April 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Problem sending DTMF using an FXS channel
by devel＠thom.fr.eu.org 08 Apr '10

08 Apr '10

Hello, Im having trouble with calls to remote IVR using DTMF, when the A leg is an FXS port. What happens is when the key is pressed on the phone, the DTMF is sent inband to the callee party as voice, but also detected by freeswitch and so resent by freeswitch to the callee party. This results in unusability of called IVR. Is there any setting that could be used to prevent freeswitch from detecting DTMF and/or prevent freeswitch from resending the DTMF. Thanks François

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Help to call a VXML SIP number
by velusamy Krishnan 08 Apr '10

08 Apr '10

Dear All, I have received the XXX(a)sip.voxeo.net from my VXML application. I have written the following dial plan to call that number. <extension name="VXML"> <condition field="destination_number" expression="^(999)$"> <action application="bridge" data="sofia/external/ 9996079973(a)sip.voxeo.net"/> <action application="hangup"/> </condition> </extension> I have registered the FreeSWITCH and called the 999 extension. But the call was just trying. Finally I have got the NO_ANSWER information in console. 2010-04-06 15:12:23.001741 [INFO] mod_dptools.c:2353 Originate Failed. Cause: NO_ANSWER Is there configuration I need to do to call this number?? Please help me??

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Reload Configurations
by Aloysius Lloyd 08 Apr '10

08 Apr '10

Hi All, I add some SIP trunks to FreeSWITCH. reloadxml reload mod_sofia - Not allow to load the configuration when there are active calls. How to load the configurations when there are active calls in the system. Thanks Lloyd

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FS cannot produce any log file
by Bi,Xianchun 08 Apr '10

08 Apr '10

Hi, I have installed FreeSWITCH in my test server, the OS is Ubuntu 8.04 Server In module configuration file conf/autoload_configs/modules.conf.xml, log_file module is set active. <load module="mod_logfile"/> And in the log_file configuration file conf/autoload_configs/logfile.conf.xml, the settings are as following: <configuration name="logfile.conf" description="File Logging"> <settings>  <param name="rotate-on-hup" value="true"/> </settings> <profiles> <profile name="default"> <settings>  <param name="logfile" value="/var/log/freeswitch.log"/>  <param name="rollover" value="10485760"/>   </settings> <mappings>  <map name="all" value="debug,info,notice,warning,err,crit,alert"/> </mappings> </profile> </profiles> </configuration> But the logfile never appers. I also try using touch to create a dummy file, but the file never grow. Restarting FreeSWITCH has no effect. What wrong with my settings? Anyone could give any suggestion? Thanks in advance.

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fatal: unrecognized argument: --format=%h %ci
by Mark Campbell-Smith 08 Apr '10

08 Apr '10

Hi! I just performed a 'git pull %% make install' and noticed the following: Compiling src/switch_apr.c ... Compiling src/switch_buffer.c ... Compiling src/switch_caller.c ... Compiling src/switch_channel.c ... fatal: unrecognized argument: --format=%h %ci Compiling src/switch_console.c ... Compiling src/switch_mprintf.c ... Compiling src/switch_core_media_bug.c ... is the fatal argument anything to worry about? Thanks

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xml_odbc module
by Henry Huang 08 Apr '10

08 Apr '10

I just downloaded the 1.0.6 version and some previous trunk versions, but I can't seem to be able to locate xml_odbc source code anymore. Our SIP registration is utilizing it for realtime user adding.... Can anyone tell me what happen to that module? or it's just been forgotten to put on? thanks -- Henry Huang UniC Solution - Communication Unified VoIP & Open Source software Consultant

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Causing a Polycom phone to dial by a remote instruction
by Yehavi Bourvine 08 Apr '10

08 Apr '10

Hello, I would like to offer our users a WEB interface to manage their calls and initiate a call through this user interface. I can create two call legs via FreeSwitch and bridge them together; however, the call would show on the phone as an incoming call and not outgoing call. Is there a way to command a Polycom from remote to dial a number? Thanks! __Yehavi:

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Call Screener
by Todd 07 Apr '10

07 Apr '10

Hey List- I have a fresh FS install on Centos 5. I am trying to use the limit_hash application to screen caller ID numbers before transferring calls to another call server on my internal network. <action application="limit_hash" data="cidlimit ${caller_id_number} 50/1" /> <action application="transfer" data="sip/sofia/192.168.15.2" /> Is this the correct/best way to transfer all calls to another SIP server that will be managing all calls? Where is the best place to put this application so that it will run on every incoming call from several SIP providers? I have installed all the sample data, and have several phones registered to extensions that can make internal calls. I am just looking for the general outline of where these applications should go after I have setup the several SIP gateways for my external providers. I am still noo and would greatly appreciate any advice as to the best way to configure FS to simply receive all calls from the outside, run the limit_hash, then transfer to another SIP server. Thanks List!

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Setting "accountcode" Dynamically
by Jerry Richards 07 Apr '10

07 Apr '10

Regarding the "accountcode" channel variable in FS: http://wiki.freeswitch.org/wiki/Variable_accountcode. Is there any example or method of dynamically setting this accountcode for a the call has already been established? This would apply for a case where a call is received and the callee would like to set the accountcode during the conversation. Thanks, Jerry

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mod_dahdi_codec compile
by Christian Hiller - Baig Tel LTD 07 Apr '10

07 Apr '10

Hello, i would like to use mod_dahdi_codec, but when compiling from trunk it doesnt show up in /usr/local/freeswitch/mod What options are nesesarry? chris

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