February 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Unauthenticated Gateways for wholesale VOIP
by stramec 05 Feb '15

05 Feb '15

Hi I’m new to freeswitch and I’m just playing with a test box to see if it is viable for a small scale, wholesale SIP transit softswitch. I don’t need PBX features, I just want an incoming SIP trunk and an outgoing SIP trunk, the only catch is they both most be unauthenticated. Based solely on Source IP address and the incoming trunk must listen on port 5060. In my mind this should be the simplest of set-ups but I’m struggling to get the unauthenticated to work both sides. I’ve done some googling but haven’t come up with a clear example. Could someone point me in the right direction? Thanks -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Unauthenticated-Gateways-for-… Sent from the freeswitch-users mailing list archive at Nabble.com.

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can I use Event_Socket_Outbound to only monitor?
by Shai Perelman 05 Feb '15

05 Feb '15

hi, I am using https://wiki.freeswitch.org/wiki/Event_Socket_Outbound my goal is to monitor what happens on the extension but without affecting the normal behaviour. I have this dialplan. TagTypeDataOrder[image: add] <http://148.251.142.45/app/dialplan/dialplan_detail_edit.php?id2=8a56aba2-5c…> condition username 9998 1[image: edit] <http://148.251.142.45/app/dialplan/dialplan_detail_edit.php?id=02e9a3bf-15c…> [image: delete] <http://148.251.142.45/app/dialplan/dialplan_detail_delete.php?id=02e9a3bf-1…> action socket 109.65.149.22:9999 async 3[image: edit] <http://148.251.142.45/app/dialplan/dialplan_detail_edit.php?id=0ca4a9d4-020…> [image: delete] <http://148.251.142.45/app/dialplan/dialplan_detail_delete.php?id=0ca4a9d4-0…> I am calling *9664 (music on hold) and the call is blocking while wating for commands from my side. I want it to answer automatticaly like it would do with out the socket module, and just send me the events regarding that extension thanks Shai ITD Communications

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Gateway for Israel numbers
by Shai Perelman 05 Feb '15

05 Feb '15

Hi, any body knows a good and cheap gateway for working with freeswitch with israel numbers? thanks Shai

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Need help with Video proxy-media mode
by Cinthia Leung 05 Feb '15

05 Feb '15

Hi there, I'm trying to setup FS and proxy media. Here's my setup. Kamailio SBC <-> FS <-> MCU. >show codec sees H.264 as passthru late-negotiation = true Audio is ok but I get no video. Tshark capture shows that video streams enter FS from both Kamailio and MCU. FS was just not forwarding video packets to the other side. Codec is H.264. Tested with FS Version 1.14.15 and 1.5.15b+git~20150130T165344Z~4ed7b4811a~64bit Bypass media works fine. Being able to proxy media is a requirement for us. I hope it's just something obvious that I missed. I'll provide logs if needed. Thanks in advance! Cindy

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Call generator for performance tests
by Stanislav Sinyagin 05 Feb '15

05 Feb '15

I wrote a simple script that generates calls via ESL at a given calls-per-second rate. This could be useful if you need to run a simple performance test: https://github.com/voxserv/freeswitch-perf-dialer Depending on the customer's request, some new features will be added (such as a pool of caller and called numbers). I hope some of you will find it useful. cheers, stanislav

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Convert existing call to a conference and hangup the old call
by Kumar, Sharath 04 Feb '15

04 Feb '15

Hello, I am new to Freeswitch and wanted some input on how best to achieve the below scenario. 1. A and B are in a call. 2. A adds user C to the call. - At this point user A and B are added to a dynamically created conf and C is added as well. 3 A adds user D to the call. D is added to the conf. The fact that step 2 is a conf call and users C and D are provisioned and available to the FS. I am using FS_CURL so the PHP has access to the DB. I know how to add A,B,C and D directly into a conference using mod_conference. I made it work. However, I don't understand how I can access an existing call i.e between A and B and turn that into a conference. and also I guess the original call should be hungup when B accepts the conf invite. Is there a way to do this in the dial-plan ? i.e when A calls C, the PHP knows that this is a conf call, how do I kill the old call though. I guess I can use the uuid of the previous call, I don't know what to issue to hang up the first call. My clients are webrtc based and currently they don't natively support either hold/resume or transfer(using refer). They do however support multiple calls from/to the same user at the same time. i.e 2 lines. I guess both will be active. Thank you. Any help would be greatly appreciated!! -Shaks.

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Monitoring Uptime: landline-->VOIP-Provider-->freeSWITCH-->landline case?
by Arfst 04 Feb '15

04 Feb '15

Hello, I need advice - what kind of monitoring or testing applications (OS,free) do you know for testing the uptime of a VOIP landline number provider and phones (voip + landline) behind a freeSWITCH ? Since this post is a start for a educational journey I am also glad for literature documentation regarding monitoring of telephone system, preferable with open source applications. All the best Arfst

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No Event on playback stop during session_displace?
by Joshua Gigg 04 Feb '15

04 Feb '15

I'm attempting to use session_displace to play an audio file to one side of a bridged call. However, I don't seem to get any events when that file stops playing? I can see a MEDIA_BUG_STOP, and that contains the current_application of session_displace. Should there be a PLAYBACK_STOP? (Running 1.4.15) -- Joshua Gigg

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Re: [Freeswitch-users] Help need for Lua Script development
by Mayank Nakrani 04 Feb '15

04 Feb '15

Yes. I know it wont run outside freeswitch but i wanted it to assist me in resolving method names and variables the way it does for JAVA. Just enabling faster coding. After dev work i neways would copy the script to freeswitch folder n run from within it. (the way mod_java gives u freeswitch.jar lib file which when imported in ur java project ) Regards Mayank ---------- Forwarded message ---------- From: Michael Jerris <mike(a)jerris.com> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Cc: Date: Tue, 3 Feb 2015 11:50:13 -0600 Subject: Re: [Freeswitch-users] Help need for Lua Script development It sounds like you are trying to run scripts that can only run via freeswitch with embedded lua in a seperate lua process. This is not possible. Those need to be run from within the freeswitch process. > On Jan 31, 2015, at 4:11 AM, Mayank Nakrani <mayank19882001(a)gmail.com> wrote: > > Hello Guys, > > Im starting with my Lua scripts dev. Have already had luck writing an Callback script. > Now i would to go deeper integrating it with Mysql and other libraries and generating some dynamic dialplan. > > Till now i used to write my scripts in Text editor and would to know is it Possible in IDE like Eclipse. > I installed Lua development plugin in Eclipse but was wondering Whats the correct way to import freeswitch libraries in the project so that i can see methods/functions in in freeswitch API for easier and faster coding. > > I tried loading "libfreeswitch" and "mod_lua" lib using "require" command but its failing with > > [com.naef.jnlua.LuaRuntimeException: error loading module 'libfreeswitch' from file './libfreeswitch.so': > ./libfreeswitch.so: undefined symbol: apr_os_default_encoding ] > > Im gud in Java and can use mod_java but would like to stick with lua as i heard its more closer to freeswitch. > > Any help :) >

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Many destroyed channels still present in channels and calls table when using odbc!
by Babak Yakhchali 04 Feb '15

04 Feb '15

Hi I'm using odbc in the core with mysql on a remote machine in local lan, sometimes number of channels grows rapidly but many of them are not existing according to uuid_exists. Channels will persist in database till I delete them manually. How can I debug the problem? Thanks

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