April 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

100& LSB compliant FS init script
by Madovsky 13 Apr '10

13 Apr '10

For a lot of Linux distrib it should work.... interesting for cluster mans enjoy Franck

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ClueCon MMX - Update
by Michael Collins 13 Apr '10

13 Apr '10

Greetings, We would like to let everyone know that ClueCon MMX preparations are moving forward. New sponsors have signed up and we are starting to fill up our speaker slots. Stay tuned for more information. In the meantime, if you have not already signed up for ClueCon then please call 877.742.CLUE (2583) and get registered right away! Looking forward to seeing you in Chicago, The ClueCon Team 877.742.CLUE http://www.cluecon.com

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Embedded Freeswitch Installation for Windows?
by Frank Church 13 Apr '10

13 Apr '10

Can Freeswitch be installed simply by zipping up the folder and unzipping it to the destination? Does it require some DLLs to be installed in the Windows system folder and some registry entries as well? -- Frank Church ======================= http://devblog.brahmancreations.com

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media_bug_answer_req ignored if aleg is already answered.
by Fraser Redmond 13 Apr '10

13 Apr '10

If you have a dialplan which answers the incoming call, runs through some IVR's/javascript/etc, then bridges a call to another ext, the variable media_bug_answer_req is ignored. media_bug_answer_req stops the ringtone from being included in a session recording, specifically it tells the recording to not start until after it is answered - but it is basing that on either leg being answered (or at least the current leg), rather than on both legs (or the b-leg) being answered. Is that something I should log in Jira, or is there something I'm missing? (I tried running the record_session on the b-leg via execute_on_answer, but then I couldn't find a way to get it to stop recording, so the audio file was never created.) Cheers, Fraser

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mail queue
by Brian West 13 Apr '10

13 Apr '10

I dumped the mail queue this morning in hopes it would clear up anyone getting duplicates. The log file reached 2 gigs and the MTA decided to just exit since the OS wouldn't let it open the logfile for append over the weekend. Which caused this problem in the first place. Sorry for any trouble. Thanks, Brian

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False Answer Supervision (FAS) - Question
by Alfredo Quiroga-Villamil 13 Apr '10

13 Apr '10

All: A while back I tried to solve a false answer supervision issue I was intermittently receiving from underlying carriers. Back then I tried to find a solution using asterisk but had other pending things and put this off until now. Does anyone have any recommendations on how to possibly handle or get around FAS using FS. If I am not mistaken what would be needed is to have something that upon receiving the first 200 message, it simply ignores it, never propagating it and waits for the next 200. I can control this now a little bit better since it's only happening when the calls are sent to a couple of GrandStreams (FXO). My idea is to perhaps write a little application and add it to the DialPlan. The little script/app. would have the logic to ignore the first 200 and hopefully get around this issue. Is that something that is doable? Any other ideas on how to do this assuming is even possible to do it? Thanks in advance, Alfredo

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sip uri incomming calls
by thomas.ji＠gmx.at 13 Apr '10

13 Apr '10

hello list! can somebody tell me, what i have to do to enable incoming sip uri calls to a freeswitch, which is using the default config? in other words, i would like to make a extension available via sip uri from the internet. what do i have to change in which files? thank you very much for your help! thomas -- GMX.at - Österreichs FreeMail-Dienst mit über 2 Mio Mitgliedern E-Mail, SMS & mehr! Kostenlos: http://portal.gmx.net/de/go/atfreemail

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SIP Trunk Mitel 3300
by Kenneth Noisewater 13 Apr '10

13 Apr '10

Hi, I've already read the interop list, but I'm wondering if anyone on here has anymore experience/info on trunking freeswitch to a Mitel 3300? Specifically, I want to use freeswitch for acd and sip registrations, and just use our mitel for switching to the PSTN. Does anyone have some good info to share? Thanks, Kenny

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Bad HTTP POST data in xml_cdr
by Yuriy Ivzhenko 13 Apr '10

13 Apr '10

I use xml_cdr for logging And i have problems with non-english letters in variable 'caller_id_name' HTTP POST data terminates ".......20%3Ccaller_id_name%3E%" when variable 'caller_id_name' contains non-english letters so some variables do not get to HTTP CGI program Can i restrict users with non-english letters in 'caller_id_name' or rewrite this variable for a-leg?

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call bargein (urgent call)
by Nikolay Kondratyev 13 Apr '10

13 Apr '10

Hi all, i need a sort of "call barge in" functianality, ar may be "urgent call" will be better name... I know about existing "eavesdrop" application, but it does not suite me because of two reasons: 1. callee must be on active call, or eavesdrop will fail. 2. say user 1001 dialed 881002 to eavesdrop user 1002 talking to 1003. If 1003 hangs up, the call will terminate. Existing "intercept" application also does not suite. So i need the following: A dispatcher must be able to call any user urgently, that is if a user is idle, it just should be a call, and when a user A is talking to user B, and dispatcher makes urgent call to A, all three must occur in a conference. I'm sure it's possible, but i'm rather new to FS and looks like i'm lost while searching how to achive it.... Can anybody please advise how to do it? Thanks in advance, Nikolay.

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