- FreeSWITCH-users - FreeSWITCH Mailing Lists

Global limit on the number of conference legs
by Andy Spitzer 08 Jan '09

08 Jan '09

Woof! Does FS have a way of limiting the total number of conference legs on a box? I am aware that each individual conference profile can have a "max-members" param, but what I'm looking for would span multiple conferences, with a maximum leg limit per server, regardless of the per conference limit. Something like switch.conf "max-sessions", but only for conference legs, with an appropriate message played to the caller if it is exceeded. --Woof

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bootstrap error
by jocke eriksson 07 Jan '09

07 Jan '09

Happy New Year to you all ! I'm doing an install and ran into some trouble, any ide why ? bootstrap: libtool version 2.2.4 found. You need libtool version 1.5.14 or newer installed to build FreeSWITCH from SVN. Regards Jocke

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Problems with ODBC
by fidibus83 07 Jan '09

07 Jan '09

Hello, I need your help again! I want to use ODBC in javascripts. First I installed unixodbc and I did all the steps what are mentioned in wiki: mod spidermonkey odbc. But when I test my ODBC Setup by running isql I get this error: [ISQL]ERROR: Could not SQLConnect And I cant start FS. I get this Error: ./freeswitch: error while loading shared libraries: libodbc.so.1: cannot open shared object file: No such file or directory I uncomment: conf/autoload_configs/spidermonkey.conf.xml <load module="mod_spidermonkey_odbc"/> build/modules.conf languages/mod_spidermonkey_odbc Here my configurations: odbc.ini [fs01_odbc] Driver=MySQL SERVER=localhost PORT=3306 DATABASE=freeswitch Socket= /var/run/mysqld/mysqld.sock odbcinst.ini [MySQL] Description = ODBC for MySQL Driver = /usr/local/lib/libodbc.so UsageCount = 1 Can you help me? What do I wrong? Thanks! Regards, fidibus

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Need some info
by Aadil M 07 Jan '09

07 Jan '09

Hi All, I am a new comer to FreeSwitch. The thing is I am planning to setup a call center along with a few of my friends. Hence I was looking around for some IP PBX and so stumbled upon FreeSwitch. This call center would be a 10 to 20 person setup. Here are a few of the questions that I have? 1. Can I have just on PSTN line and configure multiple extensions on FS by using relevent hardware?If not then how many PSTN lines would I require? 2. I did Google for the hardware, it seems I would require some FXO/FXS cards. Am I right? Where can I get them? I am sorry if you find these question too novice or lame. Kindly forgive me. Looking for a positive reply from the community. Regards, Aadil

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close channels properly
by Carole O. 07 Jan '09

07 Jan '09

Hello, I use the following code to call VoIP speakers and make an announcement: <extension name="security_announcement"> <condition field="destination_number" expression="^9001$"> <action application="set" data="conference_auto_outcall_caller_id_name=security_announcement"/> <action application="set" data="conference_auto_outcall_id_number=9001"/> <action application="set" data="conference_auto_outcall_timeout=60"/> <action application="set" data="conference_auto_outcall_flags=mute"/> <action application="set" data="conference_auto_outcall_announce=announcement/announcement_security_delay.wav"/> <action application="conference_set_auto_outcall" data="user/1021@$${domain}"/> <action application="conference_set_auto_outcall" data="user/1022@$${domain}"/> <action application="conference" data="conf_speaker@profilespeaker_announcement"/> </condition> </extension> 1021 and 1022 are the speakers. At the end of the announcement, since there is no noise anymore, the speakers stop listening but they do not send any messages to tell Freeswitch it can close the opened channels. The channels are closed only after a timeout of 5 minutes. Does anybody know how I could force freeswitch to close all the channels after the announcement? I have seen there is the application sched_hangup but when I used it it only closes the channel to the caller and not the other ones. Thanks a lot for your help and Happy Holidays! Carole -- View this message in context: http://www.nabble.com/close-channels-properly-tp21127913p21127913.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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vmd
by Kristian Kielhofner 06 Jan '09

06 Jan '09

Hello everyone, I know 1.0.2 if fresh (yeah!) but what exactly is the status of vmd? The wiki page says it can't be used as an app yet it shows up in "show application" with start/stop. How might this be used from the dialplan? Thanks! -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com

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New FreeSWITCH release
by Michael Collins 06 Jan '09

06 Jan '09

FYI, If you haven't already heard, we've now released v1.0.2 of FreeSWITCH! Please digg the new release story: http://digg.com/software/FreeSWITCH_New_Release_For_The_New_Year The source can be downloaded here: http://files.freeswitch.org/ in both tar.gz and tar.bz2 formats A Windows MSI file can be downloaded here as well: http://files.freeswitch.org/freeswitch-1.0.2.msi Thanks for your support and keep on FreeSWITCHing! -MC (mercutioviz)

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BLF on Aastra 9133i
by Mark Greene 06 Jan '09

06 Jan '09

Hey guys, I have a user who is in charge of checking multiple voicemail boxes. They are registered under extension 0650, which will have voicemails as well. So the user has a total of 3 VM boxes to check. The MWI light works great for thier extension (0650), but I can't use it for the other voicemail boxes that the user is in charge of checking without asking the user to call and check each mailbox every time the MWI light comes on. The Aastra 9133i has 7 programable keys. One of the many options for programming the keys is to assign keys to be BLF keys. So I think a good solution is to label two of the programable keys VM Box 2 and VM Box 3 and set them to be BLF keys. Then when a VM is left on one of those boxes I can just have an action that turns the BLF on for the correct key. Can this be done with FS. How could I go about doing it? Thanks in advance, - Mark

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sofia deflect issue
by jonathan augenstine 06 Jan '09

06 Jan '09

I have a lua script that plays a prompt and then collects digits from an inbound call. After making a database query the call is either dropped or transferred to another server. The way I transfer the call is via the 'deflect' command. I execute that command as follows: session:execute("deflect", "user@target_server"); When I do a tcpdump, everything seems to progress as expected. I see the correct SIP messages exchanged, but after the last 200 OK is transferred Freeswitch starts sending BYE messages to the client and the client hangs up with an "Abnormal call termination". According to RFC 3515 there are no BYE messages in the protocol exchange. I have set auto hangup to false. Any ideas on what I might be doing wrong? Jonathan

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[gtalk to sip] Ringing tone
by kriko 06 Jan '09

06 Jan '09

When placing a call between gtalk and sip, one end first picks up and then ringing starts on the other side. In that time user which answers doesn't hear any tone that could indicate a call in progress and it seems a bit odd. Is it possible to play a track in the meantime, so the caller gets a feeling that something is going on? -- kriko

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