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Troubles bridging calls with mod_opal
by Gilbert Amar 20 Feb '10

20 Feb '10

Hello, I am having troubles to make FS/opal works. Please do not ask me to use SIP instead of H323 it is out of scope. I am trying here to bridge an incoming h323 call to another h323 party. The result is that most of the time (sometime it actually works) the bridge is done but the callee cannot be heard by the caller. If I dial directly the callee without FS/opal it works. If I dial from a sip client like xlite then it works. Here is our setup: FreeSWITCH Version 1.0.trunk (16659M) (at 2010-02-19 12:57 UTC) FS runs on a CentOs 5.2 The calling party is a soft OpenPhone The called party is a Swissvoice IP10S I try to enable/disable several codecs with no success. Any idea where I should look ? Please find attach the FS log and the Openphone log Thanks Gilbert

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Re: [Freeswitch-users] Establishing a Call from .Net based application
by Diego Toro 19 Feb '10

19 Feb '10

Hi, the answer is yes, you can to use mod_managed wich offer C# managed class to call control http://wiki.freeswitch.org/wiki/Mod_managed. Or using managed ESL (libs/esl/managed) which offer C# managed class to receive and send events and commands to FreeSwitch. Diego Toro http://lacarretade.blogspot.com/ --- On Wed, 1/20/10, Scott Fernandez <scottferri09(a)gmail.com> wrote: > From: Scott Fernandez <scottferri09(a)gmail.com> > Subject: Re: [Freeswitch-users] Establishing a Call from .Net based application > To: freeswitch-users(a)lists.freeswitch.org > Date: Wednesday, January 20, 2010, 2:17 AM > Thanks Dome. Will try it out and get back to > you if I come across any issues. > > Regards, > Scott. > > On Wed, Jan 20, 2010 at 11:02 AM, > Dome Charoenyost <dome(a)tel.co.th> > wrote: > > Please try http://wiki.freeswitch.org/wiki/Webapi > > > you can create class and map to webapi. > > > > Dome C. > > > > 2010/1/19 Scott Fernandez <scottferri09(a)gmail.com>: > > > Hi, > > > > > > Is there any API modules available for me to initiate > a call from .Net based > > > application?. > > > > > > The idea is to include the API modules if any with the > .NET base classes so > > > that the API commands will be made available on it. I > know it is doable when > > > I use socket programming in .NET in which Telnet > session is created. > > > However, this would potentially hamper the performance > of the application > > > because of multiple sessions that will be created for > each call. > > > > > > Other than that, Is there any Freeswitch API modules > (like plug-ins) > > > available in order to include it into the .Net classes > and start building > > > the customized application? > > > > > > Any help from any one is highly appreciated. > > > > > > Thanks, > > > Scott > > > > > > > _______________________________________________ > > > 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 > > > > > > > > > > _______________________________________________ > > 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 > > > > > -----Inline Attachment Follows----- > > _______________________________________________ > 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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FreeSWITCH Conference Call Starting Shortly!
by Michael Collins 19 Feb '10

19 Feb '10

Come on in! http://wiki.freeswitch.org/wiki/FS_weekly_2010_02_19 -Michael

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1.0.4 vs. trunk vs. bugs
by Lon Baker 19 Feb '10

19 Feb '10

Guys, With all due respect I want to suggest a policy change. A while ago it was announced that bug reports against 1.0.4 would not be accepted and the solution was to work off the trunk or the latest nightly build. It seems more reasonable to have a release/production branch that can be depended on for production use. This branch would only accept reports and fixes for critical bugs. The development branch is where feature requests and non-critical bugs reports would be filed for the next production release. The current process leaves a gap between production ready and development code that may become greater over time. Just a thought. Lon

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Troubles bridging calls with mod_opal
by Gilbert Amar 19 Feb '10

19 Feb '10

Hello, I am having troubles to make FS/opal works. Please do not ask me to use SIP instead of H323 it is out of scope. I am trying here to bridge an incoming h323 call to another h323 party. The result is that most of the time (sometime it actually works) the bridge is done but the callee cannot be heard by the caller. If I dial directly the callee without FS/opal it works. If I dial from a sip client like xlite then it works. Here is our setup: FreeSWITCH Version 1.0.trunk (16659M) (at 2010-02-19 12:57 UTC) FS runs on a CentOs 5.2 The calling party is a soft OpenPhone The called party is a Swissvoice IP10S I try to enable/disable several codecs with no success. Any idea where I should look ? Please find attach the FS log and the Openphone log Thanks Gilbert

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Presence PUBLISH Not Updating After Softphone OffLine Then Available
by Jerry Richards 19 Feb '10

19 Feb '10

I found a scenario where presence status is not distributed to subscribers. This is using the latest changes (as of Feb 03, 2010). The scenario follows: 1) Register two Bria softphones (A and B), which each have the other as a contact. 2) Set softphone B's presence status to 'Appear Offline'. 3) Softphone A correctly reflects contact B is offline. 4) Set softphone B's presence status to 'Available'. 5) ******* There is no change to contact B's status at softphone A ******* I posted a log at http://pastebin.freeswitch.org/12054. At line 773, there is an error when FS is processing the PUBLISH from softphone B: 773.2010-02-05 10:29:21.254221 [ERR] sofia_presence.c:674 DUMP PRESENCE SQL: I did notice that after about 30 minutes, softphone B's status gets reflected at softphone A. Thanks and Best Regards, Jerry

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Re: [Freeswitch-users] voivemail quality
by Anthony Minessale 19 Feb '10

19 Feb '10

Our g729 would have been done a lot sooner if we did not spend all our time making sure there is no audio hiccups on a 600ms latent connection. Yes we made it possible and now we have g729 too nonetheless. The only thing howler filled is their wallet. Let's see them as a gold sponsor of cluecon if they are doing us such a big favor........ On Feb 18, 2010 10:58 PM, "David Knell" <dave(a)3c.co.uk> wrote: On Fri, 2010-02-19 at 14:26 +1000, jay binks wrote: > I think suggesting the user test with other c... Edward already said that things worked with G.711 - that ought to be enough testing with other codecs. > your attacking rupa for making suggestions about where the issue is > and what to do in order to... It doesn't, directly. And, had Rupa simply asked if the OP had raised the issue with Howler - which is a completely valid thing to do, particularly with the evidence in front of him - that'd have been one thing. But he didn't. > pot, kettle, black ??? How so, exactly? --Dave > > > J > > On Fri, Feb 19, 2010 at 2:14 PM, David Knell <dave(a)3c.co.uk> wrote: > Rupa -...

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IVR greeting - first two words missing
by Aloysius Lloyd 19 Feb '10

19 Feb '10

Hi All, I setup a simple IVR. Here is the script. <menu name="test_ivr" greet-long="test/test-ivr.wav" greet-short="tset/test-ivr.wav" invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav" exit-sound="voicemail/vm-goodbye.wav" timeout ="10000" inter-digit-timeout="2000" max-failures="3"> <entry action="menu-exec-app" digits="/^(20[0-9])$/" param="transfer $1 XML default"/> <entry action="menu-exec-app" digits="9999" param="transfer 9999 XML default"/> </menu> </menus> </configuration> *Dial Plan* <action application="answer"/> <action application="ivr" data="test_ivr"/> Every time when I reach the IVR . I am getting first one or two words missing( or may be not clear). How can I fix this issue. Thanks, Lloyd

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Re: [Freeswitch-users] SQL ERR: library routine called out of sequence
by Anthony Minessale 19 Feb '10

19 Feb '10

There is no evidence of this on any box I have used. Do you have selinux on maybe. The best I can do now is declare we do not support your current configuration. On Feb 18, 2010 11:48 PM, "Yehavi Bourvine" <yehavi.bourvine(a)gmail.com> wrote: Unfortunately it did not help. I still get these error messages. Just for the record, I have two systems: - Production one which is running Fedora-10 and exibits this problem. - Test system which is running Fedora-12 and does not exibit this phenomenon; however, there is almost zero traffic on this system. I plan in upgrading the production system to Fedora-12, but it is not that trivial... Thanks, __Yehavi: 2010/2/16 Anthony Minessale <anthony.minessale(a)gmail.com>... _______________________________________________ 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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voivemail quality
by Edward Stevenson 19 Feb '10

19 Feb '10

I have V1.0.4 running on a test/production server. It's working quite well, except for voicemail retrieval over a satellite internet connection. Voice calls over satellite sound fine, other than the 600ms delay. I'm using the Howler Tech G729 module for G729 transcoding. I've noticed that in voice calls, the bandwidth over the satellite is a steady 24 kbps in both directions. When accessing voicemail, the bandwidth fluctuates with voice in the call. What I mean by that is, pauses in audio in the voicemail, or in the IVR, cause the bandwidth of the call to drop off momentarily. It's almost like it's using a variable bit rate. The audio sounds like there's packet loss. Pops and garbled speach. This is not noticable over a land internet connection. I'm using the built in voicemail. If I change the phone's codec to G711, the call is perfectly clear. Perhaps the Howler module doesn't like transcoding from the L16 codec that Freeswitch seems to use to play the wav file? Anyone else had any similar issues? -- View this message in context: http://old.nabble.com/voivemail-quality-tp27648642p27648642.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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