- FreeSWITCH-dev - FreeSWITCH Mailing Lists

Announcing the FreeSWITCH Technology Preview VMWare Appliance.
by Ken Rice 15 Jan '09

15 Jan '09

FreeSWITCH + Centos 5.2 rolled into a VMWare appliance that's compatible with VMWare for Windows, VMWare Player, and VMWare Fusion on the Mac. Couple of things you need to know. 1) This is a Centos 5.2 X86_64 that was updated before we started testing it. 2) FreeSWITCH SVN Trunk is installed and operational. See /etc/motd on the running image for all the good information. We'll be unvailing a wiki page for this shortly. For now you can get the head start by downloading this at http://files.freeswitch.org/FreeSWITCH-VM1.0.0.zip Have fun guys! Ken krice(a)freeswitch.org krice(a)rmktek.com

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Announcing the FreeSWITCH Technology Preview VMWare Appliance.
by Ken Rice 15 Jan '09

15 Jan '09

Hey guys, I'm not trying to start 1 a day releases, Things just happened to fall that way... FreeSWITCH + Centos 5.2 rolled into a VMWare appliance that's compatible with VMWare for Windows, VMWare Player, and VMWare Fusion on the Mac. Couple of things you need to know. 1) This is a Centos 5.2 X86_64 that was updated before we started testing it. 2) FreeSWITCH SVN Trunk is installed and operational. See /etc/motd on the running image for all the good information. We'll be unvailing a wiki page for this shortly. For now you can get the head start by downloading this at http://files.freeswitch.org/FreeSWITCH-VM1.0.0.zip Have fun guys! Ken

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Reg: Configuring the Freeswitch
by Sanju 13 Jan '09

13 Jan '09

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Skypiax, Skype compatible endpoint
by Giovanni Maruzzelli 12 Jan '09

12 Jan '09

Ciao FreeSWITCHers, mod_skypiax is now usable, for Skype calls and finding bugs :-). Its wiki page (http://wiki.freeswitch.org/wiki/Skypiax, at the moment it's just the README files concatenated) begins like that: WHAT IS SKYPIAX This software (Skypiax) uses the Skype API but is not endorsed, certified or otherwise approved in any way by Skype. Skypiax is an endpoint (channel driver) that use the Skype client as an interface to the Skype network, and allows incoming and outgoing Skype calls from/to FreeSWITCH (that can be bridged, originated, answered, etc. as in all other endpoints, eg sofia/SIP). Think at Skypiax as similar to OpenZAP for analog lines: for each channel you need an interface (a Skype client). So, for eg, for two concurrent calls, you will need two channels, two Skype clients running on server. If your server's Skype client(s) has got the Skype credits, Skypiax works for SkypeOut calls too. You can use it from the dialplan, eg with the provided modified "default.xml" dialplan, you can call "sip:skype/remote_skypename__OR__skypeout_phonenumber" for calling via the Skype network from a SIP softphone to remote_skypename or to a phone number via SkypeOut, or you can call the "2908" extension from any phone to be bridged to the Skype Test Call). With the provided skypiax.conf.xml all incoming Skype calls will be routed to the "5000" extension, the IVR in default FreeSWITCH installation. On Linux the Skype client uses a lot of CPU. To lower its CPU consumption, you can use the Xvfb "fake" X server and (more important) the snd-dummy ALSA "fake" sound driver. Scripts are provided for this. But for a low number of channels it would works with regular X servers and ALSA drivers. On a Linux machine with 3GB ram and a quad core intel6600, we got no problem with 20 concurrent calls, and plenty of room for adding more Skypiax channels (100? not tested). On Windows, no need to do anything special, the Skype client is lighter on CPU. Skypiax is now pre-beta, but usable for testing and finding bugs :-). You can download Skypiax source code with subversion with the command: svn co http://svn.freeswitch.org/svn/freeswitch/branches/gmaruzz/src/mod/endpoints… mod_skypiax then, follow the README file in the mod_skypiax directory. More info on skypiax: http://wiki.freeswitch.org/wiki/Skypiax http://www.celliax.org Sincerely, Giovanni Maruzzelli ========================================= Company : Celliax Website: www.celliax.org Address : via Pierlombardo 9, 20135 Milano Country/Territory : Italy Business Email: gmaruzz at celliax dot org Cell : 39-347-2665618 Fax : 39-02-87390039

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mod_skypiax inching forward
by Giovanni Maruzzelli 09 Jan '09

09 Jan '09

Hi FreeSWITCHers! mod_skypiax, the Skype compatible endpoint, is slowly inching toward release :-) When the demo is online (will go on and off for development), you can test it (so helping finding bugs) by calling with Skype the Skype Names: skypiax20, skypiax19, skypiax18, ...., skypiax1 Happy New Year !!! Sincerely, Giovanni Maruzzelli ========================================= Company : Celliax Website: www.celliax.org Address : via Pierlombardo 9, 20135 Milano Country/Territory : Italy Business Email: gmaruzz at celliax dot org Cell : 39-347-2665618 Fax : 39-02-87390039

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FreeSwitch query
by Earlence Fernandes 09 Jan '09

09 Jan '09

Hi, I am trying to develop a VoiceChat app for the Android platform. I came across FreeSwitch from the XMPP site where it is listed as a Jingle compatible server. I have a few questions. 1. Does FreeSwitch act as a VoIP server on which I can register user accounts?. 2. If so, can I use any SIP stack(most probably in Java) to connect to this server and establish voice calls between two custom built clients? Cheers, Earlence

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Crashes on Win32, sqlite problem?
by Giovanni Maruzzelli 09 Jan '09

09 Jan '09

On Vista 32 bit, fully updated, VC Express 2008, I often (once in a day roughly) have crashes with: Assertion failed: inMutex, file ..\..\sqlite\src\os_win.c, line 1619 Seems a known issue: http://jira.freeswitch.org/browse/FSCORE-180 For me, it seems unrelated to traffic or whatever... If I can, I'll find out more, this mail just for not forgettin :-) Sincerely, Giovanni Maruzzelli ========================================= Company : Celliax Website: www.celliax.org Address : via Pierlombardo 9, 20135 Milano Country/Territory : Italy Business Email: gmaruzz at celliax dot org Cell : 39-347-2665618 Fax : 39-02-87390039

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mod_fax
by Steve Underwood 08 Jan '09

08 Jan '09

Hi all, I finally started to play with mod_fax today. First, a couple of little observations. Although there is a config file for fax, modules.conf.xml doesn't contain an entry for mod_fax, and dialplan/default.xml doesn't contain a demo like <extension name="test_rxfax_stream"> <condition field="destination_number" expression="^9011$"> <action application="answer" /> <action application="playback" data="silence_stream://2000"/> <action application="rxfax" data="rxfax.tif"/> <action application="hangup"/> </condition> </extension> <extension name="test_txfax_stream"> <condition field="destination_number" expression="^9012$"> <action application="txfax" data="txfax.tif"/> <action application="hangup"/> </condition> </extension> as it does for other modules. For more serious things..... If the far end of a SIP FAX transaction sends a reinvite to switch to T.38, FS sends a 488 back and everything fouls up. Other boxes send back the previous codec as the new one to use, and everything carries on smoothly in audio mode. I'm not a SIP expert, so I don't know the details of what it says on the topic, but in the real world successful continuance of a call requires a response other than 488. As an aside, the called party should be the one to initiate an attempt to use T.38, but in the real world the calling party often does. If T.38 is not available (which it isn't ever right now), and the call starts with a low bit rate codec, we should initiate a reinvite to use Alaw or ulaw. If that fails we might as well abandon the call. mod_fax currently follows the practice of my old and crude demo programs for *, and has apps called rxfax and txfax. This is taking a very narrow view of a FAX machine, and I think is too limiting. I think the following is how things should be: - One app, probably just called FAX. - It will be started with a flag saying if it should act as the calling party or the called party. - The app will be given optional lists of files to send, and files to receive. - The app will do its best to exchange all the files it can, including the use of poll mode FAXing. The module documentation says page by page events should be added (which spandsp supports), and this seems a sound idea. FAXback and other services might be implemented through this. Regards, Steve

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Audio formats without transcoding
by seven du 07 Jan '09

07 Jan '09

Brian, I tested using originate {ignore_early_media=true,absolute_string=g729,ringback=/sounds/ somefile}sofia/default/1000 &bridge({ignore_early_media=true, ringback=/sounds/somefile,absolute_codec_string=g729}sofia/gateways/ xxxx/00000) To listen the ring back, you need set ignore_early_media, the ringback variable maybe not needed at A-leg To make codec g729 work, I commented one line in switch_g729_init in mod_g729.c: // codec->flags |= SWITCH_CODEC_FLAG_PASSTHROUGH; However, I need to carefully make sure no transcoding happens, see http://lists.freeswitch.org/pipermail/freeswitch-dev/2009-January/001750.ht…

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New FreeSWITCH release
by Michael Collins 06 Jan '09

06 Jan '09

FYI, If you haven't already heard, we've now released v1.0.2 of FreeSWITCH! Please digg the new release story: http://digg.com/software/FreeSWITCH_New_Release_For_The_New_Year The source can be downloaded here: http://files.freeswitch.org/ in both tar.gz and tar.bz2 formats A Windows MSI file can be downloaded here as well: http://files.freeswitch.org/freeswitch-1.0.2.msi Thanks for your support and keep on FreeSWITCHing! -MC (mercutioviz)

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