February 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Session sleep method
by Malay Thakershi 03 Feb '11

03 Feb '11

Hello, I am using .NET module. There is sleep method that has two arguments. What is second argument for? I use it: mObjMainSession.sleep(200, 0); What happens if I pass something other than zero as second argument? I am having inconsistent delays between call paths and that is why I am exploring this argument. Thank you for any help. Malay

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conference audio in last git
by Madovsky 03 Feb '11

03 Feb '11

seems that the audio (audio files and voice) is less stable as before in conference only. some packet loss or like a noise gate with a too high threeshold also without CNG there's noise between ivr voice

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How to record a conference using javascript application
by kapil.rastogi 03 Feb '11

03 Feb '11

Hi, I want to record the chatting b/w all members during the conference. I am using the following statement in my javascript application: session.execute("conference", "conf-room--"+room_id+"@default record /usr/local/freeswitch/abc.wav"); But i am unable to record the conference room chatting. But when i record the same from fs_cli command line, it is working well. Please tell me how to record the conference chatting. Regards, ----- Regards, Kapil Rastogi Telemune Software Solutions P Ltd. kapil.rastogi(a)telemune.net +919013204760 -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/How-to-record-a-conference-us… Sent from the freeswitch-users mailing list archive at Nabble.com.

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ANNOUNCEMENT CudaTel IP PBX (powered by FreeSWITCH) Reaches Critical Mass
by Anthony Minessale 03 Feb '11

03 Feb '11

I have been splitting my time developing not only FreeSWITCH but the CudaTel Phone system by Barracuda Networks that uses FreeSWITCH as the telephony engine. I'm proud to announce that not only has the product reached a 2.0 status, it's now 100% deployed within Barracuda Networks running the entire company's phone services across multiple locations. The CudaTel is using FreeSWITCH right from GIT with no special modifications at all. The best part is it works great as a PBX you can then feed out to your existing FreeSWITCH setup to route the traffic to custom applications or whatever you can think of. The next step is to develop more exciting and impressive PBX features that can be added for free with an energize update package available at the time of purchase. CudaTel comes with the same great support you receive today on FreeSWITCH. This is also a great opportunity for those of your out there looking to get into the reseller market. Contact me at the consulting link at the top of this page or call 408-588-3633 to learn how you can become a reseller of CudaTel. Use your existing VoIP expertise to support and deploy CudaTel to others. Check it out at http://www.cudatel.com -- 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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mod_perl mod_lua
by deniro 03 Feb '11

03 Feb '11

Hi I have compiled version of freeswitch 1.6.x on ubuntu 10.04 I want to install extra freeswitch modules like mod_perl and mod_lua Is there any way to install these modules without re-compileling freeswitch. I looked at freeswitch site and it is talking about editing modules.conf file and compiling from source and I dont even have modules.conf file as I searched. thx deniro--

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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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Simple_conference.lua
by John Smith 03 Feb '11

03 Feb '11

Hi, I’m trying to have the script http://wiki.freeswitch.org/wiki/Simple_conference.lua running on my FreeSwitch server. I get an error when FS wants to play any phrase given in this example: *[ERR] switch_ivr_play_say.c:150 Can't find macro conference_welcome* * *It seems normal as I didn’t find any reference to such phrases in the different xml phrase configuration files. There seems also to be no adequate wav file in sounds/en/us/callie/voicemail/8000. My question is: Do these files exist somewhere or does the example assume that we have to record our own wav files? Thanks John

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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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Fs runninf on Clound Service (XEN)
by dome＠tel.co.th 03 Feb '11

03 Feb '11

Dear All, I'm testing FS on http://cserver.cloudwww.com/ then give me IP : 61.x.x.x and in eth0 eth0 Link encap:Ethernet HWaddr 00:16:3E:2C:91:DC inet addr:10.6.32.11 Bcast:10.6.33.255 Mask:255.255.254.0 UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 RX packets:160267 errors:0 dropped:0 overruns:0 frame:0 TX packets:93251 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:1000 RX bytes:189671769 (180.8 MiB) TX bytes:9669248 (9.2 MiB) So i can access 61.x.x.x without problem. but i want to know how to config sofia profile for binding 61.x.x.x in default config. i can send call to 61.x.x.x but when call out i got 10.6.32.11 and i can't find 61.x.x.x in sip header someone help me please BG Dome C.

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