June 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Cron-like execution in FS
by Matthew Fong 10 Feb '21

10 Feb '21

I'm wondering if there's any features that allow the cron-like execution of code inside of Freeswitch, preferably with lua--or if I am stuck using the api interface and running the cron outside of freeswitch. --matt

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answer command
by Maxim Tsvetov 21 Sep '20

21 Sep '20

Hello, I found in WIKI list of supported commands http://wiki.freeswitch.org/wiki/Mod_commands. I need command "answer" but it doesn't exist (Freeswitch 1.04, Win32) That's what Freeswitch replies for this command: "answer: Command not found!" Сould you please help? Regards, Maxim Tsvetov

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mod call center tier behaviour
by abubacker 04 May '20

04 May '20

I am working in mod_callcenter Before my call center configuration is like this /referene : callcenter.conf.xml/ /<agents> <agent name="1000@default" type="callback" contact="[call_timeout=10]user/1000" status="Available" max-no-answer="0" wrap-up-time="0" reject-delay-time="1" busy-delay-time="1" /> <agent name="1001@default" type="callback" contact="[call_timeout=10]user/1001" status="Available" max-no-answer="0" wrap-up-time="0" reject-delay-time="1" busy-delay-time="1" /> <agent name="1003@default" type="callback" contact="[call_timeout=10]user/1006" status="Available" max-no-answer="0" wrap-up-time="0" reject-delay-time="1" busy-delay-time="1" /> </agents> <tiers>  <tier agent="1000@default" queue="support@default" level="1" position="1"/> <tier agent="1001@default" queue="support@default" level="1" position="2"/> <tier agent="1003@default" queue="support@default" level="1" position="3"/>  <tier agent="1003@default" queue="secondary" level="1" position="3"/> <tier agent="1001@default" queue="secondary" level="1" position="2"/> <tier agent="1000@default" queue="secondary" level="1" position="1"/> </tiers> </configuration> / Tier list : /freeswitch@FMS-Queue> callcenter_config tier list support@default queue|agent|state|level|position support@default|1000@default|Ready|1|1 support@default|1001@default|Ready|1|2 support@default|1003@default|Ready|1|3 +OK/ When I renamed the 1003@default as 1006@default / <agents> <agent name="1000@default" type="callback" contact="[call_timeout=10]user/1000" status="Available" max-no-answer="0" wrap-up-time="0" reject-delay-time="1" busy-delay-time="1" /> <agent name="1001@default" type="callback" contact="[call_timeout=10]user/1001" status="Available" max-no-answer="0" wrap-up-time="0" reject-delay-time="1" busy-delay-time="1" /> <agent name="1006@default" type="callback" contact="[call_timeout=10]user/1006" status="Available" max-no-answer="0" wrap-up-time="0" reject-delay-time="1" busy-delay-time="1" /> </agents> <tiers>  <tier agent="1000@default" queue="support@default" level="1" position="1"/> <tier agent="1001@default" queue="support@default" level="1" position="2"/> <tier agent="1006@default" queue="support@default" level="1" position="3"/>  <tier agent="1006@default" queue="secondary" level="1" position="3"/> <tier agent="1001@default" queue="secondary" level="1" position="2"/> <tier agent="1000@default" queue="secondary" level="1" position="1"/> </tiers>/ After that I have reload the mod_callcenter http://pastebin.freeswitch.org/14810 The 1003@default is not overwritten , but 1006@default has added newly why 1003@default is still exist /freeswitch@FMS-Queue> callcenter_config tier list support@default queue|agent|state|level|position support@default|1000@default|Ready|1|1 support@default|1001@default|Ready|1|2 support@default|1003@default|Ready|1|3 support@default|1006@default|Ready|1|3 +OK/ Is it a bug or Is it a default behaviour ? -- Best regards, N.Abubacker , Associate system engineer , bk systems pvt ltd , Ph : 9144-43902701 Disclaimer:http://www.bksystems.co.in/email-policy

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dynamic music on hold for mod_conference
by Lappe, Adam 19 Mar '20

19 Mar '20

Hi all, i am trying to dynamically set the mod-sound variable in mod_conference, so I can play different music on hold to my conference rooms without having to create multiple profiles. Is there a way to set my moh-sound path while I create the conference? Or can I change the moh-sound file via the event socket in a existing conference? I think this would also lead to success. Thanks in advance, Adam

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Re: [Freeswitch-users] stream a file multicast with mod_esf
by Sluschny, Thomas 10 May '18

10 May '18

Hi Brian, i thought if i can stream from portaudio it is almost the same with streaming from file, so it should working already now. Is this not the design idea of channels and media to do so? regards, thomas PS: sry for improper formatted mail, i cant reply at the moment and have to copy mail from archive :( ________________________________ >From brian at freeswitch.org Thu Feb 12 07:07:36 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 12 Feb 2009 09:07:36 -0600 Subject: [Freeswitch-users] stream a file multicast with mod_esf Message-ID: <6F68FEDB-8976-44D4-9A67-F871EC7D5120(a)freeswitch.org <mailto:6F68FEDB-8976-44D4-9A67-F871EC7D5120@freeswitch.org> > esf is for multi cast paging... it currently won't let you play files... we would have to create a multicast playback application. /b On Feb 12, 2009, at 8:00 AM, Sluschny, Thomas wrote: > Hi, > > i want to stream a file per IP multicast with mod_esf. > > I can stream IP multicast with: > pa call stream XML > and in XML dialplan: > <extension name="stream"> > <condition field="destination_number" expression="^stream$"> > <action application="answer"/> > <action application="esf_page_group" data="224.168.168.168 34567"/> > <action application="esf_page_group" data=""/> > </condition> > </extension> > and i can also play files with 'playback' app, > > BUT: how can put these 2 things together? > > May be its trivial, but i cant get it make working with 'originate' > or 'uuid_broadcast' or 'bridge', 'transfer' and so on ... > > Thanks in advance, > Thomas

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Voicemail via web interface
by John 24 Apr '18

24 Apr '18

Hello, Can you point me to any additional information about the voice mail via web interface? I have it up and running; however if you click the play button there is no playback, if you click download it will play in MS media player. Thanks John

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mod_conference scalability
by Brian 19 Apr '18

19 Apr '18

Hi, I'm new to FreeSWITCH and I'm testing the scalability of mod_conference to see if it will scale better that other solutions. My scenario is to have one speaker, and many listeners (mute). Since I have only one speaker, I was expecting this to scale well because there is no audio mixing required, just send each frame of the single speaker to each listener. Unfortunately, my testing was disappointing, and it didn't scale nearly as well as I'd hoped (based on what I've read on how FreeSWITCH is supposed to be generally very scalable). Here's my server setup is this: FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon server, 4 Gig of RAM. I've set file logging to "notice" level. My conference profile is configured to suppress several events, hoping that it would improve performance. Here are a few scenarios I tested, and roughly where I reached the point of audio failure on the conferences: Scenario 1: 1 conference, 1 speaker, audio failed at approx 300 listeners (mute) Scenario 2: 4 conferences, 1 speaker per conference, audio failed approx 110 listeners per conference (so just over 400 total channels on the system). Scenario 3: 16 conferences, 1 speaker per conference, audio failed at 32 listeners per conference (so just over 500 total channels on the system). Looking at the output from "top", it seems that in all 3 scenarios, the audio quality failed when the % CPU for the FreeSWITCH process exceeded 300%. I was hoping maybe someone else might have done similar testing, or maybe has suggestions on how to improve the performance. Or perhaps an alternate solution to the one speaker, many listener case? Thanks, Brian.

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Receive chat
by Sameer Khan 05 May '15

05 May '15

Hi, Is there any command which can receive chat from sip channel so I can put that received message in to a variable.

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Transfer hangup but b-leg ring
by FERNANDO VILLARROEL 29 Dec '14

29 Dec '14

Hi. I am trying to use nibblebill and lcr. If originator (example 1006) account have zero balance and trying to call to 1004, the call hangup but the phone 1004 ring anyway. My dialplan: <context name="default"> <extension name="Outbound to PSTN 11 Digits"> <condition field="destination_number" expression="^(10[01][0-9])$"> <action application="set" data="hangup_after_bridge=false"/> <action application="set" data="continue_on_fail=true"/> <action application="lcr" data="$1 pg_prefix2"/> <action application="set" data="nibble_account=${accountcode}"/> <action application="set" data="nibble_rate=${lcr_rate_1}"/> <action application="bridge" data="${lcr_auto_route}"/> </condition> </extension> <extension name="hangup"> <condition field="destination_number" expression="^(hangup)$"> <action application="hangup"/> </condition> </extension> The log of call: http://pastebin.freeswitch.org/19228 How i can do?

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create dummy channel to record conference
by Alexandre Mouillé 07 Oct '14

07 Oct '14

Hi, I hope I'm not posting this in the wrong mailing list. I'm working on a custom conference that uses the conference module. I need a record function that keeps recording (or at list appends the same wav file) when no one is in a given conference. The solution that was used in the Asterisk program I'm porting was to create a dummy channel that goes in the conference and records what it hears in a file. - Can I create a dummy channel with an "originate /loopback" that would go into a conf and record what it hears (I'm not sure what loopback really is) - Otherwise, can I use originate from the FS box to call itself, one side connects to the conference and the other records. (I'm not sure how to do that, and how I could transmit variables (to tell which file to record to) to the called program) Thanks

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