July 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

fs_cli and button TAB
by Fernando Villarroel 11 Jul '13

11 Jul '13

Dear All. The commands running well but my problem is the follow: When i press TAB on fs_cli console this not show list of command or when i try to do reloadxml fs_cli not autocomplete when i press the buttonTAB? Any idea? Regards. Enviado desde mi iPhone

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using the originate command
by Christopher Hall 11 Jul '13

11 Jul '13

I have written code to interact with FreeSWITCH using the event socket interface. I am originating calls from my code but my understanding of what is required is very limited. I would like to be able to originate a call and have the call I originate processed through the dial plan and I don't appear to be able to make this happen. An example of my problem: Using a default configuration on FreeSWITCH: If I originate a call as below with a SIP phone registered on extension 1000, my sockets server is called as expected. api originate {originate_timeout=30}sofia/internal/1000%192.168.1.5 &socket(127.0.0.1:10000 async full) if I then modify the configuration of my server such that the public dial plan contains the following <include> <extension name="Test"> <condition field="destination_number" expression="^10000$"> <action application="set" data="domain_name=$${domain}"/> <action application="socket" data="127.0.0.1:10000 async full"/> </condition> </extension> </include> And call 10000 from the SIP phone registered on 1000, again my sockets server is called. If however, I attempt to originate a call to 10000 from my code using api originate {originate_timeout=30}sofia/internal/10000%192.168.XX.XXX &socket(127.0.0.1:10000 async full) I get the response. -ERR USER_NOT_REGISTERED Is there a way in which I can originate a call from my code and have it processed in the same way as a call originated from a SIP phone? I would like to be able to do this so that my code can simply originate calls to numbers in the knowledge that FreeSWITCH will then route the call accordingly and that doesn't appear to be the case at the moment as the originate does not appear to be processed by the dial plan. Thanks Chris.

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FreeTDM originate status
by Ashish gautam 11 Jul '13

11 Jul '13

I am generating outgoing calls to PSTN network through event socket originate action. I want to get the status that the call was generated successfully or not i.e. if the call was failed, what was the reason for that (Network failure, user busy, out of reach etc). How can I get that. Please help. -- Ashish

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ClueCon room sharing
by João Mesquita 11 Jul '13

11 Jul '13

Folks, Anyone willing to share a room for ClueCon? I get there on the 5th and leave on the 9th. João Mesquita

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Execute after FIFO when success call is bridged
by Antonio Silva 11 Jul '13

11 Jul '13

HI, I'm trying to get some information after calling a FIFO, when the FIFO exits with TIMEOUT or the member hangups the call is ok, i get the info(). But when the fifo is connected with the agent the info() is not executed... Is it possible to do what i want? My configuration is the following: <extension name="200"> <condition field="destination_number" expression="^200$"> <action application="set" data="continue_on_fail=true"/> <action application="set" data="hangup_after_bridge=false"/> <action application="answer"/> <action application="set" data="fifo_music=$${hold_music}"/> <action application="set" data="fifo_orbit_exten=_continue_:10"/> <action application="fifo" data="200(a)marces.com in"/> <action application="info"/> </condition> </extension> the fifo conf: <fifo name="200(a)marces.com" importance="0"> <member timeout="60" simo="1" >{fifo_member_wait=nowait}user/200(a)marces.com</member> </fifo>

1 0

PHP + FREESwitch
by Wesley Akio 11 Jul '13

11 Jul '13

Hi all, I've been playing with FS for a while, mostly integrating with Java and PHP (and some Javascript to a lesser extent). I have few lines of code sitting in my computer and would like to know how many of you actually use PHP with FS to see if it would be worth reviewing and publishing some of it... Best, Wesley Akio TuntsCorp.com

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Odp: Re: Odp: Regex Help
by Mariusz Czulada 11 Jul '13

11 Jul '13

"|" is expression is treated as a parameter separator to "regex" api command. Try instead: freeswitch@internal> regex m:/1234567890*123456*12*/^((\d{10})\*(\d{6})\*(1|3|12)\*)$/$1 $2 $3 $4 1234567890*123456*12* 1234567890 123456 12

2 1

PHP ESL & Event Socket Outbound
by Milos Jovanovic 11 Jul '13

11 Jul '13

Hello, I am trying to create PHP application that will be receiving information about each call from freeswitch and send commands to control freeswitch. I found an example at https://wiki.freeswitch.org/wiki/PHP_ESL#ivrd BUT unfortunately they do not explain how I can retrieve information from freeswitch (so I can do the routing in PHP). I guess I should use PHP ESL but the documentation is pretty bad so if someone has some working example, I'd appreciate it. I tried to read from stdin this way: *#!/usr/bin/php -q <?php // set a couple of things so we dont kill the system ob_implicit_flush(true); set_time_limit(30); // Open stdin so we can read the data in $in = fopen("php://stdin", "r"); // Connect echo "connect\n\n"; // READ DATA $text = fgets($in); file_put_contents('/usr/local/freeswitch/scripts/socketlog.txt' ,$text, FILE_APPEND); // Answer echo "sendmsg\n"; echo "call-command: execute\n"; echo "execute-app-name: answer\n\n"; // Play a prompt echo "sendmsg\n"; echo "call-command: execute\n"; echo "execute-app-name: playback\n"; echo "execute-app-arg: /usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav\n\n"; // Wait sleep(5); // Hangup echo "sendmsg\n"; echo "call-command: hangup\n\n"; fclose($in); ?>* *Unfortunately, everything I get as a result in my log file is:* *Event-Name: CHANNEL_DATA*

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How to calculate server requirements for Freeswitch.
by Mahendra Bhegade 11 Jul '13

11 Jul '13

Hi All, I plan to use Freeswitch and would have 20,000 conference calls with about 3-4 person in each of the conference. Some of them would record their conference calls, How do I go about calculating the server resource needs. What are all the factors that I should be looking at, Mahendra Bhegade

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mod_spandsp status - stability and compatibility?
by Kirk Wolf 10 Jul '13

10 Jul '13

According to the Wiki page: "We are finishing mod_spandsp. It requires full field testing now." Our requirements: (currently running Version 1.2.10 git 46825dd on Ubuntu 12.10 ) - receive faxes via SIP trunk (a dedicated flowroute inbound DID) - email to fixed address upon receipt I would like to ask for experience of other users as to whether the current state of the module is such that we can expect reasonably good compatibility and stability. Thanks, Kirk Wolf Dovetailed Technologies http://dovetail.com

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