September 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

any VOIP service provider support the freeswitch now?
by Dave 11 Sep '08

11 Sep '08

I like to try out the freeswitch in real life call, but besides the connection through the asterisk, it seems that it has no way to directly support by VOIP service provider. I am sure some of the VOIP service provider do see the potential business in this area, but not sure why we still didn't see the crowded. Can someone share with me the list of the VOIP service provider who can support the freeswitch connection, or any good method to connect to VOIP service throguh the SIP. Thanks. David

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Mono for A-leg or B-leg only
by Евгений Золотов 11 Sep '08

11 Sep '08

Can we register with help of "record_session" ONLY A-leg or B-leg with record to *.wav in MONO? Thanks, Evgeniy.

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Can't set up users outside of local domain...
by Kin Quek 11 Sep '08

11 Sep '08

Hi All, I am new to FreeSwitch so please bear with me if my question is too obvious. I set up FS in Boston and it is running fine. I want to set up a user in LA with username 1003 for example. I edited default.xml as below: <include>  <domain name="$${domain}" value="12.345.67.89"> <params> <param name="dial-string" value="{presence_id=${dialed_user}@${dialed_domain},transfer_fallback_extension=${dialed_user}}${sofia_contact(${dialed_domain}/${dialed_user}@${dialed_domain})}"/> </params> <variables> <variable name="record_stereo" value="true"/> </variables> <X-PRE-PROCESS cmd="include" data="default/*.xml"/> </domain> </include> ----------- I set up directory.xml as such: <domain name="$${sip_profile}"> <user id="1003"> <params> <param name="password" value="1234"/> </params> <variables> <variable name="sip-force-contact" value="sip:1003@12.345.67.89"/> </variables> </user> </domain> ---------------- My user in LA tried to register as user 1003 using X-Lite: Username: 1003 Password: 1234 Auth username: 1003 Domain: 12.345.67.89 ------------- The remote user won't be able to register with FS. The warning messages are attached: [Warning] sofia_reg.c: 1251 sofia_reg_parse_auth() can't find user [1003(a)12.345.67.89] You must define a domain called '12.345.67.89' in your directory and add a user with the id="1003" attribute and you must configure your device to use the proper domain in it's authentication on credentials. ------------- I tried different attempts including hard code the domain name value in default.xml and the error message remained the same. I know this is possible because my user at LA set up something similar with Asterisk and we have used it many times. I think it should be possible on FS as well. Any help well be highly appreciated! Thanks in advance. Best regards, Kin Quek

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transfer an extablished call
by Gayatri Kulkarni 11 Sep '08

11 Sep '08

Hey guys, sorry if i am bothering, but please answer this query of mine: i originate a call using this: originate sofia/internal/1000 &bridge(sofia/internal/1001) Now, i want to transfer this call which is already established to sofia/internal/1002 what should i do? simply "&transfer(sofia/internal/1002)" or would i require to give call-id / uid or something? if yes - how would i find that out? Regards, Gayatri Kulkarni

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Dialplan actions
by Jonas Gauffin 11 Sep '08

11 Sep '08

Hello Is it possible to do something like this (pseudo code): set call_timeout=20 bridge somehere if timeout bridge somewhereElse else javascript unexpectedFailure.js that is invoke one action if the bridge fails due to call timeout, and to do something else for every other reason. //Jonas

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Possible to use fresh channelvariables in conditions?
by Jon Bruel 11 Sep '08

11 Sep '08

I have tried this: <condition field="${SHOWANUMBER}" expression="^(2015[0-9][0-9][0-9][0-9])$"> <action application="eval" data="ACTION=${SHOWANUMBER}"/> <anti-action application="eval" data="ANTI_ACTION=${SHOWANUMBER}"/> </condition> It runs through the conditions, but the channel variable SHOWANUMBER is not initiated at that time. When it comes to the code above, it chooses the anti-action. It looks as if the condition-mechanism cannot be used if the channel variable is not defined when the system goes through the conditions in the first run? Are there any alternative ways - except for using a script language? Thanks, Jon

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Passing variable argument in a dial plan
by Adeel Ansari 11 Sep '08

11 Sep '08

Hi all, I am hitting a dial plan using a command. And the application defined in the dial plan is running successfully with the argument passed. Below is the illustration. *Command* originate sofia/gateway/gizmo1/848458448557 121 *Dial Plan* <extension name="java_test" continue="true"> <condition field="destination_number" expression="121"> <action application="java" data="/home/adeel/programs/freeswitch/scripts/freeswitch.jar net.digitalapps.freeswitch.CallTest abcdefgh"/> </condition> </extension> My Java program successfully showing me the argument "abcdefgh". Now I want to pass a variable, not a fixed like "abcdefgh", as an argument. How can I do it. Thanks. -- Best, Adeel Ansari http://www.linkedin.com/in/adeelansari

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Tracemonkey engine
by Ruchir Brahmbhatt 11 Sep '08

11 Sep '08

Hi All, FreeSwitch uses SpiderMonkey javascript engine which is very good. Now how about using TraceMonkey engine which is next version offering very high performance. Its fastest javascript engine according to this post: http://developer.mozilla.org/devnews/index.php/2008/09/09/aboutmozilla-fire… Thanks, Ruchir Brahmbhatt Director Ecosmob Technologies Pvt. Ltd.

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FS and DIDs
by Jair Santos 11 Sep '08

11 Sep '08

Hi all, the wiki provides information about how to receive an inbound call http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Receiving_an_inbound_c all_from_a_Gateway The problem is that I can have only one DID per gateway. I would like to use one gateway that will allow several FS extensions (Virtual Numbers) to make calls, and point several different DIDs to these extensions individually. Is that possible ? thanks Jair Santos

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mod_python post-processing after hangup
by Novak Joe 11 Sep '08

11 Sep '08

Hi, I'd like to run some process after a hangup, in the background. I'm using mod_python and would like to run something after a call ends. In the snippet below I'd like the call to hangup at 'session.hangup(1)' then the consoleLog to print the 'Testing' message 15 seconds later. At present, if I hangup my call, it works, however if I stay on the line the call doesn't hangup until the call to sleep comes back. How can accomplish this functionality without using mod_event_socket, is there a way? Should I stick this in the hangup_hook? Will the hangup_hook maintain state for a database call? Cheers, Joe ----------------------------- ----------------------------- import os from freeswitch import * import time def hangup_hook(session, what): consoleLog("info","hangup hook for %s!!\n\n" % what) return def input_callback(session, what, obj): if (what == "dtmf"): consoleLog("info", what + " " + obj.digit + "\n") else: consoleLog("info", what + " " + obj.serialize() + "\n") return "pause" def handler(session, args): session.answer() session.setHangupHook(hangup_hook) session.setInputCallback(input_callback) session.streamFile(session.getVariable("hold_music")) session.hangup(1) time.sleep(15) consoleLog("info", "Testing.") ----------------------------- -----------------------------

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