January 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Are you using mod_httapi
by Michael Collins 14 Jan '12

14 Jan '12

Hello all! I know that many of you already know about mod_httapi. I am collecting examples and experiences from people who have had a chance to try it out. If you are using mod_httapi, even just for testing/tinkering, please contact me off list. Thanks! -Michael

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Re: [Freeswitch-users] valet_park timeout
by Bob Smith 13 Jan '12

13 Jan '12

Michael, Permit me to perhaps expand a little on why I said what I did : 1. As a newbie to Freeswitch, it is right and proper that I take the time to review the available documentation (including purchasing the eBook in my case). The wiki documentation is severely lacking... some of it is out of date, a lot of it is very sparse, and there many un-necessary and unprofessional comments embedded in it (e.g. "???What is it acknowledging???", "What does this do ???" etc.). I have been spending many frustrating hours trying to educate myself in the mysterious FreeSwitch through the wiki and eBooks. Why ? Because I consider it the correct approach as opposed to littering the "freeswitch-users" list with a ton of newbie questions. 2. Building on the point above, surely you should be encouraging ease of adoption by a new community of users. Uses who will no doubt eventually end up contributing to the project as their experience with Freeswitch grows, and their time permits. It is not exactly fair or reasonable to expect newbies to dig around the source code in order to uncover useful functions which are undocumented, but have long been stable. Particularly, if like me, their background is not in C programming. 3. I was not asking the developers to write a book each time they code a new feature. Even just a two column table (function/variable/etc + one or two phrase description of function/variable/etc.) would be better than the old, out of date Wiki of today. The developers are the ones best placed to know when a feature is (a) introduced and (b) becomes stable enough in the code tree that they are happy with it. As you well know, half of a coders job is documenting .... uncommented code is sloppy, undocumented software is unusable. As developers, you should be proud of your work, proud of all the new features you're introducing, and the existing features you are enhancing. So surely you should be blowing your trumpet by even modestly documenting your new work by a new phrase or two in the wiki ? Take my example of valet_park timeouts …. I spent much time trying to search through the lists, more time digging fruitlessly through the wiki and the eBook in search of the faintest trace of documentation of the new variables introduced. I tried poking through the source code, but being unfamiliar with (a) C and (b) the FreeSwitch tree, I couldn't really comprehend what was going on. So I ask on Freeswitch-Users and get a stupid answer that says "LOOK CLOSELY" (little did the poster know just how much time I had expended "looking closely" !!!) … I spent days looking closely, days which could have been avoided if the developers took 2 minutes to add 4 lines to the Wiki just listing and ever so briefly describing the new variables ! Everyone is entitled to their own opinion, of course, but I suspect this is one area where ours may vary ! B

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audo sync issues with record_session to mp3
by Tom Parrott 13 Jan '12

13 Jan '12

Hi, We are running a cluster of 3 SIP servers all running Freeswitch. Each call is recorded using the record_session application to record to mp3 files using mod_shout. Longer calls, after about 10 minutes start to introduce sync issues between the A-leg and the B-leg. We are running record_session on the A-leg, and it seems to get ahead of the B-leg. For example the caller on the A-leg will be heard to answer a question whilst the person on the B-leg is asking it. I am aware that recording to .mp3 can use additional resources, but these are quad core servers with 4GB of ram and they are only doing around 10 concurrent calls each. Additionally I have seen others experiencing these issues with Wav files too: http://lists.freeswitch.org/pipermail/freeswitch-users/2010-July/060107.html I am using the git head version from Tue 11 Oct 2011. I have also tried turning off enable_file_write_buffering on the A-leg however this makes no difference either. Does anyone have any suggestions? Thanks

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Empty WAV file recorded
by Hynek Cihlar 13 Jan '12

13 Jan '12

Hi all, an attempt to uuid_record ends with wav file with no data only header. ct The scenario is following: 0. FS <---> ESL app 1. Bridge is established on one incoming and one originated channel. One channel hangs up. 2. uuid_record start is issued on the active channel. 3. uuid_record stop is issued on the active channel. 4. The result is a wav file with always the same length of 44 bytes. When recording to mp3, the file is always correctly created. Any idea how to diagnose the problem? Hynek

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speech to text engine
by Michel Daggelinckx 13 Jan '12

13 Jan '12

found this one on thepiaf forum. mightbe usefull for a freeswitch module http://nerd.bz/xqYO2D

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AS-SIP
by Jacob.E.Miles＠L-3Com.com 13 Jan '12

13 Jan '12

Is there support for AS-SIP (Assured Services Session Initiation Protocol)? I have only spent maybe 2 minutes googleing this so far with no results yet, so I thought my quickest way of finding out would be to ask you guys. I know AS-SIP uses the SIP Communications Resource Priority (RFC 4412) and the SIP Reason Header for Preemption Events (RFC 441) to allow MLPP (Multi Level Precedence and Preemption). Jacob Miles Software Engineer L-3 Communications - Integrated Systems Greenville Jacob.E.Miles(a)L-3Com.com 903.457.4422

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Re: [Freeswitch-users] valet_park timeout
by Bob Smith 13 Jan '12

13 Jan '12

Thank you Brian. Although I appreciate the constant development of FreeSwitch, I don't think it's too much to ask that the developers should keep the Wiki pages up to date in respect of features that have been added and/or removed. There is no mention of those parameters on http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_valet_park. ;-(

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help about diaplan
by Miha Zoubek 13 Jan '12

13 Jan '12

Hi, to explain what I would like to achieve. Users in different geographic location needs to call a certain special numbers different. If someone in location a is calling number 112, FS need to add prefix 454 before 112 (so 454112). Someone in location b when call to 112, FS needs to add 324 (so 324112). What is the best way to achieve that? Is it possible to add users in location A in group a and users on location B in group B and that mach in dial if it is group A than do this, if it is group B that do this? I noticed <condition field="group"> do not work? To mention I need to put this condition before my dialplan, so if the condition for dialed number it not matched, that proceed with my default dialplan. Thank you! regards, Miha -- Best regards / Lep Pozdrav Miha Zoubek Softnet d.o.o.

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ESL Managed IVR and CallControl
by Michael Lutz 13 Jan '12

13 Jan '12

Hi, I have been playing with mod_managed for a while now but nobody realy seems to be able to help me out. So now I am trying to go to ESL. (which most of the people suggest anyway). What I need is control from .net over the complete call! (so even when bridged and returns). I need to build a main library where all my generic functions will be put in so I can use them in my different scripts. What I''ve figured out is problably the best way is to write a server which connects to my FS an listens to all events. When a call comes in I need to fire a stored procedure to my sqlserver and based on the reply I will decide if the call needsto be answerred, early media needs to be played, or the call needs to be rejected. In any case, at the end of every call (answered or not) I need to finalize my call with another stored procedure. When teh call is answered based on stored procedure result I need to run a script (also managed code) run a script from another dll so I have all my scripts separate but they share the corelib I create. I have been searching the web for a long time now to see if I can find something simular, but I only see generic examples wich don't actually show how to control multiple calls in a scenario like this. Who can help me out point me to the right direction or has experience in setting something up like this? Thanks in advance, Michael Lutz.

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Voicemail and fax names
by Gilad Abada 13 Jan '12

13 Jan '12

Hi Does anyone know what names the vm and fax files? (DB or FS)? Thanks Sent from my mobile device.

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