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Simple IAX2 setup - help with converting from asterisk to freeswitch
by Matthew Law 26 May '10

26 May '10

Hi, I have compiled freeswitch trunk from svn on debian. I am trying to convert the following simple working asterisk config to freeswitch and I would be really grateful if someone could point me in the right direction: pbx:/etc/asterisk# cat iax.conf [general] bindport=4569 bindaddr=my.ip.address jitterbuffer=yes disallow=all allow=ulaw allow=alaw context=deadend [voiptalk] type=peer username=XXXXXX secret=XXXXXX host=iax.voiptalk.org [0843XXXXXX] type=friend username=08433XXXXXX context=incoming requirecalltoken=auto [1000] type=friend host=dynamic mailbox=1000 secret=XXXXXX context=phones requirecalltoken=auto pbx:/etc/asterisk# cat extensions.conf [general] autofallthrough=yes [outgoing] exten => _0[1-9].,1,Dial(IAX2/XXXXXX@voiptalk/44${EXTEN:1}) exten => _00.,1,Dial(IAX2/XXXXXX@voiptalk/${EXTEN:2}) [internal] exten => 901,1,VoiceMailMain() exten => 901,2,Hangup() exten => 902,1,MeetMe(1234,cdM) exten => 902,2,Hangup() [incoming] exten => XXXXXXXX,1,Dial(IAX2/1000,30) exten => XXXXXXXX,2,VoiceMail(1000@internal) [phones] include => internal include => outgoing pbx:/etc/asterisk# cat voicemail.conf [internal] 1000 => XXXX,Matt,me(a)myomain.com format=wav49 maxsilence=0 Sorry to be a bonehead, but I'm struggling with the wiki docs (especially with regard to IAX2) and also concerned about security. I installed the samples when I compiled freeswitch but wondering if that is a security risk as the PBX box is on the public internet? Many thanks, Matt

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call to the phone registering with xml_curl failed
by Erwin Davis 26 May '10

26 May '10

Hi, I have two phone (x-lite), one (ext 1018) registering with xml_curl and the other (ext 1001) registering locally. I was able to make call from ext 1018 to ext 1001. But the call from ext 1001 to ext 1018 failed. I cut and pasted the console log in http://pastebin.freeswitch.org/13044. Any clue? Thanks, e

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Re: [Freeswitch-users] Questions on Building an application for FreeSWITCH
by Kristian Kielhofner 26 May '10

26 May '10

Are you going to be calling my cell phone to ask if I owe $10,000 or more to the IRS? Seriously, the world does not need another robodialer to harass people... With that being said, write your app in whatever you want using ESL and connect to the socket to originate calls and listen for events. -- Kristian Kielhofner http://blog.krisk.org ----- Original Message ----- From: freeswitch-users-bounces(a)lists.freeswitch.org <freeswitch-users-bounces(a)lists.freeswitch.org> To: freeswitch-users(a)lists.freeswitch.org <freeswitch-users(a)lists.freeswitch.org> Sent: Wed May 12 15:40:58 2010 Subject: [Freeswitch-users] Questions on Building an application forFreeSWITCHI am looking for a boot in the right direction on the following project. I am looking to build an "application" with FS where it will frequently check a database and then make an outbound call when a record with certain conditions is found in the DB. It needs to determine if the call was then answered by a machine/vm, busy, or answered by a person, play a message and get a response back to be recorded in the db. It may find many calls that need to be made at the same time. It will be running on a PRI. I am confused as to what approach to use to accomplish this. Lua Script, Perl Script, Javascript, Phython, PHP, Ruby....etc. Liverpie, ESL....etc. Or a totally different approach. I realize that there is not only one "correct" way to build this, but I am looking for advise from those who have done this type of thing before. I value the opinion of those who have gone down this road and would be willing to share their thoughts on a recommended path to take to accomplish this. Thanks _______________________________________________ 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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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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