September 2007 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Start javascript app
by Jonas Gauffin 06 Sep '07

06 Sep '07

Hello I want to start a javascript application with some parameters (without having an active session). Is that possible? A bit more details: I got a web administration site for my freeswitch solution. The user can manage all it's soundfiles from it. I want to let the user record files through freeswitch. A javascript application should be launched (with filename and user agent address as parameters) when the user presses a record button. The js app should then call the specified UA and record the specified file. I've come up with the following solution: 1. The webapp tells my server to record a file 2. My server saves filename + ua address 3. The server tells FS to originate a call to the UA by using the event socket interface 4. The JS and it's parameters are supplied when FS asks for the dialplan through mod_xml_curl The current solution works, but it would be nicer to be able to launch a js app directly. Best regards, Jonas

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Specify caller id when originating
by Jonas Gauffin 05 Sep '07

05 Sep '07

Hello I'm trying to specify caller_id_name and caller_id_num when originating, but it doesn't seem to work: originate {caller_id_name=Voicemail,caller_id_number=8888}sofia/mydomain.com/jonas voicemail The default caller id/number is still being used. //Jonas

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Cullis, Tim is out of the office.
by tim.cullis＠convergys.com 05 Sep '07

05 Sep '07

I will be out of the office starting 04/09/2007 and will not return until 12/09/2007. I will respond to your message when I return. -- "NOTICE: The information contained in this electronic mail transmission is intended by Convergys Corporation for the use of the named individual or entity to which it is directed and may contain information that is privileged or otherwise confidential. If you have received this electronic mail transmission in error, please delete it from your system without copying or forwarding it, and notify the sender of the error by reply email or by telephone (collect), so that the sender's address records can be corrected."

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Is Freeswitch a good option for dynamic routing?
by Chris Ziomkowski 03 Sep '07

03 Sep '07

I need to create a simple wholesale switching application that consists of receiving a call via H.323 or SIP, running a custom algorithm and lookup to translate the incoming DNIS into a new number, and the processing this new number through a standard A-Z routing table for termination. The 2 platforms for this application I am considering are Asterisk and Freeswitch. Asterisk has the advantage that I'm familiar with its structure and configuration, therefore I can modify it and make it work quickly. The downside is it is clumsy for wholesale switching and does not scale well. Freeswitch seems to be a better bet given that I would like to scale this application to several thousand simultaneous calls, however I am under a time crunch to get a test application up and running. Can someone suggest to me if Freeswitch would be a good platform for this? If so, where would be the best place to plug in my custom translation routine? Can Freeswitch route signalling only, e.g. RTP/media goes directly from gateway to gateway without passing my network? If so, how can that be configured? I apologize for asking without completely studying all the available documentation and code, but as I said I am under pressure and I was hoping some benevolent individual would point me in the right direction to expedite this. Thank you for any assistance. Best Regards, Chris Ziomkowski

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