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Question about sharing conference between servers
by Tina Martinez 12 Aug '09

12 Aug '09

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The application "hash"
by Alan Chandler 12 Aug '09

12 Aug '09

I was trying to explore the documentation for the application "hash" which is in the default dialplan. Its vaguely obvious what its doing, but I wanted to be sure. It appears to be listed as a application under dp_tools, but when I click on it I get taken to a page that talks about limit_hash rather than hash. Is there any documentation for this? Am I looking in the wring place? -- Alan Chandler http://www.chandlerfamily.org.uk

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random route selection
by Juan Backson 12 Aug '09

12 Aug '09

Hi, I would like to implement a random route selection based on some arbitrary percentage. Does anyone know if there is any good way of doing that within freeswitch? If there isn't any api that I can use, does freeswitch has any random generator that I can be used for this purpose? br, JB

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FS 1.0.4 official core dumping LUA streamFile
by Charles Boening 12 Aug '09

12 Aug '09

Greetings, I have the following LUA script (at end of email) in a fresh FS 1.0.4 install. I originally did an upgrade from one of the 1.0.4preX versions but when I came across this issue I went fresh just to make sure there wasn't an incompatibility with my previous config. What I'm seeing is a seg fault and a core dump after playing a sound file. I originally had a file I recorded but when I ran into this issue I figured I'd try an included sound file but that doesn't seem to make a bit of difference. 2009-08-11 22:52:20.327071 [INFO] switch_cpp.cpp:1130 Starting test.lua 2009-08-11 22:52:20.327071 [INFO] switch_cpp.cpp:1130 Caller [XXXXXXXXXX] connected 2009-08-11 22:52:20.327071 [INFO] switch_cpp.cpp:1130 Pre streamFile Segmentation fault (core dumped) Any ideas? Thanks, Charlie freeswitch.consoleLog("INFO", string.format("Starting test.lua\n")) session:answer(); session:setHangupHook("session_hangup_hook") calleridnumber = session:getVariable("caller_id_number") calleridname = session:getVariable("caller_id_name") if session:ready() then freeswitch.consoleLog("INFO", string.format("Caller [" .. calleridnumber .. "] connected\n")) freeswitch.consoleLog("INFO", string.format("Pre streamFile\n")) session:streamFile("conference/8000/conf-welcome.wav") freeswitch.consoleLog("INFO", string.format("Post streamFile.\n")) end function session_hangup_hook(status) freeswitch.consoleLog("INFO", "Session hangup: \n") --[[ .. status .. "\n") ]]-- error() end session:hangup()

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freeswitch time conversion
by Juan Backson 12 Aug '09

12 Aug '09

Does anyone know how to take the epoch time in switch_event_t and convert it into a format such as "Sat Jul 5 02:44:33 2009"? Is there any existing facility that I can use for this purpose? br, JB

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Re: [Freeswitch-users] FW: Sangoma/FS...
by Moises Silva 12 Aug '09

12 Aug '09

Hello Peter, I'd appreciate if you can keep the discussion going in the freeswitch-users mailing list, there are other people there that will benefit of the discussion or even can help. Read my comments below. On Wed, Aug 12, 2009 at 5:59 AM, Peter Olsson < peter.olsson(a)visionutveckling.se> wrote: > Sorry for spamming you :) But I have some more results now. I’ve tried > using another lab PBX with Q.SIG enabled, and when using that one I’m able > to connect calls as I should. At least incoming to FS, outgoing seem to have > some problems still.. So the problem for the PBX I used yesterday seems to > be both related to Q.SIG (maybe) and the PBX itself (it does connect to > other providers though, so I know the trunk works). > > > > Should I take some dumps from the PRI card to try to find out why it didn’t > work with the first one, or is this “as expected”, since they have Q.SIG > enabled? > > I have no experience with Q.SIG, so I won't be able to help much. One thing though, is that if I were you, I'd be using openzap with libpri support, is that what you are using, or are you using the ISDN openzap stack? As of the dumps, they may help, or not, but pastebin them anyways so I can make an un-educated guess. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. moy(a)sangoma.com

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hangup_after_bridge=false only works if bypass_media=false
by Jeremiah Johnson 12 Aug '09

12 Aug '09

This is an integral part of my application. I need to have FreeSWITCH outside of the media path as well as be able to do multiple bridges for the same "A" leg. /*WORKS*/ <action application="set" data="hangup_after_bridge=false"/> <action application="bridge" data="sofia/gateway/${mygateway}/1${client_one}"/> <action application="bridge" data="sofia/gateway/${mygateway}/1${client_two}"/> /*DOES NOT WORK*/ <action application="set" data="hangup_after_bridge=false"/> <action application="set" data="bypass_media=true"/> <action application="bridge" data="sofia/gateway/${mygateway}/1${client_one}"/> <action application="bridge" data="sofia/gateway/${mygateway}/1${client_two}"/> In the "DOES NOT WORK" example, the "A" leg hangs up as soon as the leg for client_one hangs up.

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problem when adding more extension
by Tzury Bar Yochay 12 Aug '09

12 Aug '09

Hi, I wanted to add more extension to freeswitch. to add extension 1050 with password 1234 I did the following: $ cd /usr/local/freeswitch/conf/directory/default created 1050.xml having all '1000' strings replaced by '1050' by typing $ sed s/1000/1050/g < 1000.xml > 1050.xml rescan and reload the xml by typing into the CLI freeswitch@internal> sofia profile internal rescan reloadxml However, when I tried to login with these credentials I got the following in the fs_cli: 2009-08-12 06:35:01 [DEBUG] sofia_reg.c:1469 sofia_reg_parse_auth() SIP username 1050 does not match auth username 2009-08-12 06:35:01 [DEBUG] sofia_reg.c:869 sofia_reg_handle_register() Send challenge for [1050(a)server_address.net] below are the content of 1000 and 1050 xml files please advise. $ cat 1050.xml <include> <user id="1050" mailbox="1050"> <params> <param name="password" value="1234"/> <param name="vm-password" value="1050"/> </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="1050"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Extension 1050"/> <variable name="effective_caller_id_number" value="1050"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/> <variable name="callgroup" value="techsupport"/> </variables> </user> </include> $ cat 1000.xml <include> <user id="1000" mailbox="1000"> <params> <param name="password" value="1234"/> <param name="vm-password" value="1000"/> </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="1000"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Extension 1000"/> <variable name="effective_caller_id_number" value="1000"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/> <variable name="callgroup" value="techsupport"/> </variables> </user> </include>

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Re: [Freeswitch-users] VoiceMail transcription
by Pete Mueller 12 Aug '09

12 Aug '09

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Spanish Prompts
by bakko 12 Aug '09

12 Aug '09

I'd like to begin record spanish prompts for FS. Do you know any software/hardware to make it? Thank you BR

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