May 2017 - FreeSWITCH-users - FreeSWITCH Mailing Lists

gsmopen one-way voice issue
by Jeff Chua 24 Sep '17

24 Sep '17

I'm configured freeswitch with gsmopen but can't seem to figure where why I only get voice out from SIP-client to mobile-phone but not voice from mobile to SIP client. When I tried to call from mobile-phone to IVR extension 5000 and press 2 for the standard echo test (9196), I could hear the echo back. When I tried to call 9195 for the five second delay echo test, I couldn't anything. I've add these settings in vars.xml as well. <X-PRE-PROCESS cmd="set" data="media_mix_inbound_outbound_codecs=true"/> <X-PRE-PROCESS cmd="set" data="inbound-late-negotiation=false"/> <X-PRE-PROCESS cmd="set" data="inbound-zrtp-passthru=false"/> <X-PRE-PROCESS cmd="set" data="disable-transcoding=false"/> <X-PRE-PROCESS cmd="set" data="inherit_codec=false"/> <X-PRE-PROCESS cmd="set" data="bypass_media=false"/> <X-PRE-PROCESS cmd="set" data="proxy_media=false"/> What am I missing? Thanks, Jeff

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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

2 1

Verto calls
by Agustí Ubalde 06 Jul '17

06 Jul '17

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 Agustí Ubalde Bellot Chief Developer C/ Comte Urgell 240 3º-A Barcelona 08036 aubalde(a)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 <http://www.presenceco.com/>*

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Conditional call forward
by Ashwin Rath 06 Jul '17

06 Jul '17

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

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mod_xml_curl performance questions
by Michael Avers 01 Jul '17

01 Jul '17

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

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question about HA solution
by Denys Pozniak 19 Jun '17

19 Jun '17

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: <param name="auto-create-schemas" value="false"/> <param name="auto-clear-sql" value="false"/> Is there is a way to solve this? BR, Denys

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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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