- FreeSWITCH-users - FreeSWITCH Mailing Lists

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>

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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

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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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Problem with rtp streams and stair-stepped (or ramped) increased lag
by Nathan Neulinger 10 Jul '13

10 Jul '13

Having some occasional user reports of problem calls where they are fine, and then something happens on the call and lag increases significantly, sometimes up to 2+ seconds. I've got severe examples on both cisco and polycom phones. What I'm seeing looks like the call is going perfectly for a period of time (sometimes up to a few minutes), and then there is a period of time with higher latency packets, and the overall delay in the graph increases steadily at that point, and never recovers. It sounded like rtp-autoflush-during-bridge might be something relevant, but that's set to the default. Also saw "rtp-rewrite-timestamps", but unclear if that is necessary here. I've got pcap captures of the rtp streams if that'll help any. -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

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Jira and Developers requesting more information
by Ken Rice 10 Jul '13

10 Jul '13

Greeting FreeSWITCHers! So, I hate to do this, but we¹ve got the perfect example here of a ticket in Jira... http://jira.freeswitch.org/browse/FS-4918 Last December, the reporter gives us a fairly good bug report with most of the good information, but also Anthony was able to deduce a possible problem, and present a patch. Several days later the reporter comes back that the patch solves a segfault issue, but the developer requested more information as he added some debugging into the patch to issolate the root cause... This tells me this patch probably fixes the issue, but theres a possibility that the problem may still be lurking int he shadows for later... Now here we are 5 months later and the reporter hasn¹t responded with the requested info and we are left with nothing to do but close the ticket knowing he hasn¹t poped back up complaining about the segfault and nothing more. Now, you may be asking whats the problem here, the problem has been fixed? The problem comes from the fact as part of his commitment to supporting FreeSWITCH, Anthony (the developer) committed a fair bit of his time to track down and fix the issue for which he asked in trade nothing but more information... Which is well with in the spirit of Open Source Software... But the reporter kinda left us hanging here. He got a developer to work on (and probably fix) his problem but doesn¹t pay the developer back for his work with the needed requested... And now we have another open stale bug waiting around on the reporter who will probably never submit the info... So please if the developer ask for more info give it to them, that¹s what are you are expected to do in exchange for their time at the very least. TL;DR version... If developers ask for more info, don¹t be a tool, give them the requested info... -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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Re: [Freeswitch-users] FreeSWITCH Start up Script - Ubuntu Server
by sreekanth 10 Jul '13

10 Jul '13

just add "/usr/local/freeswitch/bin/freeswitch -nc" to /etc/rc/local before exit 0 My rc.local file looks like below #!/bin/sh -e # # rc.local # # This script is executed at the end of each multiuser runlevel. # Make sure that the script will "exit 0" on success or any other # value on error. # # In order to enable or disable this script just change the execution # bits. # # By default this script does nothing. /usr/local/freeswitch/bin/freeswitch -nc exit 0 ------------ cheers ! http://askadmin.com http://sipcosystems.com -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-Start-up-Script-Ub… Sent from the freeswitch-users mailing list archive at Nabble.com.

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mod_callcenter and start_dtmf
by Mimiko 10 Jul '13

10 Jul '13

Hello. Recently I have found problem which was in callcenter application. Agents randomly had heard a beep sound like calling members where pushing some number to send a dtmf done. This was annoying, because this was in a middle of conversation. This tone had appearing randomly not on every incoming call to callcenter. It was high volume of 0.5-1 sec long. I couldn't track the source of this sound, as in the resulting recorded file of the conversation there where no suspicious sounds, nor a beep from callers, nor the beep from or to agents. So I was lost. Several days ago I remembered about using start_dtmf application on incoming calls from phone provider because the do not use rfc standard for dtmf. More strange thing was that this strange beep sound didn't appear on regular incoming call to some extension. It was only when external callers from phone provider were calling into callcenter number. So I used stop_dtmf just before sending incoming call to callcenter application. With this command those strange beeps disappeared. Then I've changed start_dtmf to spandsp_start_dtmf and didn't use spandsp_stop_dtmf. Beeps does not appear too. So there seems to be a problem with core handling of in-band dtmf with start_dtmf aplication when call is routed into callcenter, and the callcenter bridged the call to the agent. Should I rise an issue to the jira, or this is an internal known issue of core dtmf handling and will not be resolved? Mimiko desu.

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