- FreeSWITCH-users - FreeSWITCH Mailing Lists

Quick Freeswitch success story
by Michael Collins 07 May '08

07 May '08

FYI, Just thought I'd share. I spoke with someone in my area who had heard of FS. This is the first time I've run into someone in person who had heard of FS but who had not heard of it from me. FYI, the guy is a former ATT sales rep and is now working for Telepacific. He said that he's got a client running both FS and Asterisk. I'm hoping to contact that person and start a central California OSS telephony group. Anyway, just wanted you all to know that FreeSWITCH has officially invaded central Cali! :-) -Michael P.S. - if you are in or near Fresno and want to start an OSS telephony group, or are part of FOSUG, please contact me off list.

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sofia profiles reload/restart
by Rodrigo P. Telles 07 May '08

07 May '08

Hi Folks, I have found in the wiki that in order to reload a sofia profile you just need to fire 'sofia profile <profile_name> restart' through the socket or console but in fact sofia doesn't apply the new configuration for a profile using that command. Looks like I'm using the command in the right way: freeswitch@localhost> sofia profile outbound restart API CALL [sofia(profile outbound restart)] output: restarting: outbound freeswitch@localhost> 2008-05-06 22:03:24 [NOTICE] sofia.c:646 sofia_profile_thread_run() waiting for worker thread 2008-05-06 22:03:24 [NOTICE] sofia.c:1486 config_sofia() Started Profile outbound [sofia_reg_outbound] Can some one help me on that? I appreciate any answer. Thanks in advance, Rodrigo P. Telles

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Help with outgoing calls
by Bill O'Neal 07 May '08

07 May '08

Provider.xml located in directory:default: Note: I tried to register the provider through the sip_provider/outbound location but would not register. I have set up a simple dialplan to call my cell phone. The call comes in and connects fine but the outgoing doesn't work. Below is the outgoing dial plan. Can anyone help - Thanks. <file:///E:\vpoint.xml##> - <include> <file:///E:\vpoint.xml##> - <user id="xxxxxxxxxx"> <!-sip provider user name same as phone number --> <file:///E:\vpoint.xml##> - <gateways> <file:///E:\vpoint.xml##> - <gateway name="xx.xxx.xx.xx"> <!-sip provider ip address --> <param name="username" value="xxxxxxxxx" /> <!-sip provider user name, same as phone number --> <param name="password" value="xxxxxx" /> <!- Sip provider password <param name="expire-seconds" value="600" /> <param name="register" value="true" /> <param name="retry_seconds" value="5" /> <param name="context" value="public" /> </gateway> </gateways> <file:///E:\vpoint.xml##> - <params> <param name="password" value="xxxxxxxx" /> </params> </user> </include> LLocated in my default.xml in the conf/dialplan <file:///E:\default.xml##> - <extension name="outbound"> <file:///E:\default.xml##> - <condition field="destination_number" expression="mycellphonenumber"> <action application="set" data="effective_caller_id_number=sipproviderusername@sipprovideripaddres s" /> <action application="bridge" data="sofia/gateway/sipprovideripaddress/mycellphonenumber" /> </condition> </extension>

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Meeting today
by Brian West 07 May '08

07 May '08

Just a reminder... The meeting is today. 1PM central time (-6GMT) http://wiki.freeswitch.org/wiki/Wiki_meet_2008_05_07 Please add anything to the agenda that you feel we need to talk about. Thanks, Brian West sip:brian@freeswitch.org

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Clock Skew
by Peder ＠ NetworkOblivion 07 May '08

07 May '08

Any idea what this means: 2008-05-05 09:01:44 [CRIT] switch_time.c:351 softtimer_runtime() Forward Clock Skew Detected! It is a test box that I installed with RC3 and don't have anything on it right now. The clock appears to have the correct time. I know what skew means, but what does it mean in terms of freeswitch? What is it skewing from?

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Voicemail directory
by Brian Snipes 06 May '08

06 May '08

Does the current mod_voicemail ( or any other part of fs ) currently support a user directory lookup feature? I'm thinking along the lines of 'Type in the first 3 letters of a persons last name' sort of lookup app. Brian

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Meeting this wednesday.
by Brian West 05 May '08

05 May '08

FreeSWITCHers, This weeks agenda can be found on our Weekly Conference Wiki Page. http://wiki.freeswitch.org/wiki/Weekly_Conference_Call Please be sure to review last weeks also. Thanks, Brian West PS: Add to it anything you think we need to go over.

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Session Question
by Nick Temple 02 May '08

02 May '08

Hello, I am extremely new to VOIP in general and have been using freeswitch only a couple of days, so if I have missed something in the docs please point me in the right direction. I am attempting to create an outbound call system -- mostly research and playing at this point. The initial goal is to have the system ring a number, then deliver a wave file. I have it (mostly) working in two different flavors. The first seems to work the best right now: (works fine via XMLRPC or the CLI): originate sofia/gateway/lesnet/1502NNNNNNN '&javascript(call1.js)' call1.js: session.waitForAnswer(10000); session.streamFile("/tmp/test.wav"); exit(); The second way does everything in Javascript: jsrun call2.js where call2.js is nearly as trivial: sessionx = new Session(); if (sessionx.originate(session, "sofia/gateway/lesnet/15025482003", 30)) { sessionx.waitForAnswer(10000); // Wait 30 seconds sessionx.streamFile("/mnt/test.wav"); }; exit(); The difference is that while both methods work on my land line, only #1 works when calling my cell phone ... for #2, the call is made, but I can't hear the media file playing. After lots of experimentation, the only thing I can figure at this point is that call1.js has some sort of defaults setup that I fail to do in call2.js Thanks for a great product! Nick

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Binding freeswitch to a specific ip address
by Brian Snipes 02 May '08

02 May '08

What is the best way to accomplish this? Can 'local_ip_v4' be set in vars.xml and have it override any autodetection of the IP address on the server? Thanks, Brian

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Freeswitch Video Pass-through
by Chris Chen 02 May '08

02 May '08

Can anybody who actually put the video working in Freeswitch share their settings and experience here? I tried the video recording in the default dialplan and also tried to have two SIP endpoints video calling each other, all not working. I put the h263-1998, H263-2000, H264 codec in the global codec list, and tried Xlite 3.0 and Bria 2.2 with camera. Thanks, Chris

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