- FreeSWITCH-users - FreeSWITCH Mailing Lists

session_hangup_hook
by Madovsky 08 Feb '11

08 Feb '11

the caller from outside call a registered user, I'd like to run a script once the caller hangup. is session_hangup_hook available for outbound call also ?

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Setup Project
by Daniel Wells 08 Feb '11

08 Feb '11

So when I launch the Visual Studio 2010 Project I get an error telling me that the setup project cannot be opened (specifically the setup.wixproj file). Without this project it would seem a rather complicated process to get my compiled code to run a separate box (compiling on dev box but want this to run on a separate server). Any ideas? Sincerely, Daniel Wells

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Opus codec in FreeSWITCH
by Jason White 07 Feb '11

07 Feb '11

I have started testing the Opus 0.9.0 experimental codec in a recent build of FreeSWITCH - thanks to Anthony Minessale for writing the module, and thanks to William F. Acker for helping with the testing. For those who aren't familiar with it, Opus is a combination of CELT and SILK, which is currently under discussion within the IETF. After building and loading the module, specify Opus-0.9.0 as your codec. The versioning of the codec's name is a good idea; this should prevent mutually incompatible versions from interacting with each other as development of the codec proceeds. My initial impressions are favourable; the audio quality is comparable to CELT at 48 khz, based on my early listening experience. I did notice some occasional pops in the audio, presumably attributable to packet loss. SILK apparently has some error correction functionality built in, but I don't know what its status is in the combined Opus codec. If it's present, then it didn't smooth out the glitches, at least in my environment.

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close all conference from fs_cli
by Madovsky 07 Feb '11

07 Feb '11

Is there a way to close/hangup all active conferences from CLI ? Thanks

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Re: [Freeswitch-users] Simple_conference.lua
by Francois Barrouin 07 Feb '11

07 Feb '11

You could consider the following alternatives: - use TTS to generate your prompts: http://wiki.freeswitch.org/wiki/Categories - take the prompt files from Asterisk Surprising that FS doesn’’t include these prompts. François 2011/2/3 John Smith <johns1433(a)gmail.com> Hi, I’m trying to have the script http://wiki.freeswitch.org/wiki/Simple_conference.lua running on my FreeSwitch server. I get an error when FS wants to play any phrase given in this example: [ERR] switch_ivr_play_say.c:150 Can't find macro conference_welcome It seems normal as I didn’t find any reference to such phrases in the different xml phrase configuration files. There seems also to be no adequate wav file in sounds/en/us/callie/voicemail/8000. My question is: Do these files exist somewhere or does the example assume that we have to record our own wav files? Thanks John _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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Group confusions
by Lars Zeb 07 Feb '11

07 Feb '11

I went to two individual extensions and registered them for group 01 by dialing 8101. I then tested that group by dialing 8201 - both extensions rang. I then went to the cli and entered the command: group delete:01:1013@192.168.10.29 where 1013 is one of the two extensions I had joined into group 01 earlier and the url is the domain and fs address. However, when I dialed 8201 after entering this cli command, both extensions rang. I had expected that only one would ring. I am confused. Are these two different types of "groups"? Thanks, Lars

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Console Shutdown always crashes
by Daniel Wells 07 Feb '11

07 Feb '11

Any idea why the console always crashes on shutdown (compiled on Windows). These are the last few logs to the console: [CONSOLE] switch_loadable_module.c:1405 mod_siren has no shutdown routine [NOTICE] switch_loadable_module.c:518 Deleting Codec SPEEX 99 Speex 32000hz 20ms [NOTICE] switch_loadable_module.c:518 Deleting Codec SPEEX 99 Speex 16000hz 20ms [NOTICE] switch_loadable_module.c:518 Deleting Codec SPEEX 99 Speex 8000hz 20ms [CONSOLE] switch_loadable_module.c:1405 mod_speex has no shutdown routine I thought that it must have been because there was no shutdown routine for mod_speex but I noticed mod_siren doesn't either and it passes it ok. Sincerely, Daniel Wells IT Systems Engineer

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mod_dinglaing errors and google voice
by Daniel Wells 07 Feb '11

07 Feb '11

So I finally have what appears to be a working version of mod_dingaling on my windows box. I say working only because the console starts and I see evidence in the logs that freeswitch is communicating with my google account though mod_dingaling (I see presence information about some of my contacts). However I am not sure my dialplan is working (just a guess since this my first experience with freeswitch-or any PBX for that matter). I have been trying to follow the instructions as found here: http://wiki.freeswitch.org/wiki/Google_Voice I have a client connected using the default user 1000. I created a dingaling profile called gv1000.xml (I am using the second example of the dialplan on the above site and inferred that the client profile needed to renamed). My profile is identical to the one in the example with the exception of the username, password and "exten" (which is set to 1000). I put a copy of the dial plan in the "default" directory. But I am getting an error when trying to make outbound calls. Does this mean I have done something wrong, or is google blocking this now? freeswitch@dw-laptop> 2011-02-07 10:31:44.361514 [WARNING] sofia_reg.c:1247 SIP auth challenge (INVITE) on sofia profile 'internal' for [18XXXXXXXXX(a)192.168.2.97] from ip 192.168.2.97 2011-02-07 10:31:44.477521 [NOTICE] switch_channel.c:811 New Channel sofia/internal/1000(a)192.168.2.97 [cfda05d4-2f61-4ab0-8ca6-f759cfac53b7] 2011-02-07 10:31:44.689533 [INFO] mod_dialplan_xml.c:331 Processing 1000 <1000>->18XXXXXXXXX in context default 2011-02-07 10:31:44.689533 [NOTICE] mod_sofia.c:2185 Ring-Ready sofia/internal/1000(a)192.168.2.97! 2011-02-07 10:31:44.689533 [NOTICE] mod_dptools.c:697 Ring Ready sofia/internal/1000(a)192.168.2.97! 2011-02-07 10:31:44.689533 [NOTICE] mod_dingaling.c:721 Close Channel N/A [CS_NEW] 2011-02-07 10:31:44.689533 [ERR] switch_ivr_originate.c:2638 Cannot create outgoing channel of type [dingaling] cause: [DESTINATION_OUT_OF_ORDER] 2011-02-07 10:31:44.689533 [INFO] mod_dptools.c:2621 Originate Failed. Cause: DESTINATION_OUT_OF_ORDER 2011-02-07 10:31:44.689533 [NOTICE] mod_dptools.c:2684 Hangup sofia/internal/1000(a)192.168.2.97 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] 2011-02-07 10:31:44.900545 [NOTICE] switch_core_session.c:1306 Session 1 (sofia/internal/1000(a)192.168.2.97) Ended 2011-02-07 10:31:44.900545 [NOTICE] switch_core_session.c:1308 Close Channel sofia/internal/1000(a)192.168.2.97 [CS_DESTROY] Sincerely, Daniel Wells

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conferenceproblems after execute_application
by Thomas Hoellriegel 07 Feb '11

07 Feb '11

Hi all, i have a working freeswitch git on ubuntu lucid 64bit. I setup a conference. I add these line in my call-control-section: <control action="execute_application" digits="3" data="playback /fssounds/helpconference.wav"/> I press 3 the file will be played correctly. After the playback i go back to the conference. I talk to others in the conference. after the "execute_application" command i have many delays 3 or 5 or more seconds. My voice is very clearly. the people mean i.m a voicecomputer-). I must exit the conference and i start again. my voice is normal. is execute_application a problem? can you help me please? thanks. --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste für blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc

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mod_fsk not detecting call waiting caller id
by Anirudha Shimpi 07 Feb '11

07 Feb '11

Waiting for digit 'D' did not yield any results. Any other ideas? I am very much interested in getting call waiting caller id to work with Freeswitch and analog lines. Any suggestions would be very helpful. If there are any modifications that can be done to mod_fsk to get it to work, or writing a new module for this purpose, I am ready to help in any fashion. Thanks

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