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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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Distinguishing Blind REFER from application transfer
by mayamatakeshi 24 Jan '10

24 Jan '10

Hello, I'm using mod_xml_curl to provide dialplan. In my application, I need to differentiate if a call has entered the dialplan again due to uuid_transfer or due to Blind Transfer. I know I can recognize a Blind Transfer by checking variable_sip_h_Referred-By and variable_sip_refer_to. However, these variables are not cleaned up when the call reenters the dialplan due to application transfer or uuid_transfer. I realize I can differentiate them by adding some prefix like this: uuid_transfer TRANSFER,DestinationNumber XML default Or it is my responsibility to call application set or uuid_setvar to unset the variable(s) in this case? it seems to me the bad thing here would be to have to add lots of preventive <action application="set" data="sip_refer_to=" /> in my dialplan. Before doing so I thought in ask if the REFER-related variables being preserved upon dialplan reentry would not be a bug (well, it could be a feature useful in some scenarios I suspect). br, takeshi

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Svar: Re: Svar: Re: Home setup with home company
by mailinglist 24 Jan '10

24 Jan '10

Ok, I set it up like this: But now it gives me: 2010-01-22 11:52:08.667564 [NOTICE] switch_channel.c:602 New Channel sofia/external/2680xxxx(a)87.54.25.116 [16baec2f-4407-df11-8fb3-000c29b7b4cb] 2010-01-22 11:52:08.800123 [INFO] mod_dialplan_xml.c:252 Processing 2680xxxx->4692xxxx in context public 2010-01-22 11:52:08.830071 [NOTICE] switch_ivr.c:1349 Transfer sofia/external/2680xxxx(a)87.54.25.116 to XML[8203@default] 2010-01-22 11:52:08.830071 [INFO] mod_dialplan_xml.c:252 Processing 2680xxxx->8203 in context default 2010-01-22 11:52:08.830071 [NOTICE] switch_ivr.c:1349 Transfer sofia/external/2680xxxx(a)87.54.25.116 to enum[8203@default] 2010-01-22 11:52:09.163439 [INFO] switch_core_state_machine.c:136 No Route, Aborting huh >>> Michael Collins 21-01-10 9:39 >>> On Wed, Jan 20, 2010 at 2:45 AM, mailinglist <mailinglist(a)fribert.dk> wrote: Hi Michael It's running on pfsense, so it's kinda locked to the version it currently is. Looks very nice though. Looking beyond that, is the action / anti-action list corrent? I would say that you need to add an anti-action under the day of week check and go to vm if it does not match. Right now if the DOW is 0 or 6 then the entire extension will "fail" and the dialplan will just move on. Remember that if any conditions fail then the entire thing extension "fails" unless you are doing interesting things with the break= parameter. See the dialplan page on the wiki for examples of how to use break in your conditions. -MC Best regards Fribse >>> Michael Collins 20-01-10 1:53 >>> On Tue, Jan 19, 2010 at 9:02 AM, mailinglist <mailinglist(a)fribert.dk> wrote: I have a very small one man constultancy company that has the occasional call, unfortunately we are getting more spam calls after hours than real calls during work hours, so I would like to set up a TOD system. First step for me is just playing with the TOD example, I've gotten this far: group/company@${domain_name}"/> My idea with this, was that in the time 9-17 mon-fri, it rings, the rest of the time it's sent directly to the voicemail. I would of course like to have it not take messages outside work hours, but that's just refining :-) But it picks up the call, and then nothing... We have a much cleaner way of doing TOD and DOW handling. You'll need to get to the latest FreeSWITCH version. Look in conf/dialplan/default.xml for this example: Use that condition instead of the two conditions you're now using and see if you have better success. -MC _______________________________________________ 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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Javascript self.session.getVariable
by Mouncif Benniane 24 Jan '10

24 Jan '10

I have the following in javascript: caller_id = self.session.getVariable("caller_id_number") for some reasons it returns: ReferenceError: self is not defined, I am following this page: http://wiki.freeswitch.org/wiki/Session_getVariable any ideas?

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