April 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Entire Wiki.FreeSwitch.org on Single PDF ?
by Mitul Limbani 15 Apr '09

15 Apr '09

BODY { font-family:Arial, Helvetica, sans-serif;font-size:12px; }Hello there, In my previous encounter with FreeSwitch, I had found that Bret had posted on the Mailing List somewhere about availability of the entire FreeSwitch Wiki Documentation on a single PDF, this is useful coz at the offset apart from Wiki there is no other offline media to learn it. Is the same PDF available looking at the growth of Wiki pages and the updation. I look forward to hear from you guys, Thanks & Regards, Mitul Limbani, Founder & CEO, Enterux Solutions, The Enterprise Linux Company (TM), www.enterux.com +91-9820332422 ------------------------- Msg sent via Enterux Enterprise Email Server : http://www.enterux.com/

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FS/Sofia not sending bye when FreeSWITCH ends the call?
by Peter Olsson 15 Apr '09

15 Apr '09

When I do a call from my Avaya SIP Server to FreeSWITCH. And then let FreeSWITCH do a hangup of the call, FreeSWITCH doesn't seem to send a "BYE" back to the Avaya PBX. I've narrowed it down to this simple example in the dialplan; <action application="answer"/> <action application="sleep" data="5000"/> <action application="hangup"/> In this case no BYE is issued, and the phone still thinks the call is alive. If you want to I could send the SIP headers as well for this scenario.. Regards, Peter Olsson

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Pcapr for packet captures
by Nicolas Brenner 15 Apr '09

15 Apr '09

Has anybody tried this site?: - http://www.pcapr.net/browse/voip Looks pretty good for posting sip dumps and the like. It's a bit more sophisticated than pastebin.

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Mod_java loading error
by Baskar 15 Apr '09

15 Apr '09

Hi, I have loaded the java module in freeswitch. But when i run freeswitch in console i get this error. 2009-04-13 17:38:33 [NOTICE] modjava.c:244 mod_java_load() Java Framework Loading... 2009-04-13 17:38:33 [ERR] modjava.c:133 load_config() Error loading /usr/java/jdk1.6.0/jre/lib/i386/client/libjvm.so 2009-04-13 17:38:33 [CRIT] switch_loadable_module.c:839 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_java.so **Module load routine returned an error** Can any one specify what is the error. I am using Freeswitch 1.0.2 in CentOS 5.2. Thanks in advance. -- Warm Regards, N.Baskar

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Recommended tools for creating/extending a sip test suite?
by Mike Fedyk 15 Apr '09

15 Apr '09

Hi all, I'm looking for suggestions on which open source tools to use for creating (or extending if there is already a project for this) a sip test suite. I have already heard of sipp, but I want to know what others are using and how they go about this before starting from scratch myself. Some things I'd like to do: - Dialplan/ voice menu/provider/did testing: Call number, press 1, expect to receive call on another extension. (kinda like expect) - Load testing Basically I want to be able to automate how a human may interact with my installation to reproduce bugs and make sure they don't come back. That way I can make sure my changes (wherever they may be in my stack, dialplan, freeswitch, openser/kamailio/opensips, etc.). Any pointers and/or tips will be much appreciated.

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Help with play_and_get_digits from mod_dptools
by Diego Viola 14 Apr '09

14 Apr '09

Hi everyone, I have a question... I have this on my dialplan: <extension name="outbound2"> <condition field="destination_number" expression="^(1888)$"> <action application="play_and_get_digits" data="2 5 3 7000 # phrase:rac_demo /invalid.wav foobar \\d+"/> <action application="transfer" data="${foobar} XML default"/> </condition> </extension> What I want to do is play and read some digits and as soon as I get those digits, transfer to that extension... but this never happens, even if I terminate with a #. I do the same thing with Lua and it works with Lua, but I need it to work with play_and_get_digits from mod_dptools, because I plan to use this with event socket outbound, with an application which I'm currently working on. Any ideas? Thanks, Diego

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ANNOUNCEMENT: FreeSWITCH 1.0.4pre4 Now Available
by Michael Collins 14 Apr '09

14 Apr '09

The FreeSWITCH team is pleased to announce the immediate availability of version 1.0.4pre4. Details are available here: http://www.freeswitch.org/node/173 All are encouraged to upgrade as soon as possible. Thanks to everyone for their feedback, ideas, and bug reports. Please keep them coming. -Michael http://www.cluecon.com

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Use of loopback channels and bridge() in scripts...
by Peter Olsson 14 Apr '09

14 Apr '09

I have two problems that I haven't been able to solve. I've done the same tests in both javascript, and in .NET. The two scripts are pretty simple, they just answer an incomming call, creates a new session, wait for an answer on the second call leg, and then bridge the two channels together. In both cases everything works just fine, but the audio is distorted. The destination I'm calling is "loopback/5000" - the sample IVR application included in FreeSWITCH. I first thought it was a codec issue, but even after trying to switch to different codecs the problem was the same. It more sounds like it's a timestamping issue - the voice is not distorted enough to be a bad codec, but it reads way to fast (mayby twice the "normal" speed). When doing a direct transfer() to the other destination this works just fine, but I need to be able to have some extra logic to tell if the destination is available or not. The second problem occurs only in .NET. After doing this sample there is as loopback channel still hanging around. It seems like the call creates a loopback-a and loopback-b, the loopback-b dissapears as it should (when the call has been disconnected), but the other one stays there. When doing the same in javascript this doesn't seem to occur. I'm using the latest SVN trunk, and my OS is Windows XP. I found bug FSCORE-349 in Jira, which seems to point in to the direction that there might be a bug with the loopback channels in some cases, but I could not find anything about the audio which plays too fast. Has anyone else experienced this? Regards, Peter Olsson

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skypiax Round Robin interface
by dujinfang 14 Apr '09

14 Apr '09

Hi, I made a patch, so skypiax is possible to do a RR hunt besides the sequential interface ANY. Usage: originate skypiax/RR/other_skype_name sk list http://jira.freeswitch.org/browse/MODENDP-211

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Google Talk Integration...
by Moiz Chinoy 14 Apr '09

14 Apr '09

Hi, I have tried google talk integration with FS and is working fine. Great Work! Is it possible to have multiple concurrent incoming calls on the same gmail account? -- Regards, Moiz Chinoy.

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