 ACK Content-Length: 0

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Freeswitch and IoT
by Marcel Haldemann 06 May '17

06 May '17

Hi Guys, is it possible to create a channel using event socket (without calling a number) ? LG, Marcel

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