January 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Choppy VM when using loopback (on Debian lenny)
by Øyvind Albrigtsen 03 Feb '11

03 Feb '11

Hi, I'm using latest git-version of Freeswitch, and when I go to voicemail when calling a number the sound playback is choppy and it skips some of the digits in the number I called. If I set sleep to 2000 it sounds better, but it's still a little choppy and still skips some of the digits. This is with the default settings coming with the git version. When I call 4000 to listen to voicemail it sounds fine. The kernel is the default kernel which is compiled with CONFIG_HZ_250=y Regards Oyvind

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Is it possible to use att_xfer on channels already bridged via loopback?
by Dmitry Sytchev 03 Feb '11

03 Feb '11

Is att_xfer or mod_loopback is broken in FS-current? I use FreeSWITCH Version 1.0.head (git-7eceff4 2011-01-16 22-33-50 +0000) Seems there were no updates of att_xfer or mod_loopback since that. I use loopback channel as destination when doing att_xfer to re-enter dialplan. With loopback_bowout=false and loopback_bowout_on_execute=false this works. But when any of connected parties tries to do att_xfer again, all channels get hangup on transferer hangup. Scenario: A calls B, B answers A launches att_xfer via *7, B listens to MOH A dials C and we do att_xfer to loopback/C C answers, A hangs up to complete transfer C and B are now bridged via loopback, `show channels` shows 4 channels include 2 loopback legs. Now, C also tries to do in-call transfer with *7. C launches att_xfer via *7, B listens to MOH C dials D and do att_xfer to loopback/D D answers, C hangs up to complete transfer B and D are hung up instead of be bridged together. There are also issues with MOH wile running att_xfer, but they are not so important as att_xfer behavior itself. -- Best regards, Dmitry Sytchev, IT Engineer

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Caller ID using Fifo
by Joao Leme 03 Feb '11

03 Feb '11

Hi there, I would like to know if there is a way to see the caller ID on my Sip Client (X-Lite for example) of the caller that I answear from a Fifo queue? Thanks, John

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execute_on_ring executing on answer
by François Delawarde 02 Feb '11

02 Feb '11

Hi, Doing some testing with this morning's git (Fri Jan 28) I just found out that the execute_on_ring application runs when the destination answers instead of when it rings. So far, I can't seem to find out the reason. Could it be some configuration issue? Here a call log showing the phenomenon with a simple bridge: <action application="bridge" data="[execute_on_ring=info,absolute_codec_string=^^:PCMA:G722]sofia/192.168.10.1/sip:2103@192.168.10.22:5060"/> http://pastebin.freeswitch.org/15168 Thanks, François.

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open g729
by Gustavo Espeche 02 Feb '11

02 Feb '11

Hello, some one can compile open g729 to work with freeswitch? http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ i appreciate a lot if some one has some experience in it. Best Regards. Gustavo Espeche www.easyipcall.com

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keeps receiving 407
by jesse 02 Feb '11

02 Feb '11

I set up a sipp uac client. here is the call flow messages: http://freeswitch.pastebin.com/Sx5Z4egM why does FS send 407 after 200 OK? thanks!

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dingalingissue
by budi wibowo 02 Feb '11

02 Feb '11

hi, anybody has "no voice" experience when call using mod dingaling? i try from my gmail account and call still normal but from FS always no voice. it was working perfectly before thx budi

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freeswitch.com is returned to us!
by Anthony Minessale 01 Feb '11

01 Feb '11

http://www.freeswitch.com/ Thank you everyone who helped with this! -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900

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Developer Dinner Coming Up - Help Feed The FreeSWITCH Team!
by Michael Collins 01 Feb '11

01 Feb '11

Well, it's that time again. The FreeSWITCH developers are gathering at a secret location in Wisconsin next week and the community is invited to participate in this year's "buy the devs dinner" event. How to do it: Go to freeswitch.org and click on the Paypal link. Toss a few bucks in the hat and send along your bon apetit message. If for some reason you cannot use Paypal but still wish to participate then by all means call Brian West on his SIP phone: 2000(a)bkw.org. (Hint: you can dial 9191 if you have a default FreeSWITCH config.) Brian is well-prepared to take your money. :) To those who have donated recently we say, "Thank you!" If you wish to donate again specifically for this occasion you are more than welcome to do so. In any case, thank you in advance for your generosity. You are great community members and the FreeSWITCH team is absolutely glad to have you with us! -Michael Collins

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minimum UDP packet size
by Madovsky 01 Feb '11

01 Feb '11

is it useful to set minimum UDP pakcet to the MTU rate like 1500 and avoid any fragmentation ? Thanks

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