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GSM gateway to Freeswitch audio problem
by Deepika Yadav 08 Jun '16

08 Jun '16

Hi, I want to make calls forwarded from GSM gateway to Freeswitch extension When I dial a 10- digit mobile number of a sim configured in the GSM gateway in a way that it forwards the incoming call to freeswitch extension. I do not hear anything except for some background noise Freeswitch codec configuration : <X-PRE-PROCESS cmd="set" data="global_codec_prefs=GSM,PCMU,PCMA"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=GSM,PCMU,PCMA"/> GSM- Gateway Codec settings : priority 1 - G.711 u-law priority 2 - G.711 a-law Dialplan in public context : <include> <extension name="public_did"> <condition field="destination_number" expression="^(9811806537)$"> <action application="transfer" data="5000 XML default"/> </condition> </extension> </include> Freeswitch logs: 2016-06-07 05:10:51.000656 [NOTICE] switch_channel.c:1055 New Channel sofia/external/00919716517818@192.168.20.36:5060 [33e4e958-2c6e-11e6-9af0-278859dd14e6] 2016-06-07 05:10:51.000656 [INFO] mod_dialplan_xml.c:635 Processing 00919716517818 <00919716517818>->9811806537 in context public 2016-06-07 05:10:51.000656 [NOTICE] switch_ivr.c:1861 Transfer sofia/external/00919716517818@192.168.20.36:5060 to XML[5000@default] 2016-06-07 05:10:51.000656 [INFO] mod_dialplan_xml.c:635 Processing 00919716517818 <00919716517818>->5000 in context default 2016-06-07 05:10:51.000656 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/00919716517818@192.168.20.36:5060! 2016-06-07 05:10:51.000656 [NOTICE] mod_dptools.c:1268 Channel [sofia/external/00919716517818@192.168.20.36:5060] has been answered 2016-06-07 05:10:51.640675 [INFO] switch_rtp.c:5846 Auto Changing port from 192.168.20.36:31012 to 103.25.231.101:31012 2016-06-07 05:11:06.560680 [NOTICE] sofia.c:952 Hangup sofia/external/ 00919716517818@192.168.20.36:5060 [CS_EXECUTE] [NORMAL_CLEARING] 2016-06-07 05:11:06.560680 [NOTICE] switch_core_session.c:1641 Session 262 (sofia/external/00919716517818@192.168.20.36:5060) Ended 2016-06-07 05:11:06.560680 [NOTICE] switch_core_session.c:1645 Close Channel sofia/external/00919716517818@192.168.20.36:5060 [CS_DESTROY] Regards, Deepika

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mod_H.323, working stable ?
by Marcel Haldemann 08 Jun '16

08 Jun '16

Howdy fellow Freeswich-Users, We are using FreeSWITCH since while now with SIP. However, recently I got ask what I would take to support H.323. I then checked the Docs and the commits in the git repos for mod_h323 and mod_opal. The changes in the repos seems to be similar (not mutch (only some VS2015 updates) going on in recent years). Regarding the Docs it seems better to use mod_h323 rather than mod_opal, as it is in a "beta" stage (info from 2008, maybe outdated ?). The underlying project H323plus seems to have stopped development in 2014. Is there someone, currently using H.323 with FreeSwitch and made it working stable (preferably using FS 1.6.x) ? If so, any hints on what mod to use (mod_h323 or mod_opal) ? any traps ? regarding docs in the H323 mod the trunk version of PTLIB does not work. What Version did u use ? I think it would be better to use mod_h323 or is it suggested to use mod_opal ? What are your experiences in H.323 with FreeSwitch, does it work stable ? Trying to get an impression whether it's a good idea to use the H.323 mod (or opal), I'm happy about any feedback. Also if someone countered unresolvable problems with the H.323 mod. Kind regards

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Hanhup hook problem when I make a-leg only call
by Kamil Nigmatullin 08 Jun '16

08 Jun '16

Hello, I want to make a program, that makes a call to the user and calls my script when it ends. I run the following script: dialA = "{origination_caller_id_number="..telephone_from..",ignore_early_media=true}sofia/gateway/iskratel/"..telephone_to while(legA:ready() and dispoA ~= "ANSWER") do dispoA = legA:getVariable("endpoint_disposition") freeswitch.consoleLog("INFO","Leg A disposition is '" .. dispoA .. "'\n") legA:execute("set", "api_hangup_hook=luarun autoinform_cleanup.lua"); legA:execute("set", "session_in_hangup_hook=true"); legA:execute("playback", "/usr/local/freeswitch/sounds/alert.wav"); legA:execute("hangup"); end and simple log of autoinform_cleanup.lua dat = env:serialize() freeswitch.consoleLog("INFO","Here's everything:\n" .. dat .. "\n") It makes a call, but when autoinform_cleanup.lua is called I see this in console /usr/local/freeswitch/scripts/autoinform_cleanup.lua:1: attempt to index global 'env' (a nil value) -- Kamil Nigmatullin Tel: 77272323748 mob: 7 (707) 2517003 Skype: kamil.nigmatullin

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ICE problem
by Carlos Ruiz Díaz 08 Jun '16

08 Jun '16

Hi guys, I set up latest FS from master, and there seems to be a problem with the ICE handshake and establishing audio flow. The problem that I detected is that ICE setup takes 2 or more seconds to complete, and around 10% of the time it fails. Same SIPjs configuration works fine with my current setup, which is based on *git 7cf1238 2015-08-07.* Is there anything that I need to configure or take into account for latest master? Thanks, -- Carlos

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FreeSWITCH training and ClueCon 2016!
by Kathleen King 07 Jun '16

07 Jun '16

[image: https://cluecon.com] <https://cluecon.com> [image: https://cluecon.com] <https://cluecon.com>

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Silence Detection in Recording
by Jeevan Singh Dhami 06 Jun '16

06 Jun '16

Hi, I have recorded the voice using freeswitch and their found silence in recording if recording goes below *given silence threshold. after that freeswitch stop the recording .but there was no indication given by freeswitch for silence detection .Could you please tell me there is any way to find the indication parameters in freeswitch. Either any solution for knowing silence found in recording in freeswitch.* *please share the solution in same mail .i will appreciate your feedback. * -- Thanks & Regards Jeevan Singh Dhami Software Engineer – Development

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ICE issue
by Bastien LE REST 06 Jun '16

06 Jun '16

Hi, When I enable ICE using PJSIP, FreeSwitch answers "488 Not Acceptable Here". When I disable ICE, I don't have any problems, the calls work just fine. I tried to use another SIP client with ICE and the call works fine. So I think the problem has to do with PJSIP. I'm working on the migration from an Asterisk architecture to a FreeSwitch architecture. I have to deal with PJSIP and ICE support because my client is a mobile application who is already published on the stores. PJSIP trace: https://pastebin.freeswitch.org/view/ddeaae4b FreeSwitch trace: https://pastebin.freeswitch.org/view/d380e01d Could you please have a look at the trace and tell me what am I doing wrong? Thank you in advance. Kind regards, Bastien.

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Git error
by Nduwayezu, Joselyne 05 Jun '16

05 Jun '16

Hello, I'm trying to have clone this one: *1) git clone git://git.freeswitch.org/freeswitch-contrib.git <http://git.freeswitch.org/freeswitch-contrib.git>* but i have the following error "fatal: unable to connecte to git.freeswitch.org git.freeswitch.org[0: 209.105.235.6]: erno=Connection refused git.freeswitch.og[1: 2607:f348:1021::6]: erno=Network is unreachable But when i ping git.freeswitch.org or ping 209.105.235.6, the ping success 2) *wget http://music.incompetech.com/royalty-free/Parisian.mp3 <http://music.incompetech.com/royalty-free/Parisian.mp3> * the error is: HTTP request sent, awaiting response .... 404 Not Found Any ideas? Thanks NDUWAYEZU Joselyne -- Haute École de Bruxelles

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Inbound calls from GSM to extension that called person on gsm
by Bipin Patel 05 Jun '16

05 Jun '16

I'm having one client who has a special requirement, i have setup FS for them using static XML files and a GSM dongle, they are able to call other extensions and gateways as well as dial out of the GSM line but they wanted incoming calls from GSM to ring the extension that called the person using the GSM line. Meaning extension 1008 calls a number which FS routes out of the GSM line so the called party sees the GSM number always and suppose if the called party is busy and doesn't answer or rejects call then when later he dials back in I want FS to know which extension had called that number so incoming calls from the GSM would ring that extension only. I know I can do this dynamically using XML curl and CDR etc but is there a simple way to achieve this using static XML files only as this FS setup is very small and very basic

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mod_gsmopen wont compile on rpi3
by Bipin Patel 05 Jun '16

05 Jun '16

sorry if this seems like a stupid question but im a newbie to linux, i compiled FS 1.6 on rpi3, later i wanted to add the mod_gsmopen module so i enabled that in modules and recompiled but it never got compiled so then i tried compiling it standalone using the below commands but make clean and make install keep failing with message no rule to make target clean/install apt-get install gsm-utils apt-get install libgsmme-dev apt-get install usb-modeswitch-data usb-modeswitch cd /usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen/libctb-0.16/build make DEBUG=0 GPIB=0 make DEBUG=0 GPIB=0 install ldconfig cd /usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen/ make clean make install -- Regards, Bipin ------------------------------------------------------------------------

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