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Polycom buddy watch
by Jeff Lenk 27 Jan '10

27 Jan '10

Hello, Do the Polycom buddy watch presence updates work for "away" "busy" "Dnd" etc. I am running SIP 3.1.3. FS is sending the notifies for when the other phones are in use and that works fine. The Polys are not sending publish events to FS at all for the extended presence states as seen with "sofia profile internal siptrace on". Is this normal? Thanks Jeff _________________________________________________________________ Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. http://clk.atdmt.com/GBL/go/196390709/direct/01/

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mod_spidermonkey memory leak?
by まつばらともや 27 Jan '10

27 Jan '10

I have tested the SVN trunk 15341. When the following scripts are executed, it seems to do the memory leak. Is the mistake found in the following scripts? Or Is there a problem in mod_spidermonkey? -- test script -- session.execute("ring_ready"); session.answer(); session.setVariable("ringback", "%(1000, 2000, 440, 460)"); var bleg = new Session(); var sound_wav = "sounds/test01.wav"; var sound_leg = "both"; var op = "api_on_answer=uuid_broadcast "+session.uuid+" "+sound_wav+" "+sound_leg; var ret = bleg.originate(session, "{"+op+"}" + "sofia/gateway/profile0_gateway1/1000"); if(!session.ready()){ return; } if(!ret){ // bleg not answered. var sound_wav = "sounds/test02.wav"; session.streamFile(sound_wav); if(session.ready()){ session.hangup(); } return; } if(bleg.ready()){ bridge(session, bleg); }

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Call limits (time)
by Nik Middleton 27 Jan '10

27 Jan '10

Hi Guys, I'm looking to migrate my billing platform to use FS. So far so good, however, if a user is low on credit I need to limit the call length. In other words, from the rate card, prior to connecting the call, I know they are going to call a mobile at say $0.10/min, but they only have $2 of credit, so I want to terminate the call after 20 mins, preferably with a message to the originator saying they only have X mins left. Is there a way of achieving this with the originate command? Regards

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FreeSWITCH 1.0.5 and Buying Dinner For The Dev Team
by Michael Collins 27 Jan '10

27 Jan '10

Hello all, The FreeSWITCH development team is planning to meet in one place during the week of February 8 to release version 1.0.5! We would like to invite everyone to show their appreciation by donating a few dollars to help pay for dinner for the development team. More information is available here: http://www.freeswitch.org/node/234 Let's all show the team how much we appreciate them by giving them a well-deserved dinner! -Michael

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mod_fax ECM
by michal kalinowski 27 Jan '10

27 Jan '10

Hello, What mean ECM and for what is used ? I found in my fax.conf parameter: <param name="use-ecm" value="true"/> BR, Michał

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Trouble w/ incoming calls from gateway
by Joseph L. Casale 26 Jan '10

26 Jan '10

Trying to track down what changed and I am sort of baffled. I had an SPA3102's FXO port setup as a UA (1004) as I couldn't figure out how to set it up as "gateway" from fs' perspective. Everything was working fine. Now an incoming call gets dropped with this: Hangup sofia/internal/XXXxxxXXXX(a)192.168.13.1 [CS_NEW] [INCOMPATIBLE_DESTINATION] Its set to dial ext 2000 (a group call) and being that its setup as a UA, 1004, how does this develop the line above? Thanks, jlc

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Conference talk detection
by Dan Lane 26 Jan '10

26 Jan '10

Hi, I have a conference bridge with a simple web interface that shows who is talking using the event api. Unfortunately the users don't like the way the first few ms of speech after silence is cut off. Presumably this is an unavoidable side-effect of cutting out audio when someone isn't speaking. Is there any way to have a conference where all audio is sent to the conference but start/stop talking events are still generated? Currently setting energy-level to 0 disables start/stop talking events :( Cheers, Dan

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Re: [Freeswitch-users] Lua: io.popen/read blocking in other incoming calls
by Dan 26 Jan '10

26 Jan '10

Thanks, I'm using os.execute now and piping the output to a temp file. It's not the most elagant solution, but it works. Call volume is light so I don't think there will be any scalability issues. I'll take a look at the sites you mention and see if I can find something a little bit less hacky. Thanks again. Dan- ----- Original Message ----- From: "Fernando Gregianin Testa" <testa(a)voicetechnology.com.br> To: freeswitch-users(a)lists.freeswitch.org Sent: Sunday, January 24, 2010 12:21:51 PM Subject: Re: [Freeswitch-users] Lua: io.popen/read blocking in other incoming calls You may consider use lua socket.http package as an alternative to popen+wget. Check: https://web.tecgraf.puc-rio.br/luasocket/ http://www.tecgraf.puc-rio.br/~diego/professional/luasocket/http.html Maybe you can be interested also in http://github.com/fertesta/restinlua Em 19/01/2010, às 21:17, Dan escreveu: I would, but I need to post a a wav file that gets recorded, I didn't see a way to supply the location of a file to use as the post data. It looks like you have to url encode the data in the script and pass it all in the call. Thanks Dan- ----- Original Message ----- From: "Rupa Schomaker" < rupa(a)rupa.com > To: "freeswitch-users" < freeswitch-users(a)lists.freeswitch.org > Sent: Tuesday, January 19, 2010 4:06:36 PM Subject: Re: [Freeswitch-users] Lua: io.popen/read blocking in other incoming calls On Tue, Jan 19, 2010 at 3:03 PM, Dan < freeswitch-users(a)digitaldan.com > wrote: My lua script is calling wget through lua's io.popen to send and receive data from a web service. While the f:read to wget is running, other incoming calls will block on the same io.popen call until the first call closes the pipe (with f:close()). You might want to look at the api that mod_curl exposes to do what you want. No need to do an expensive system call just to call a webservice. -- -Rupa _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Fernando Gregianin Testa testa(a)voicetechnology.com.br +55 11 35882166 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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How to get chat message via event
by afshin afzali 26 Jan '10

26 Jan '10

Hi, It seems that the chat messages don't fire via events by default and just exchange between parties. Is it true? Is it possible to enable those via events? appreciate all, -- afshin

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mod_erlang_event: disconnects
by Timur Irmatov 26 Jan '10

26 Jan '10

Hi! We have build a small and simple call center using FreeSWITCH and mod_erlang_event. My erlang process keeps track of available agents and routes incoming calls to them. Calls are sent to my application via: <action application="erlang" data="switch_event:! erlswitch@localhost" /> switch_event is a registered process, which spawns a new process for each incoming call and returns new pid when it receives {get_pid, UUID, Ref, From} message from FreeSWITCH. The problem is, that pretty frequently processes which handle incoming calls receive messages like {'EXIT', <5406.48.0>, noconnection} from FreeSWITCH. As I understand from googling, this happens when remote C node disconnects (and I see TCP connections from FreeSWITCH to epmd daemon being torn down and reestablished). FreeSWITCH drops calls at that moment. Have anyone seen this? Is there any fix/ advice? My system is Debian Lenny (5.0.3), 64-bit system, erlang installed from Debian packages, no backports. -- Timur Irmatov, xmpp:irmatov@jabber.ru

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