May 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] G723 codec
by Budi wibowo 26 May '10

26 May '10

Hi, is it g723 in fs? Or just pass through? Other idea regarding codec issue is put transcoding box outside fs box. You will also have flexibility for conncted client. Any client any codec can connect Br Budi ------Original Message------ From: Steve Underwood Sender: freeswitch-users-bounces(a)lists.freeswitch.org To: freeswitch-users(a)lists.freeswitch.org ReplyTo: freeswitch-users(a)lists.freeswitch.org Subject: Re: [Freeswitch-users] G723 codec Sent: May 26, 2010 01:07 On 05/26/2010 01:42 AM, Madovsky wrote: > Hi all, > is there a chance that in future there will be a mod_g723 transcoder ? > I assume you mean G.723.1. It is very difficult to licence the patents related to G.723.1 unless you are a huge user. G.729 has a well defined scheme for licencing the relevant patents on an incremental basis, starting at a modest numbers of channels. G.723.1 is quite problematic. For that reason it is unlikely that G.723.1 will be available as a host software transcoder module any time soon. There are, however, hardware card options which can provide this codec. Steve _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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make current -> "Failed to add directory 'libs/openzap'"
by GillesToo 26 May '10

26 May '10

Hello After updating a test CentOS 5.4 host with "yum update", I wanted to also update Freeswitch. I followed the Installation Guide (http://wiki.freeswitch.org/wiki/Installation_Guide) thusly: --------- # cd /usr/src/freeswitch # diff build/modules.conf.in modules.conf [nothing shown] # make current [...] U libs/libsndfile/examples/sndfile-play-beos.cpp U libs/libsndfile/reconfigure.mk U libs/libsndfile svn: Failed to add directory 'libs/openzap': object of the same name already exists make: *** [current] Error 1 --------- Then, I ran "make current" a second time: --------- # make update cd . && /bin/sh /usr/src/freeswitch/build/config/missing --run aclocal-1.9 cd . && /bin/sh /usr/src/freeswitch/build/config/missing --run automake-1.9 --gnu configure.in:898: required file `build/freeswitch.pc.in' not found Makefile.am:256: ADD_LIBEDIT does not appear in AM_CONDITIONAL Makefile.am:263: ADD_LIBEDIT does not appear in AM_CONDITIONAL Makefile.am:268: ADD_ODBC does not appear in AM_CONDITIONAL make: *** [Makefile.in] Error 1 --------- FWIW, this host has an one-FXO OpenVox PCI card and things were working OK. Is there something I can try besides wiping out /usr/src/freeswitch/ and just compiling from scratch? Thank you. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/make-current-Failed-to-add-di… Sent from the freeswitch-users mailing list archive at Nabble.com.

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rxgain/txgain dahdi
by Troy Anderson 26 May '10

26 May '10

Ok, I'm googled out! Can anyone point me to some docs on how to set the rx and tx gain for an FXO or FXS channel using dahdi and mod_openzap? I think it goes into /etc/dahdi/system.conf, but can't find any docs on how or where. My /etc/dahdi/system.conf looks like this: loadzone=us defaultzone=us fxsks=1 fxsks=2 fxsks=3 fxsks=4 fxsks=5 fxsks=6 fxoks=7 fxoks=8 Thanks! -Troy

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vars set in external trunk
by Madovsky 25 May '10

25 May '10

Hi all, should I us $${var} or ${var} to call a var from a dialplan set in a external sip profile ? Thanks

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get a freeswitch var from PHP
by Madovsky 25 May '10

25 May '10

Hi all, I set a special var in my dialplan before the bridge that I'd like to get from PHP before the call is done (it's a var that define which trunk choosed from distributor module). Which is the elegantest way to do it ? Best Regards Franck

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IRC log for #openzap
by Nagalenoj H. 25 May '10

25 May '10

Hi friends, Where is the openzap channel's IRC log available? Has someone tried downloading it already? -- Regards, Nagalenoj H.

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curl_xml
by Erwin Davis 25 May '10

25 May '10

Hi, I am trying to use curl_xml to dynamically send back the response for directory query from a webserver. But when the FS received the response from the web server, it kept sending 403 forbidden response to the sip client (x-lite). I post the FS log from the console in http://pastebin.freeswitch.org/13007and the XML response log in http://pastebin.freeswitch.org/13007. Thanks for your help,

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G723 codec
by Madovsky 25 May '10

25 May '10

Hi all, is there a chance that in future there will be a mod_g723 transcoder ? Thanks Franck

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Mod_skinny: Call for testing!
by Mathieu Parent 25 May '10

25 May '10

Hello, Skinny Call Control Protocol (SCCP) is a VoIP protocol used on several phones including Cisco IP Phones and Nokia Call Connect 2.0. Contrary to SIP, the phone is controlled by the server. This allow powerfull and funny things (like controlling leds, display, ring type, ...) ; this also allow to use some phones with crappy SIP images, like (guess what?) Cisco ones. Since February, mod_skinny is in freeswitch trunk. Many things happened since. Today the module has reached several milestone (it compiles on windows, call info is shown early, state handling changed to a more FreeSWITCH manner, ...). Most of basic features and more are implemented (from Wiki): * Incoming and outgoing calls * Shared lines and multiple calls per lines * CallInfo set as early as possible * New call made from: OffHook, Softkeys (Redial, NewCall), Stimulus (LastNumberRedial, VoiceMail, SpeedDial) * Answering calls via: OffHook, Softkeys (Answer) * Holding/Unholding calls via Softkeys (Hold, Resume) or Stimulus (Hold), with MOH (music on hold) * Transfer via Softkeys (Transfer) or Stimulus (Transfer) * Ending call via: OnHook, Softkeys (EndCall) * Misc: SoftKeys (Backspace) * Handle firmware version request (VersionReqMessage) per device type or per device * Patterns for dialed numbers to process (patterns-dialplan and patterns-context) Some important missing ones: * Message Waiting indicator (MWI) * Bypass Media (help needed) * Codec Negotiation (help needed) * Forward (transfer already implemented) * Complete list: <http://wiki.freeswitch.org/wiki/Skinny_Call_Control_Protocol#TODO> How can you help: * test if it compiles on your platform * test if it works with your SCCP phone (only 7960 and 7961 have been tested yet) * if you don't have an SCCP phone, test using "perl src/mod/endpoints/mod_skinny/test-skinny.pl" (you need a recent perl, aka 5.10.1, 5.10.0 has buggy thread implementation, at least on Debian Lenny). * write docs about configuration * propose patches (some are easy like API Commands or events ; some are harder ;) * review the code (quality, security, ...) * If you own a Cisco Communication Manager and you know a feature that is not in mod_skinny, make network captures! Some links: * Instructions are at: http://wiki.freeswitch.org/wiki/Mod_skinny * Bugs should go here: http://jira.freeswitch.org/browse/MODSKINNY * Development stuff is at: http://wiki.freeswitch.org/wiki/Skinny_Call_Control_Protocol * Suggested features should go here: http://wiki.freeswitch.org/wiki/Skinny_Call_Control_Protocol#TODO (or bounty or JIRA) Any help is appreciated. happy testing/hacking ;) Mathieu Parent

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did using easyroute
by Meftah Tayeb 25 May '10

25 May '10

hello, i see with easyroute i cabn route just one number and no a range how do i route a range of numbers, like all number starting with 1747XXX to a special gateway, and 1748XXX to another? thanks

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