- FreeSWITCH-users - FreeSWITCH Mailing Lists

Lua Framework for FS plus more...
by EdPimentl 09 Jun '08

09 Jun '08

To those who like to know more about FS support for Lua and want to leverage a Lua framework. Here is link to an excellent Lua framework Hopefully this should accelerate the adoption of lua and make it easier for others to develop in lua If you rather develop in Python, do not overlook Web2Py, clearly way in the lead... yes even ahead of Django. :-) http://mdp.cti.depaul.edu/examples/default/index -E

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choppy audio during group_confirm_key playback
by jart 08 Jun '08

08 Jun '08

Hello, I'm setting up an 800 number to forward to my cell phone and I want to play a little indication so I know the call is coming from the 800 number so I can answer it correctly. I setup the following XML: <action application="set" data="ringback=${us-ring}"/> <action application="set" data="group_confirm_key=exec"/> <action application="set" data="group_confirm_file=playback indication.wav"/> <action application="bridge" data="sofia/external/MY_NUMBER(a)sip.gafachi.com"/> When I answer the call, I can hear "indication.wav" being played at the beginning of the conversation. I can understand what it's saying, but it really choppy and and repeats itself. After that, the call audio is crystal clear. On rare occasions, the indication is played perfectly. At first I thought this might be an issue with my voip provider, but I don't understand why call audio after the indication would be perfectly clear. Also the choppiness doesn't sound like cheap voip termination, but more like a bug. indication.wav is a normal wav file that I recorded through freeswitch: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz I wish I could send a recording of the audio, but the group_confirm_key thingy isn't recorded when I use record_session. Any help is greatly appreciated!

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Routing calls from FreeSwitch to Asterisk
by Alexander Seith 07 Jun '08

07 Jun '08

Dear list, I read that FreeSwitch is capable to route calls, even between different protocols. Is this correct? What I'd like to do is, installing FreeSwitch and Asterisk on the same machine, collecting all secure (e.g. SIPS/SRTP) calls with freeswich and forward them in tradional sip/rtp to asterisk. Can this be (easily) done with FreeSwitch? Hope this question isn't too dumb... Thanks for responses in advance Alex

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SIP Call-ID
by Jed Stafford 07 Jun '08

07 Jun '08

I'm working to get FreeSwitch working behind a SIP load balancer. Using the sip-ip param in a profile I'm able to make the outbound calls appear to come from the load balancer IP address. Debugging the sip messages, this all appears to be working correctly. However the load balancer determines which machine to send the calls too based on the SIP Call-ID. We want FreeSwitch in the middle of the call for billing and other purposes. But i'd like FreeSwitch to use the same SIP Call-ID that was sent to it when it initiates the next leg of the call. The hash on the load balancer only cares about everything proceeding the @ sign in the sip call-id. Is there a function in the dialplan to copy the source call-id and use it for the destination leg? Hopefully I've explained this well enough, any ideas would be helpful. Regards, -Jed

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mod_mono
by Michael S Collins 07 Jun '08

07 Jun '08

We've seen a flurry of activity with mono on svn. Could someone in the know give us a progress report? Thanks, MC Sent from my iPhone

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AFK (for 2 weeks)
by Anthony Minessale 06 Jun '08

06 Jun '08

I'll be going on vacation till June 21st so our production may lag just a teenie bit. Don't worry, we should be releasing 1.0.1 while I'm gone and with the slow down it gives our community time to prepare for the upcomming ClueCon 2008 http://www.cluecon.com -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale(a)hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale(a)gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888(a)conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888(a)conference.freeswitch.org> pstn:213-799-1400

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gsm file record/playback
by Miroslav Mostic 06 Jun '08

06 Jun '08

Hi everyone! I am probably missing something very basic, but I have problems with simple file record and playback in gsm file format. I have just installed 1.0 release system and I changed only default dialplan with following two extensions: <extension name="gsm_record"> <condition field="destination_number" expression="^9991$"> <action application="answer"/> <action application="record" data="/tmp/rec.gsm"/> </condition> </extension> <extension name="gsm_playback"> <condition field="destination_number" expression="^9992$"> <action application="answer"/> <action application="playback" data="/tmp/rec.gsm"/> </condition> </extension> When I place call to 9991extension I see INFO message about opening /tmp/rec.gsm, no error messages and when I hangup I can see that file /tmp/rec.gsm exists. However when I try to play this file using 9992 extension the only thing I can hear is noise. When I just change name to /tmp/rec.wav in both extensions everything is working fine. What am I missing? Many regards, Miroslav

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Forked dialing with different channelvars set...
by Rupa Schomaker (lists) 06 Jun '08

06 Jun '08

I have a seemingly simple problem that I can't figure out how to implement. Simplified Problem: I'd like to have my cell phone and my deskphone ring at the same time. This seems simple, but there is a small caveat. I need to have a channel variable set differently for my cellphone. Specifically: voicepulse requires you to have the caller id number and caller id name set to the number in order to pass it on to the PSTN. I want to see the callerid name on my deskphone however. === Ok, normally, I'd just do a: <action application="bridge" data="sofia/internal/101%$${domain},sofia/gateway/voicepulse/$${cellnumber}"/> but then my cell phone shows Unknown or Blocked when receiving the call. If prior to that I do: <action application="set" data="effective_caller_id_name=${caller_id_number}"/> Then I get the correct callerid on my cell, but only the callerid number on my deskphone. I asked on #freeswitch and had some hints, but they either went over my head or I didn't communicate my problem right. Using multiple extension entries with continue="yes" doesn't seem to execute them in parallel (so the deskphone and the cell don't ring at the same time). The "obvious" answer is how I implemented this in asterisk -- and maybe that is my problem, thinking in terms of asterisk. The bridge equivalent would be the dial() app. This allows me to dial back into the dialplan if I use the "local" channel. This gives a level of indirection that would then let me modify the channelvars as necessary. The only way I can see to "call" back into the dialplan is to use the transfer application. But that can't do multiple extensions in parallel so doesn't solve my problem. What am I missing? -Rupa

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g729
by Nicolas Brenner 06 Jun '08

06 Jun '08

Hi, I'd like to know what's needed to add support for G729, I know there's a bounty, but I couldn't make sense out of what's posted on the wiki. I'm really interested in this, as one of my current VoIP providers restricts me to using only this codec, which limits me to using Asterisk, hence I can't fully move to FS without G729 support. What would it take to make it happen? Thanks, -- Nicolás Brenner Medularis SpA

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trixswitch 0.4
by andrew 06 Jun '08

06 Jun '08

I released the latest version of trixswitch today. Its a fast way to get FreeSWITCH 1.0 running. trixswitch.org

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