April 2010 - freeswitch-dev@lists.freeswitch.org

[Freeswitch-dev] ticketing system
by babak yakhchali 14 Apr '10

14 Apr '10

Hi I need to implement a system in which customers call support operators and if there is no idle operator they are placed in a queue and all conversations are recorded and assigned a number so later customers can hear their conversations. Can anyone help me in choosing right modules to use? I have to add I am using sip to rcv calls and currently I'm using .Net to do this

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[Freeswitch-dev] ClueCon MMX - Update
by Michael Collins 13 Apr '10

13 Apr '10

Greetings, We would like to let everyone know that ClueCon MMX preparations are moving forward. New sponsors have signed up and we are starting to fill up our speaker slots. Stay tuned for more information. In the meantime, if you have not already signed up for ClueCon then please call 877.742.CLUE (2583) and get registered right away! Looking forward to seeing you in Chicago, The ClueCon Team 877.742.CLUE http://www.cluecon.com

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[Freeswitch-dev] mail queue
by Brian West 13 Apr '10

13 Apr '10

I dumped the mail queue this morning in hopes it would clear up anyone getting duplicates. The log file reached 2 gigs and the MTA decided to just exit since the OS wouldn't let it open the logfile for append over the weekend. Which caused this problem in the first place. Sorry for any trouble. Thanks, Brian

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[Freeswitch-dev] G.729 Info On Wiki
by Michael Collins 12 Apr '10

12 Apr '10

Hi folks, Just a quick FYI, there is g.729 license info now on the wiki: http://wiki.freeswitch.org/wiki/Mod_com_g729 Thanks to Steve Ayre for starting this page. If you have g.729 knowledge to add to the page please do so. Also, the INSTALL.txt file has been updated with a bit more detail: http://files.freeswitch.org/g729/INSTALL.txt Happy transcoding! -Michael

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[Freeswitch-dev] mod_conference: no events fire if member energy level is zero
by Fowler, Peter (Peter) 12 Apr '10

12 Apr '10

If the conference participant sets his/her energy level threshold to zero (or the conference energy-level is set to zero), then The floor_change event will never fire if the particpant becomes the loudest speaker. This is due to the line If ((switch_test_flag(member, MFLAG_CAN_SPEAK) || switch _test_flag(member, MFLAG_MUTE_DETECT)) && energy _level) { ^^^^^^^^^^^^ I folks agree I'll enter a Jira .. Peter

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[Freeswitch-dev] Trying to send SIP MESSAGE to device
by Paulo Emerique 10 Apr '10

10 Apr '10

Hi all, this is my first day of hands on experience with FreeSwitch, which I have been following for many months now. We deploy Asterisk boxes, and now that Freepbx runs with Freeswitch, we intend to "migrate" to it as our main platform. I'm really enjoying FS, congratulations! Now for the actual question: The issue is I bought a GSM Gateway, wich I'm trying to send SMS thru it ... and the device documentation says to send a "SIP MESSAGE" to it like: MESSAGE sip: 1003@192.168.1.126:5060;transport=udp SIP/2.0 From: <SIP 1000(a)192.168.1.100>;tag=5031 To:<SIP: 1003(a)192.168.1.126> Call-ID: 808807EB-A8B3-DD11-BBA6-005056C00008(a)192.168.1.100 CSeq: 3 MESSAGE Contact: <sip: 1000(a)192.168.1.100> Max-Forwards: 16 Date: Tue, 08 Apr 2010 06:36:37 GMT User-Agent: Paulo p-hint: usrloc applied Content-Type: text/plain Content-Length: 17 96913511 hi paulo GSM Gateway is at IP 192.168.1.126 ... so I have tried to acomplish it by 2 ways: 1) using mod event sockets: I telnet to server and after auth, issue: sendevent SEND_MESSAGE profile: sipinterface_1 content-length: 18 content-type: text/plain user: 1003 host: 192.168.1.126 96913511 Hello Baby and returns: Content-Type: command/reply Reply-Text: +OK accepted I'm watching SIP traffic (using "sofia profile sipinterface_1 siptrace on") and I see no SIP packet going. then I tried 2) chat sip|1030@192.168.1.132|1003@192.168.1.126|Test message which returns: Error! Message Not Sent 2010-04-10 10:46:34.555510 [ERR] sofia_presence.c:109 Chat proto [dp] from [1030(a)192.168.1.132] to [1003(a)192.168.1.126] Test message Invalid Profile 192.168.1.126 So I tought: well, need to configura a profile (wich I understand is created on freepbx "SIP Interfaces" ... so I did, but FS console shows: 2010-04-10 10:25:59.681316 [ERR] sofia.c:1347 Error Creating SIP UA for profile: sipinterface_3 To collect more info, I tried to make similar steps to send SIP MESSAGE to a Granstream HT-486, so I created a "SIP Interface" to it, but returns same error msg: 2010-04-10 10:25:59.868661 [ERR] sofia.c:1347 Error Creating SIP UA for profile: sipinterface_4 Thank for your help, Paulo Cesar Emerique

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[Freeswitch-dev] UPDATE: G.729 Codec Licensing *IS* Available
by Michael Collins 08 Apr '10

08 Apr '10

Greetings all, The FreeSWITCH team would like to let everyone know that we do indeed sell g.729 licenses for $10 each. Use this link to initiate a purchase: http://www.freeswitch.org/node/235 Note: licenses are available only for Linux-based systems at this time. Please stay tuned for updates. The INSTALL.txt file has very detailed instructions: http://files.freeswitch.org/g729/INSTALL.txt Keep in mind that a single license includes one encoder and one decoder, that is, it can transcode both directions of a single phone call. If you have any other questions please email us here or join us in #freeswitch on irc.freenode.net. -Michael

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[Freeswitch-dev] Another IRC day in #freeswitch!
by Michael Collins 08 Apr '10

08 Apr '10

Hey everyone, come join us in #freeswitch on irc.freenode.net! We are going for 250 users. (No sockpuppets, please.) If you don't have an IRC client then visit http://www.freeswitch.org and click the Join Us On IRC link to use the Web IRC applet. See you in channel! -MC (IRC: mercutioviz)

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[Freeswitch-dev] streaming with mod_shout
by vip killa 08 Apr '10

08 Apr '10

I'm trying to figure out how to stream to a shoutcast server. There is no examples in the WIKI I've been able to stream to an icecast server using the command "conference name record shout://username:password@host/stream.mp3" my shoutcast server does not require a username, so what would the command be for an actual shoutcast server?

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[Freeswitch-dev] FreeSWITCH Conference Call - Moises Silva From Sangoma Talks FreeTDM!
by Michael Collins 07 Apr '10

07 Apr '10

Hello everyone! Moises (moy) is going to be speaking today about FreeTDM. Please join the conference call at noon Eastern, 9am Pacific time: http://wiki.freeswitch.org/wiki/FS_weekly_2010_04_07 Moises will start his presentation at about 12:15/9:15. Be sure to bring your questions about TDM and Sangoma! -Michael

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