June 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Slow Memory Leak with FS 1.4.2 ?
by Zoltan Medveczky 08 Jun '16

08 Jun '16

Hi, I'm running FS 1.4.2 on the following environment: OS: RHEL 6.5CPU: x86 64bit quad coreRAM: 12G The process has been up for almost 2 months and seems to be consuming an inordinate amount of memory which is slowly increasing over time: PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND 3404 contactu 39 19 7572m 2.1g 8384 S 0.7 18.3 578:22.95 freeswitch 600 8685 1 0 Apr10 ? 09:49:55 ./freeswitch -nc -conf /opt/freeswitch-1.4.20/conf -log /var/log/8x8/reach-fs-1.2.0 -db /opt/freeswitch-1.4.20/db There has been no traffic handled by this process. Are there any known issues with this version of FS in regards to memory usage? Thanks

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Distributed presence
by Emrah 08 Jun '16

08 Jun '16

Hello list, I have multiple FS servers and I am trying to share presence info across them using a PGSQL database. I cannot share the same hostname and use PGSQL in the core that way for 2 reasons, 1. The DB is on a remote server and there is some latency. It’s ok to update and show presence but not to conduct all operations. Also, if the DB server fails, everything needs to keep on running normally. 2. I need to be able to distinguish channels and calls on a per server basis. Worst case, I could probably forfeit point 1 if FS can keep on running if the DB crashes, if it’s the only way. But I cannot share the same hostnames across all boxes. I already have odbc-dsn configured for my internal profile, as well as: <param name="presence-probe-on-register" value="true"/> <param name="manage-shared-appearance" value="true"/> <param name="dbname" value="sip_presence"/> For dbname, I put the name of a database I saw FS created and which I assumed would match the purpose… But I couldn’t find much documentation about it. Any hints would be appreciated. That particular table (sip_presence) stays empty and presence isn’t shared. Thanks for any help and all the best, Emrah

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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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