December 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Park with Pre Answer
by Ron McLeod 19 Dec '09

19 Dec '09

Is there any way to park a channel without causing pre-answer (resulting is a SIP 183 Session Progress)? Thanks, Ron

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Re: [Freeswitch-users] RTP problems in recent revisions?
by Anthony Minessale 19 Dec '09

19 Dec '09

Also retest with no zrtp send a full console debug log with sip trace On Dec 19, 2009 8:33 AM, "Michael Jerris" <mike(a)jerris.com> wrote: The best help to track this down is to try to identify the specific svn revision that caused the issue and to supply a full freeswitch debug with sip trace. Mike On Dec 19, 2009, at 3:31 AM, Jason White <jason(a)jasonjgw.net> wrote: > Revision 15904 is fine, but...

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Allow/Deny REGISTER Request Based on User-Agent Header
by Jerry Richards 19 Dec '09

19 Dec '09

Is it possible to allow/deny REGISTER requests based on the User-Agent header? I need to know/manage what devices are registering. Best Regards, Jerry

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RTP problems in recent revisions?
by Jason White 19 Dec '09

19 Dec '09

Revision 15904 is fine, but after upgrading to revision 16003 I get the following. 1. No problems with a FreeSWITCH to FreeSWITCH call (Celt codec). 2. A PCMU call to a SIP provider is fine for the first 20 to 30 seconds, then the audio breaks up completely. I have ZRTP compiled in, if that makes any difference. Obviously there's a regression somewhere. Let me know if I can provide further help.

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packaging preference question
by Frank Carmickle 18 Dec '09

18 Dec '09

Hello all The packaging folk are interested in knowing if anyone has a problem with having the install set up the user and group to freeswitch:freeswitch. This would be the default on debs rpms and ports packaging. The freeswitch user would be added to daemon and audio groups. The FusionPBX packaging can then add www-data/apache to the freeswitch group. Any objections? --FC

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Ringing after call has been rejected
by bcxml 18 Dec '09

18 Dec '09

I have an incomming call being answered by FreeSwitch and passed to IVR application which rejects the call. The call is never answered by FreeSwitch, but instead of hearing a busy signal, the caller hears ringing. Can anyone advise how I can get the user to hear a busy signal after call rejection instead of ringing. Here is the debug trace http://pastebin.freeswitch.org/11558 Thanks Brian -- View this message in context: http://old.nabble.com/Ringing-after-call-has-been-rejected-tp26842055p26842… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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SIP Re-invite
by DJB 18 Dec '09

18 Dec '09

We have a customer that we are sending calls to off the FS and here is the issue: Call is initially setup fine and they send a first re-invite with media 0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first re-invite fine They then send a second re-invite with their media IP to cut through media and the FS sends a 200 OK to this fine. At this point the call is fine 30 minutes later they send a third re-invite because according to them it is strictly for the purpose of “keep alive” per RFC 4028. This third re-invite has the exact same media IP and UDP pot information as the second re-invite does. The problem is FS does not respond to this third re-invite AT ALL. It doesn’t send a 100 trying a 200 OK nothing so this causes the call to be dropped as the other end does not recieve a response from FS. One more thing, we did not see the third re-invite in sofia siptrace, but we do see it in ethereal, which is kind of odds. We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode. Thank you very much.

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LUA and return variables
by Robin Vleij 18 Dec '09

18 Dec '09

Hi guys (and girls)! I'm working on a little bit of ENUM trickery and I tried doing some (illegal) nested conditions. :-) What I want to do is to first check enum with the ENUM application, then depending on the answer do some stuff. Say that the domain part of the ENUM answer is robin.nl, then I want to do action X instead of just briding the enum answer directly as I see in most examples. But I remembered that it wasn't allowed to do nested conditions. So what I did was stacked conditions. After that I read the dialplan wiki pages again and figured that my regexp never matches because variables I "set" during some phase of the extension I can't use in the same "go" as another condition. So, now my plan is to use LUA to do the regexp. I'll pass the enum answer to a lua script which will split the answer in a user and domain part and return those two to the main app. Then based on those two vars I'll do routing or other actions (like, prefix and then route). Is this how I'm supposed to do it? I can't find many examples on manipulating ENUM answers, other than bridging them directly. I can't change the way I do stuff to ENUM answers, because in most cases I'll just route them out the standard way. Anyone with experience on fiddling with ENUM answers? -- Robin Vleij

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FreeSWITCH Weekly Conference Call Starting Shortly!
by Michael Collins 18 Dec '09

18 Dec '09

Hello everyone! Today's agenda is listed here: http://wiki.freeswitch.org/wiki/FS_weekly_2009_12_14 Also, we are going to be giving away goodies on some of the upcoming conferences, so call in and see what we've got in store. :) For the first 15 minutes we'll let everyone mingle and then we'll get into the agenda. Talk to you all soon! -Michael

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Re: [Freeswitch-users] Voicemail->Email
by François Legal 18 Dec '09

18 Dec '09

I get the same result with sendmail. This used to work in 1.0.3 , and after upgrading to 1.0.4 (then some snapshots) then 1.0.5pre8 and the problem is still there. François On Thu, 17 Dec 2009 17:33:58 +0100, Oliver Schönbeck wrote: Currently it is Version 1.0.trunk (15982) VON: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] IM AUFTRAG VON Brian West GESENDET: Donnerstag, 17. Dezember 2009 17:17 AN: freeswitch-users(a)lists.freeswitch.org BETREFF: Re: [Freeswitch-users] Voicemail->Email What SVN rev. exactly? /b On Dec 17, 2009, at 10:13 AM, Oliver Schönbeck wrote: Hello, we are running freeswitch 1.0.trunk and are currently trying to get the mod_voicemail to send the received messages to the user by using exim4 on a debian machine. So far we followed the instructions in the wiki article ( http://wiki.freeswitch.org/wiki/Mod_voicemail ). I added some lines to the bash script to enable some kind of logging: #! /bin/bash typeset LOG="/tmp/${0##*/}.out" mv $LOG ${LOG}.old >/dev/null 2>"> [[ -t 1 ]] &">exec > $LOG 2>">exim4 -t -v >> $LOG If I run the script from the command line everything is working as expected. If the script gets called by freeswitch I get the following result in my logfile: /usr/local/freeswitch/scripts/exec_exim.sh: line 6: 4920 Segmentation fault (core dumped) exim4 -t -v >> $LOG Has anybody seen similar effects before? Any advice whats going wrong is heavily appreciated. Thanks Oliver _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org [1] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [2] UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org [3] Links: ------ [1] mailto:FreeSWITCH-users@lists.freeswitch.org [2] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [3] http://www.freeswitch.org

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