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phone callers muted when joining to conference
by deniro 01 Apr '11

01 Apr '11

Hi, When join to the conference through perl program $session->execute("conference",conf-name@conf-profile); somehow phones were getting muted is there any parameters to pass while joining conference that will prevent phone callers be muted automatically thx deniro--

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Re: [Freeswitch-users] invite in conference
by Madovsky 01 Apr '11

01 Apr '11

I have more info after dozen different tests. If I invite in conf the same number several time, each time the invited leg answers, like 1 second of latency is added (exponential) so after 3 invites hangups I got 8 seconds of latency for the conference moderator voice in the invited phone. concerning the invited voice into the conference the latency stays exactly the same after 3 invites. Sorry I didn't triy to do it with 3 different numbers as my cell is cut (no credits) and my landline phone also (bill due).. ;) Thanks ----- Original Message ----- From: Madovsky To: FreeSWITCH Users Help Sent: Wednesday, March 30, 2011 6:27 PM Subject: Re: [Freeswitch-users] invite in conference /usr/local/freeswitch/bin/fs_cli -x "conference confText dial\{inconf=true,originate_timeout=20,ignore_early_media=true,instant_ringback=true}user/11111 22222 hiConf" and /usr/local/freeswitch/bin/fs_cli -x "conference confText dial\{inconf=true,originate_timeout=20,ignore_early_media=true,instant_ringback=true}loopback/11111 22222 hiConf" is this dial event can be in other place that conference::maintenance in ESL ? thanks ----- Original Message ----- From: Michael Collins To: FreeSWITCH Users Help Sent: Wednesday, March 30, 2011 1:20 PM Subject: Re: [Freeswitch-users] invite in conference what syntax are you using for the invitation? I would like to try it on my system and see if i can reproduce. -MC On Wed, Mar 30, 2011 at 9:27 AM, Madovsky <infos(a)madovsky.org> wrote: When I invite in conference, I can't see any conference esl event of the new member invited and accepted in conference is it normal ? 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 ---------------------------------------------------------------------------- _______________________________________________ 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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user and public dialplan
by Madovsky 31 Mar '11

31 Mar '11

example: - 9999 extension exisits in conf/directory - no public dialplan that matches 9999 external call is coming to public dialplan. is FS will consider that 9999 exists or not ? Thanks

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Multitenant and text chat via sip
by dome＠tel.co.th 31 Mar '11

31 Mar '11

Hi all, When i setup multitenant text chat via sip don't work. i got error invalid profile 2011-04-01 00:32:08.114100 [ERR] sofia_presence.c:133 Chat proto [sip] from [0838833133(a)wellcom.callbuzz.net] to [0816167580(a)wellcom.callbuzz.net] Xxx Invalid Profile wellcom.callbuzz.net wellcom.callbuzz.net is one of domain. It's bug or miss config somewhere in sip profile ? BG Dome C.

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PRI Test Equipment
by Max Clark 31 Mar '11

31 Mar '11

I'm looking for an appliance that can be plugged into a PRI connected to a PBX and either display or read back the digits that are passed to it. Before I build something to simulate this, is there anything out there commercially that can be purchased? Thanks, Max

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Knowing TON of each call
by Nagalenoj H. 31 Mar '11

31 Mar '11

Dear Friends, I'm using ftmod_sangom_isdn in my setup and I would want to know the TON(Type of Numbering) presented for each call. Is it possible to know the TON through channel variable in freeswitch?! Based on the Type Of Numbering of a call, I would want to modify the caller_id to a standard format and store in my database. It's always pleasure working with FS. Thanks. -- Nagalenoj H.

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Re: [Freeswitch-users] Freeswitch latest source rebuild issue under Windows
by Andrew Keil 31 Mar '11

31 Mar '11

Jeff, Just tried the latest (git head). Note: autocrlf option is set to false (tested via: git config --get-all core.autocrlf) Almost there. The original fatal error has gone now. The only error left is the following: ------ Build started: Project: mod_managed, Configuration: Debug_CLR Win32 ------ freeswitch_managed.cpp freeswitch_managed.cpp : fatal error C1192: #using failed on 'C:\FreeSWITCH\Win32\Debug\mod\FreeSWITCH.Managed.dll' 'The system cannot find the file specified.' freeswitch_wrap.2010.cxx freeswitch_wrap.2010.cxx : fatal error C1192: #using failed on 'C:\FreeSWITCH\Win32\Debug\mod\FreeSWITCH.Managed.dll' 'The system cannot find the file specified.' mod_managed.cpp mod_managed.cpp : fatal error C1192: #using failed on 'C:\FreeSWITCH\Win32\Debug\mod\FreeSWITCH.Managed.dll' 'The system cannot find the file specified.' Running Code Analysis for C/C++... Generating Code... I tried to rebuild a few times, however the same error above happened. Any ideas? See attached for my build output log. Regards, Andrew Keil -----Original Message----- From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Jeff Lenk Sent: Wednesday, 30 March 2011 3:39 PM To: freeswitch-users(a)lists.freeswitch.org Subject: Re: [Freeswitch-users] Freeswitch latest source rebuild issue under Windows Please try git head just submitted fixes -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-latest-source-rebu… Sent from the freeswitch-users mailing list archive at Nabble.com. _______________________________________________ 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 __________ Information from ESET NOD32 Antivirus, version of virus signature database 5998 (20110329) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com

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RTP Proxy mode, DTMF conversion problem
by Mac 31 Mar '11

31 Mar '11

Dear ALL, I am struggling for some time with DTMF issue. Let me introduce you my configuration. The main task is to configure RTP Proxy with full topology hiding - OPERATOR_1 ------- Freeswitch ------- OPERATOR_2 Here is a prt of my config: - sip profile. I have one new sip profile with two gateways (OPERATOR_1 and OPERATOR_2) <include> <gateway name="OP1"> <param name="username" value="X1.Y1.V1.Z1"/> <param name="password" value="09321234"/> <param name="proxy" value="X1.Y1.V1.Z1"/> <param name="from-domain" value="test.com"/> <param name="register" value="false"/> <param name="caller-id-in-from" value="true"/> <param name="extension" value="test"/> <param name="extension-in-contact" value="true"/> <param name="outbound-codec-prefs" value="$${outbound_codec_ prefs}"/> <param name="disable-transcoding" value="true"/> <param name="inbound-late-negotiation" value="true"/> </gateway> </include> - dialplan <include> <context name="public"> <extension name="term"> <condition field="network_addr" expression="^(X1\.Y1\.V1\.Z1|X2\.Y2\.V1\.Z1)$" break="on-false"/> <condition field="destination_number" expression="^46(\d{9})$"> <action application="set" data="proxy_media=true"/> <action application="set" data="bypass_media=false"/> <action application="start_dtmf"/> <action application="bridge" data="sofia/gateway/OP1/$0"/> </condition> </extension> </context> </include> Everything is fine, but i have problem with DTMF conversion from RFC2833 to inband. Refering to http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf and http://wiki.freeswitch.org/wiki/Proxy_Media i have done all the necessary things. vars.xml <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMA,G.729,PCMU,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMA,G.729,PCMU,GSM"/> The after-effect is following output: [ERR] switch_core_io.c:1055 Codec PROXY PASS-THROUGH encoder error! Placing <param name="dtmf-type" value="none"/> does not help anyway. Could sb point me where the problem is located? Thanks in advance, Mac

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How to check the FS source code version on Windows platform
by Charles 31 Mar '11

31 Mar '11

Can anyone help advise how to check the exact version of Freeswitch on windows platform? I can see the info like screenshot below in the properties of FS folder which git from latest source, it the correct way? if yes, the version should call 'Git-Head: d5ef86d7788ef0080ca3be7e2ff39bda989d4b4d ', however, it looks very strange... thanks. 2011-03-30 Charles

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Re: [Freeswitch-users] set_user effect
by Madovsky 31 Mar '11

31 Mar '11

sounds weird. sip_from_user took the new set_user user value but not sip_from_user_stripped. so now I use sip_from_user_stripped to get the original caller_id_number.... ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Wednesday, March 30, 2011 9:13 PM Subject: set_user effect If I change user with set_user, it seems that all the sip_from_xx are changed according to the new set_user value.But how can I get the original sip_from_xxx vars ? Thanks

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