January 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Investigating one-way sound?
by Fred-145 28 Jan '10

28 Jan '10

Hello With both a PC+XLite and a Siemens A580IP phone on the same LAN, sound is OK both ways when I call from the XLite application, but when calling from the Siemens, I can't hear sound coming from the XLite application. FWIW, both phones are configured to use G711a and G711u, in this order. To investigate, what kind of error message should I pay attention to in all the messages that scroll through the console? Thank you. -- View this message in context: http://old.nabble.com/Investigating-one-way-sound--tp27341219p27341219.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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compilation error on fs_cli (Windows) (latest version)
by Diego Toro 28 Jan '10

28 Jan '10

Hi, I have compilation error "error C2220" on fs_cli project on Windows using VS2008. FS: latest version (2010/01/26) VS: VS2008 SO: Windows 7 VS2008 Error log: Error 1 error C2220: warning treated as error - no 'object' file generated g:\ftp\incoming\fs\freeswitch-1.0.5-20100126-0400\libs\esl\fs_cli.c 106 fs_cli Warning 2 warning C6385: Invalid data: accessing 'global_profile->console_fnkeys', the readable size is '48' bytes, but '-4' bytes might be read: Lines: 86, 88, 90 g:\ftp\incoming\fs\freeswitch-1.0.5-20100126-0400\libs\esl\fs_cli.c 90 fs_cli Warning 3 warning C6246: Local declaration of 'p' hides declaration of the same name in outer scope. For additional information, see previous declaration at line '844' of 'g:\ftp\incoming\fs\freeswitch-1.0.5-20100126-0400\libs\esl\fs_cli.c': Lines: 844 g:\ftp\incoming\fs\freeswitch-1.0.5-20100126-0400\libs\esl\fs_cli.c 895 fs_cli Warning 4 warning C6011: Dereferencing NULL pointer 'cursor': Lines: 839, 840, 841, 842, 843, 844, 846, 849, 853, 857, 868, 870, 871, 884 g:\ftp\incoming\fs\freeswitch-1.0.5-20100126-0400\libs\esl\fs_cli.c 884 fs_cli Thank you Diego Toro http://lacarretade.blogspot.com/

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Question about Lua script: How do I detect when someone picks up the phone?
by Mike van Lammeren 27 Jan '10

27 Jan '10

Hello! Some day, I'll probably laugh at asking this question, but today I can't figure it out. I've written a Lua script that listens for a call, then dials a phone number to a second person. It plays a message, then prompts the second person to hit pound to connect. If the second person hits pound, then it bridges the two calls together. All that works great, but I can't figure out how to get the session for the second person to wait until that person answers. I'm using FreeSWITCH 1.0.4, and although there is a *getState* function documented in the wiki, it doesn't seem to exist for me. Any help would be appreciated! Mike van Lammeren

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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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