- FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_voicemail problem/question
by covici＠ccs.covici.com 28 Nov '10

28 Nov '10

Hi. I was under the impression that <param name="vmain-key" value="*"/> meant that if you pressed * while the greeting was showing, it would prompt you for the password and you could retrieve your voicemail when not at the console. The question is, is this supposed to work this way and if not, what is the function of this key? Also, what is vmain-extension and is it related to the above? thanks much for all your help. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici(a)ccs.covici.com

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Compiling on FreeBSD
by Adrian Gschwend 28 Nov '10

28 Nov '10

Hi group, I use Freeswitch on FreeBSD for ages now, always trunk which I compile every few weeks. Unfortunately it doesn't compile anymore for a while, it fails in mod_spandsp I checked the docs according to: http://wiki.freeswitch.org/wiki/Installation_Guide#FreeBSD http://wiki.freeswitch.org/wiki/Mod_spandsp so I compiled both graphics/tiff-4.0.0 and graphics/jpeg-8_3, I'm on 8.1-RELEASE But it doesn't find jpeg, like the possible error stated on Mod_spandsp page. But unless I completely miss something here it *should* be installed (in terms of the port is definitely there). -- [...] Creating mod_spandsp.la /usr/bin/ld: cannot find -ljpeg quiet_libtool: link: gcc -shared .libs/mod_spandsp_la-mod_spandsp.o .libs/mod_spandsp_la-udptl.o .libs/mod_spandsp_la-mod_spandsp_fax.o .libs/mod_spandsp_la-mod_spandsp_dsp.o .libs/mod_spandsp_la-mod_spandsp_codecs.o -Wl,-rpath -Wl,/usr/local/src/freeswitch/.libs -Wl,-rpath -Wl,/usr/local/freeswitch/lib -L/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib/.libs -L/usr/local/src/freeswitch/libs/apr/.libs -L/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff/.libs -ljpeg /usr/local/src/freeswitch/.libs/libfreeswitch.so -L/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib /usr/local/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a /usr/local/src/freeswitch/libs/apr/.libs/libapr-1.a -lcrypt -lpthread /usr/local/src/freeswitch/libs/spandsp/src/.libs/libspandsp.a -L/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff /usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff/.libs/libtiff.a -lz -lm -lssl -lcrypto -lncurses -Wl,-soname -Wl,mod_spandsp.so -o .libs/mod_spandsp.so gmake[4]: *** [mod_spandsp.la] Error 1 gmake[3]: *** [mod_spandsp-all] Error 1 gmake[2]: *** [all-recursive] Error 1 -- Any FreeBSD user which can compile that ATM? If so, what did I miss? Will update the wiki once I get it to work. cu Adrian -- Adrian Gschwend @ netlabs.org ktk [a t] netlabs.org ------- Open Source Project http://www.netlabs.org

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Re: [Freeswitch-users] SQLite CoreDB issues
by Jeff Lenk 28 Nov '10

28 Nov '10

there is core odbc support available and has been for a while. Please post the reference in the Wiki so we can get it corrected. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/SQLite-CoreDB-issues-tp576579… Sent from the freeswitch-users mailing list archive at Nabble.com.

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press a key
by Will Smith 28 Nov '10

28 Nov '10

hi, I am working on a small app that will tap to a line to record conversation. After dialing the number, I need to send username and password. Here I need to send dtmf signal (in javascript). I tried if(session.ready()){session.execute("queue_dtmf", "12345");} but looks like I did that wrong. Also, I need to detect if there is a silence at the beginning, or in other words, if there is a conversation happening or not, I try to use the function session.recordFile(...) and try to make use of the silencehits value. I am not sure that is the right way to do, and there is a good way to do that. Thank you for all your help. Will

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Call-State in LUA
by Tóth Róbert 28 Nov '10

28 Nov '10

*I would like to get call-state, instead of channel-state. Can i? Michael Collins* <msc at freeswitch.org> <mailto:freeswitch-users%40lists.freeswitch.org?Subject=%5BFreeswitch-users%5D%20Call-State%20in%20LUA&In-Reply-To=4CEAEFAE.7050606%40freemail.hu> wrote: >/ /You need to use eval. From the fs_cli it's like this: >/ / >/ /eval uuid:<uuid> ${channel-state} >/ / >/ /Execute that as an API from inside Lua and you should get your call state. >/ /-MC >/ / >/ /On Mon, Nov 22, 2010 at 2:33 PM, Tóth Róbert<trob at freemail.hu <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>> wrote: >/ / >/ />/ Hi, />/ />/ />/ />/ Is it possible to get a channel's Call-State in LUA? Is there a function />/ />/ for this similar to />/ />/ session:getState()? />/ />/ />/ />/ (I would like to know if a channel is in HELD state.) />/ />/ />/ />/ Thanks /

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dialplan ring_ready/sleep question
by Steve Richardson 28 Nov '10

28 Nov '10

Hello, I am attempting to use the dialplan ring_ready and sleep applications as follows (from my public.xml file): <extension name="foo"> <condition field="destination_number" expression="5551234567">  <action application="ring_ready" /> <action application="sleep" data="30000"/>  <action application="hangup" data="NO_ANSWER"/> </condition> </extension> My intent is to have the incoming call in the ring state for up to 30 seconds, during which time I would like to issue API commands from my custom application (which is using the event socket interface) to transfer the call to various places. The issue that I'm having is that the sleep seems to be non-interruptable. If I try to issue a uuid_transfer command to transfer the sleeping channel somewhere, it seems the sleep timer has to expire before it will transfer. Similarly, if the caller disconnects, the channel stays around until the sleep timer expires, and then it is destroyed. Is there some way to wake a sleeping channel out of sleep? I've looked for anything obvious and tried a number of things to no avail. Alternately, is there a better way to approach this? thanks, Steve

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Extensions cannot be called --> "Default" does not work
by Istvan Turbucz 28 Nov '10

28 Nov '10

Hi, I have set up an auto attendant. I give a call to a number, I hear the music. Then I try to choose an extension. (This is allowed in the options of the auto attendant.) In this case I receive "no_route_destination" because the ext. number is sent to a trunk and is matched by the dialplan ^(\d)$ What can I change that the calls are sent through "default" again? "Default" is of course set by each extensions. Internal.xml and external.xml are recovered to the default ones. Freeswicht is restarted. I use FusionPBX as GUI. I have still the problem. It is anyway possible to give a call to an external number. All the trunks and extensions are registered on a correct way. Has somebody an idea? Thank you for the help. Regards, Istvan

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Re: [Freeswitch-users] skypopen problem & X
by Ryanny Lin 26 Nov '10

26 Nov '10

hi Giovanni, yeah, a crash. In fact, I also don't know how to explain what happens. I am trying to find out how to reproduce it. oo thx^^ > Hi Ryanny, > > maybe I have not understood what happens, what you do, and what is the > problem (I mean, a crash - that I understand ;) ). > > Can you explain more? Best would be to suggest a procedure to reproduce it. > > -giovanni -- Sincerely regards, Ryanny

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fryswitch transfer call problem
by Михаил Салтанов 26 Nov '10

26 Nov '10

Hello I have some problem with freeswitch call transfer by python script. ----------------------------------------------- import os from freeswitch import * import psycopg2 import socket def handler(session, args): print 'ready to transfer' number_to_transfer = session.getVariable('number_') print number_to_transfer session.setAutoHangup('false') session.execute('set', 'hangup_after_bridge=false') session.execute('set', 'continue_on_fail=true') session.transfer( number_to_transfer, "XML", "default") Description of my problem Phone # 1 call to Phone #2 Phone #2 input some digits for transfer incoming call to phone #3 [phone #1-->phone#3] Phone #1 hangup, Phone 2 connect to phone # 3 where the my mistake? When I make transfer by xml extensions all works fine С уважением Михаил Салтанов

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Executing multiple commands on the fs_cli or using mod XMLRPC
by Pete Kelly 26 Nov '10

26 Nov '10

HI I have an application which automatically calls a user on his request. At the moment I am using the originate command to create these calls using the XMLRPC interface into freeswitch. The command I am effectively running is: originate sofia/external/2000(a)voip.example.com 2000 However the command does not return until the call is picked up or the command times out, so I can not make more than one simultaneous outbound call. I found the same when I use mod xmlrpc, it seems to queue the requests and waits for the first one to finish before executing the second. Is there a way using mod XMLRPC I can tell these commands to run immediately? I have the only way I seem to be able to create the calls simultaneously is to use the fs_cli command: /usr/local/freeswitch/bin/fs_cli --execute "sofia/external/2000(a)voip.example.com 2000" Pete

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