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Re: [Freeswitch-users] Route incoming analog calls
by Eric Michel 02 Feb '11

02 Feb '11

Steve thank you for the pointers, you got me going in the right direction. Here is what I came up with (so far it seems to work). <extension name="Line 2(6820)" continue="true"> <condition field="source" expression="mod_freetdm"/> <condition field="${channel_name}" expression="^(FreeTDM/2:2/)$"> <action application="info"/> <action application="set" data="domain_name=$${domain}"/> <action application="set" data="transfer_ringback=$${us-ring}"/> <action application="answer"/> <action application="transfer" data="2000"/> </condition> </extension> All I have to do is modify the second number in the expression. 2:3 line three, 2:4 line four etc. -Eric

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Need to change "identity display" in dialog-info event
by Helmut Kuper 02 Feb '11

02 Feb '11

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello is there a way to get the value of "effective_caller_id_name" in dialog-info's "remote identity display"? And further: Is it possible to get that value also accessible for display updates, so that after picking up a call the display is updated with the picked up call's effective_caller_id_name? regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.10 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iEYEARECAAYFAk1JjOUACgkQ4tZeNddg3dyx7wCffCCNvFCJczlmn9qKwvwXm8Ms cdYAnR7tkUactRxkkqg+9D2h8Ig9TQB2 =ZcKJ -----END PGP SIGNATURE-----

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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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How to prevent source channel hangup after uuid_bridge completion (outbound ESL)?
by Dmitry Sytchev 02 Feb '11

02 Feb '11

Hi I'm trying to originate call from outbound ESL and then bridge it to original channel where ESL is launched on. I use api originate with &park(), then uuid_bridge. Everything works as expected, but when second leg hangs up, first leg gets hangup too. How can I prevent first leg from hanging up and continue to process it in ESL? Thanks in advance -- Best regards, Dmitry Sytchev, IT Engineer

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Route incoming analog calls.
by Eric Michel 02 Feb '11

02 Feb '11

I've got a Sangoma A200 with six FXO ports and two FXS. I can call out just fine, but I'm having trouble figuring out how to route incoming calls. We've got five POTS lines and all I've been able to do is route all incoming calls to a specific extension. How do I go about routing individual POTS lines to different groups/extensions? <extension name="analog test2" continue="true"> <condition field="source" expression="mod_freetdm"> <action application="set" data="domain_name=$${domain}"/> <action application="set" data="transfer_ringback=$${us-ring}"/> <action application="answer"/> <action application="transfer" data="1000 XML default"/> </condition> </extension> That is my current conf/public/inbound dial plan. Shouldn't I be able to add a second condition that tests what FXO port is receiving the call and allow me to route it accordingly? If yes, could I get an example? The answer may be really simple, I've just been unable to find it. Thanks, Eric

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RTP jitter basic tutorial
by Madovsky 02 Feb '11

02 Feb '11

interesting http://toncar.cz/Tutorials/VoIP/VoIP_Basics_Jitter.html

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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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voicemail.tpl and sip address
by Madovsky 02 Feb '11

02 Feb '11

which var can I use in voicemail.tpl to write correctly a sip address ? actually vocemail_caller_id_number shows only th username and not the domain Thanks

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Re: [Freeswitch-users] Outbound only calls don't connect when bypass_media is true.
by Marcin Wojtowicz 02 Feb '11

02 Feb '11

I've found a solution file that creates the pcre_chartables.c source and compiled freeswitch from the latest git. The problem still remains, and it looks like it was brought up by somebody else here: http://lists.freeswitch.org/pipermail/freeswitch-users/2010-October/063891.… Here are the build errors, in order of appearance: ------ Build started: Project: libpcre, Configuration: Release Win32 ------ pcre_chartables.c c1 : fatal error C1083: Cannot open source file: 'pcre_chartables.c': No such file or directory (other files here compile ok) ------ Build started: Project: FreeSwitchCoreLib, Configuration: Release Win32 ------ Generating switch_version.h (stuff here goes ok) Generating Code... LINK : fatal error LNK1181: cannot open input file 'E:\downloads\freeswitch-1.0.7\libs\win32\pcre\Win32\Release\libpcre.lib' ------ Build started: Project: mod_spidermonkey, Configuration: Release Win32 ------ mod_spidermonkey.c LINK : fatal error LNK1181: cannot open input file 'E:\downloads\freeswitch-1.0.7\Win32\Release\FreeSwitchCore.lib' FreeSwitchCore link error repeats many times for other projects, too many to list. It doesn't build. ------ Build started: Project: mod_managed, Configuration: Release_CLR Win32 ------ freeswitch_managed.cpp freeswitch_managed.cpp : fatal error C1192: #using failed on 'E:\downloads\freeswitch-1.0.7\Win32\Debug\mod\FreeSWITCH.Managed.dll' 'The system cannot find the path specified.' freeswitch_wrap.2010.cxx freeswitch_wrap.2010.cxx : fatal error C1192: #using failed on 'E:\downloads\freeswitch-1.0.7\Win32\Debug\mod\FreeSWITCH.Managed.dll' 'The system cannot find the path specified.' mod_managed.cpp mod_managed.cpp : fatal error C1192: #using failed on 'E:\downloads\freeswitch-1.0.7\Win32\Debug\mod\FreeSWITCH.Managed.dll' 'The system cannot find the path specified.' Generating Code... The code I downloaded was from latest.freeswitch.org, the tar.bz2 zip file. >From the dependencies it looks like FreeSwitchCore depends on libpcre, so all this cascades from the missing pcre_chartables.c file. Can you send it to me?

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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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