January 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Configuring duration of conference
by Abhishek Dixit 25 Jan '10

25 Jan '10

Hi, I am working on FreeSWITCH Version 1.0.4 (14460). I am trying to set up a conference room which will allow conferencing for limited duration. This means that when parties connect to this conference room they can remain connected only till a specified duration. After this duration the conference should end with announcement. I have read wiki for mod_conference and for dialplan and studied IVR conferencing example using javascript as well. But I am unable to find a way to specify duration for the conference using dialplan and javascript application. I am planning now to use mod_event_socket interface to end conference after a specified duration. Please tell me if there can be a way to configure duration of conference using dialplan configurations or javascript application. Thanks Abhishek Dixit

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FreeSwitch Integration With Sailfin
by juan camilo ospina quintero 25 Jan '10

25 Jan '10

hi im having serious trouble trying to integrate freeswitch with sailfin, if someone could help me, taht would be awesome. I guess is with a bridge statement in the dialplan , but not sure how to do that, and if that is the right solution. thanks _________________________________________________________________ Windows Live Hotmail: Your friends can get your Facebook updates, right from Hotmail®. http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/so…

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E1 hangups
by Michal Zubáč 25 Jan '10

25 Jan '10

Hi. I'm trying to correct this behaviour, but can't figure out, where is the problem. Here's the scenario: - we're trying to execute simple dialplan * answer * play sound * wait for 3 seconds * hangup - for SIP caller it works as expected - problems are when, we try to call into it from E1 line - for E1 we're using sangoma winpipe & openzap - dialplan in freeswitch console is done in a moment ending with hangup - on the E1 line I hear nothing and after 2 seconds it disconnects - similar problem when there's only bridge to another number (E1) - it rings (on the destination phone) for a short moment (0.5-1s), but then hangs up spontaneously Thanks for any clues. mZubac

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Accessing Sangoma card inside a openVZ container
by Henry Huang 24 Jan '10

24 Jan '10

I know there are a couple of OpenVZ expert here on the mailing list. Has anyone of you tried to run freeswitch in a container with access to physical Sangoma card? The reason for this is that I would like to create an "all in one box" for small to medium size company that takes care of the most essential services. Things like an IP PBX would be a huge plus. I have been playing with OpenVZ for a while now, and I know how to let containers access devices in the /dev directory from the physical box. But to be able to use something like sangoma card, I think there are some things in the /proc directory need to be made available to the containers. But I don't know how to do this yet. Maybe some of you know.. Thanks, -- Henry Huang

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Re: [Freeswitch-users] Lua: io.popen/read blocking in other incoming calls
by Dan 24 Jan '10

24 Jan '10

I would, but I need to post a a wav file that gets recorded, I didn't see a way to supply the location of a file to use as the post data. It looks like you have to url encode the data in the script and pass it all in the call. Thanks Dan- ----- Original Message ----- From: "Rupa Schomaker" <rupa(a)rupa.com> To: "freeswitch-users" <freeswitch-users(a)lists.freeswitch.org> Sent: Tuesday, January 19, 2010 4:06:36 PM Subject: Re: [Freeswitch-users] Lua: io.popen/read blocking in other incoming calls On Tue, Jan 19, 2010 at 3:03 PM, Dan < freeswitch-users(a)digitaldan.com > wrote: My lua script is calling wget through lua's io.popen to send and receive data from a web service. While the f:read to wget is running, other incoming calls will block on the same io.popen call until the first call closes the pipe (with f:close()). You might want to look at the api that mod_curl exposes to do what you want. No need to do an expensive system call just to call a webservice. -- -Rupa _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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reloadxml/rescan profile and killgw
by Apostolos Pantsiopoulos 24 Jan '10

24 Jan '10

Wouldn't it be nice a "reloadxml"/"rescan profile" to delete all nonexistent gateways (when they are absent in the xml config) without having to issue a "sofia profile xxxxx killgw yyyyyy" command? I always seem to find forgotten gateway's in the profile because of this... Any thoughts? -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs(a)kinetix.gr -------------------------------------------

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Application="record_session" records 16000Hz in mono now?
by Scott Torr 24 Jan '10

24 Jan '10

Hi, When recording a session in the past the caller audio would be in the right channel and the callee audio in the left of a stereo recording. <action application="record_session" data="/root/recordings/test.wav" /> A 8000Hz recording has remained the same and is in stereo, but a 16000Hz recording is now in mono? Noticed change since FreeSWITCH Version 1.0.trunk (16195). What do I need to do to get stereo recording back? Why the change? Regards, Scott Torr

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strange behavior of openzap/libpri (is that a bug?)
by Tzury Bar Yochay 24 Jan '10

24 Jan '10

Hi All, In my environment I have the following components: * FreeSWITCH Version 1.0.4 * latest libpri * latest openZAP I have noticed that when originating calls to a specific number, the system sometimes reports this as UNALLOCATED_NUMBER and sometimes not. That is, I am calling a number which certainly exists, however, in most of the attempts to cal this number I get the UNALLOCATED_NUMBER error, while every 4th or 5th attempt the line get connected. I paste-bin the log at http://pastebin.freeswitch.org/11928. There is one with oz debug output at http://pastebin.freeswitch.org/11929 please advise, Tzury

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Freeswitch core dump after upgrade to latest version
by Yehavi Bourvine 24 Jan '10

24 Jan '10

Hello, We are running 1.0.5pre10 for a while, and today I tried to move to the latest tarball (from January 22nd). The software crashes with a core dump after a few seconds. The core dump. The two relevant lines (to my opinion) are: #4 0xb7547d2d in nua_bye_server_report (sr=0xb6c910b0, tags=0x0) at nua_session.c:3938 #5 0xb7541bb3 in nua_server_report (sr=0x6) at nua_server.c:643 line 3938 is an assert() statement. Any idea? Thanks! __Yehavi:

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Svar: Re: How do I invite group to join existing call?
by mailinglist 24 Jan '10

24 Jan '10

Hi Michael et al. I found the 'attended xfer' example in the documentation, and it looks like rxactly to what I want to do. http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer The example 2: Do I add the first part to my dialplan/default.xml, and the second part to my dialplan/features.xml How does it affect if I try to do a normal local dial if it just reacts on the fact that three digits have been dialed (which I should change to 2, as I use 2 digit local numbers). Best regards Fribse >>> Michael Collins 18-01-10 21:19 >>> On Sun, Jan 17, 2010 at 10:50 PM, mailinglist <mailinglist(a)fribert.dk> wrote: Hi All I would like to be able to invite a group / global to join an existing call, but how do I accomplish this, can it be done? Malfunction! Need Input! Could you give us a few more details on what you're trying to accomplish? What's the big picture? I'm curious what problem you're trying to solve. I'm sure the gang here will have thoughts to pass along. :) -MC

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