November 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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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send event to Grandstream phones when presence/dialog event has sent
by Tóth Róbert 26 Nov '10

26 Nov '10

Hi 1. I would like to send an event when presence or dialog events are sent. Because i have Grandstream phones, and i want to refresh their IdleScreens as it can be read the lower part of this page: http://www.voip-info.org/wiki/view/GXP-2000+XML+Idle+Screen Have i modify the code of sofia_presence.c or is there a simpliest way to send events bound to an other event? 2. And could you give an idea how can i insert records in the simpliest way to the database also when presence/dialog event sent? (Or is the simpliest to give code to sofia_presence.c?) Thank you Robert

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INVITE via another SIP proxy
by Ingmar Schraub 26 Nov '10

26 Nov '10

Hi, I ran into a problem today and I thought I should place the question here and hopefully someone has an idea what's going on. FYI, I am using GIT from November 15th 2010. Here is my problem: My SIP phones are all on the LAN. I can register them, make calls, etc. all work fine. I run FS with "-nonat". Now, when I add another SIP proxy in the chain, let my phones register with FS via that proxy, everything is still fine. No issues with registrations. The problem is, that INVITEs do not work. FS rejects them with "400 Bad Request". I turned on SIP tracing on FS to see how the SIP messages look like and I noticed that FS is for some reason stripping off (apparently..) the SDP part from the INVITE. Why do I believe that? Well, I used tcpdump on the FS box to capture the network traffic and see what arrives actually. The INVITE received by the network stack shows the full INVITE message including the SDP part. On the other hand, when I make a call / INVITE from a phone which is directly registered at FS, the INVITE message shown in the sip trace includes the SDP. Thus, I believe that FS has an issue parsing that particular INVITE message properly. Note: the same proxy, the same set-up works all sorts of other PBXs and Soft-Switches (Asterisk, Avaya, 3com, Nortel, Cisco, Sonus, ...). It's all tested. Of course I can provide the FS SIP trace log and the corresponding tcpdump. If any developer could have a quick look, I'd really appreciate this. Thanks and best regards, Ingmar

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Load Balancing using OpenSIPS and B2BUA
by Abid Saleem 26 Nov '10

26 Nov '10

Hi All, What is the best load balancing method to use with FS. I think using OpenSIPS method seems to be better. Could anyone tell me who will work as a B2BUA in this setup OpenSIPS load balancer server or individual FS boxes? In case we use OpenSIPS as the B2BUA which is not a full B2BUA and does not handle media, so how this scenario should be best configured to provide optimum performance. Suggessions, recommendations and experiences are required. Thanks Regards----------Abid SaleemTechnical Manager NGNTerminus Technologies

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