September 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Reload dingalings jingle_profiles
by Nikolay Rogoshchenkov 03 Sep '12

03 Sep '12

Hello, Does anybody know is it possible to reload *dingalings jingle_profiles*from console? PS: *reloadxml *and *dingalings reload* did not helps Thanks. -- Rogoshchenkov Nikolay

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Re: [Freeswitch-users] set_zombie_exec
by Shaka Dow 03 Sep '12

03 Sep '12

Dear FreeSWITCH users, I have a very simple python script that is called by an extension when dialed. But despite all my attempts on resolving the issue, even after trying various versions (1.2.1 and the latest GIT clone) I still keep receiving the following messages at the console when I hangup the call while playback is being executed: 2012-09-03 12:00:24.706709 [DEBUG] switch_core_session.c:2357 sofia/internal/1000(a)192.168.0.121 Channel is hungup and application 'playback' does not have the zombie_exec flag I tried setting the zombie_exec flag but that did not help either. This problem seems to also occur when I use Lua. Here is my extension: <extension name="test1_1"> <condition field="destination_number" expression="^1000$"> <action application="set_zombie_exec"/> <action application="python" data="test1"/> </condition> </extension> Here is the python script: import freeswitch def handler(session, args): session.answer() session.execute("set_zombie_exec") session.sleep(1000) session.execute("playback","test1/welcome.wav") session.hangup() PLEASE HELP!!!! Kind regards.

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SLA call on hold
by Don Dawson 03 Sep '12

03 Sep '12

With the 1.2.0 freeswitch version we have the following problem: Phone 10 calls SLA ext 122, answer the line from one of the 3 phones that have SLA(122) then put call on hold, all is good with sla. If one picks up the line(122) from another phone it hangs up. This does work on the following version: FreeSWITCH Version 1.0.head (git-1086cba 2011-05-23 22-51-43 -0500) Anybody tried this on the 1.2.0 version?

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Re: [Freeswitch-users] prevent internal extensions from getting calls from outside
by Juan Pablo L. 03 Sep '12

03 Sep '12

Hi, yes we have a block of DIDs, externally you would dial 2802002 and the gateway will send to the PBX 2002 which is the extension number. Then the PBX should decide if 2002 is allowed to get the call or it should present an IVR with an error message. Thanks. -----Original Message----- From: Michael Collins Sent: 1 Sep 2012 03:41:46 GMT To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] prevent internal extensions from getting calls from outside How is the caller getting to the extension in the first place? Do you have a block of DIDs or is the caller dialing an ext at an auto attn? -MC On Friday, August 31, 2012, Juan Pablo L. <jpablolorenzetti(a)hotmail.com> wrote: > Hi, i have a gateway that allows us to call outside and allows calls to reach the extensions from outside, > but i have a bunch of extensions that we dont want to be able to receive calls from outside. > We have a block of numbers, if any of those numbers is dialed from the PSTN or any external network > the call will be routed, through the external gateway, to the PBX so it is up to the PBX to decide > if the call should reach the extension or not. there are some restrictions .. i can not have different number segments > for the extensions, they all are in the same segment, a given extension could not be allowed to be reached from outside > at this time but it may be reachable in a latter time (due to change in the user status in the company), etc ... > i tried to put a variable in the directory for a user but when the call arrives at the PBX and the PBX is routing > that variable is not available yet and i m having trouble figuring out when that variable is available before bridging the call > so i can use it as a condition to route the call. > > any hint will on how to deal with will help. thanks! > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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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Aliased State Not showing in my sofia status
by Nweike Onwuyali 03 Sep '12

03 Sep '12

Hi All, I am a newbie to freeswitch and I have downloaded the windows edition and have it running on my XP Pro. However I have also downloaded a free DNS dual server and installed as well. I have not been able to register the Yealink T22P phone or snom 320 phone. I noticed that when I ran the sofia status command I did not see any alias type. What should I do? Cheers Nweike

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Codec list
by Chris Martineau 03 Sep '12

03 Sep '12

Hi, Reading the wiki/codec page I am slightly confused by what the capabilities actually are. If I wish to use a speex 8000 codec with a 40ms ptime at the client would I just set speex@8000@40i in the codec preferences in order to handle this? Do I just apply the @xi format to a codec in order to change its ptime to whatever value is needed or are only certain settings available as hinted at in the transcodable codecs section? Many thanks

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Freeswitch - Stable Version
by Roberto Linck 03 Sep '12

03 Sep '12

Hello everybody, I would like to know wich version of Freeswitch is the most stable one, in your opinion and wich linux distro is the best choice to use with this version. I make this question because I need put a Freeswitch in production to handle about 1500 concurrent calls. I know have the appropriate hardware, but I really don't know the most indicated version and linux distro. I'm using CentOS 6.2 but the newest version of Freeswitch(1.2) doesn't work ok with CentOS 6.x, so I'm opened to any option, since this can be stable. I'll appreciate any contribution. Roberto Linck roberto(a)i360tecnologia.com.br

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IVR gives mod_dptools.c:1816 Unable to find menu
by Charles Bujold 02 Sep '12

02 Sep '12

Strange error since I updated Freeswitch using "Make Current" My main IVR is not playing the long greeting but jumping directly to the exit recording and hang-up. When I check the log I see that the following error: 2012-09-02 18:07:11.551338 [NOTICE] mod_sofia.c:2646 Pre-Answer sofia/internal/250(a)192.168.20.153! 2012-09-02 18:07:11.551338 [DEBUG] switch_channel.c:3057 (sofia/internal/250(a)192.168.20.153) Callstate Change RINGING -> EARLY 2012-09-02 18:07:11.551338 [DEBUG] switch_core_session.c:778 Send signal sofia/internal/250(a)192.168.20.153 [BREAK] EXECUTE sofia/internal/250(a)192.168.20.153 ivr(Welcome to Accra) 2012-09-02 18:07:11.551338 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/250(a)192.168.20.153 [BREAK] 2012-09-02 18:07:11.551338 [ERR] mod_dptools.c:1816 Unable to find menu 2012-09-02 18:07:11.551338 [DEBUG] sofia.c:6051 Channel sofia/internal/250(a)192.168.20.153 skipping state [early][183] EXECUTE sofia/internal/250(a)192.168.20.153 playback(/usr/local/freeswitch/recordings/Goodbye Message.wav) 2012-09-02 18:07:11.551338 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16@8000hz 1 channels 20ms 2012-09-02 18:07:11.631338 [DEBUG] switch_rtp.c:3594 Correct ip/port confirmed. 2012-09-02 18:07:15.771336 [DEBUG] switch_ivr_play_say.c:1682 done playing file /usr/local/freeswitch/recordings/Goodbye Message.wav 2012-09-02 18:07:15.771336 [NOTICE] switch_core_state_machine.c:249 sofia/internal/250(a)192.168.20.153 has executed the last dialplan instruction, hanging up. 2012-09-02 18:07:15.771336 [DEBUG] switch_channel.c:2914 (sofia/internal/250(a)192.168.20.153) Callstate Change EARLY -> HANGUP The IVR is as follows: <include>  <menu name="Welcome_to_Accra" greet-long="/usr/local/freeswitch/recordings/Welcome Message Bilingue.wav" greet-short="/usr/local/freeswitch/recordings/Welcome Message Bilingue.wav" invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav" exit-sound="voicemail/vm-goodbye.wav" confirm-macro="" confirm-key="" tts-engine="flite" tts-voice="rms" confirm-attempts="3" timeout="10000" inter-digit-timeout="2000" max-failures="3" max-timeouts="3" digit-len="5"> <entry action="menu-exec-app" digits="1" param="playback /usr/local/freeswitch/recordings/Transfer to Sales.wav"/> <entry action="menu-exec-app" digits="1" param="transfer 200 XML default"/>  <entry action="menu-exec-app" digits="2" param="playback /usr/local/freeswitch/recordings/Transfer to Customer Support.wav"/> <entry action="menu-exec-app" digits="2" param="transfer 7002 XML default"/>  <entry action="menu-exec-app" digits="3" param="playback /usr/local/freeswitch/recordings/Transfer to Administration.wav"/> <entry action="menu-exec-app" digits="3" param="voicemail default ${domain_name} 300"/>  <entry action="menu-top" digits="9" param=""/>  <entry action="menu-exec-app" digits="/(^\d{3,6}$)/" param="transfer $1 XML default"/> </menu> </include> Any help would be greatly appreciated. Thanks cjb

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cleanup of lua hangup hook after call disconnection
by Muhammad Naseer Bhatti 02 Sep '12

02 Sep '12

Hi, http://wiki.freeswitch.org/wiki/Mod_lua#Special_Case:_env_object says this hangup hook is called at the end of channel disconnection. While the channel is disconnected and the lua script is already exited, and since there are no sessions at this point, (because we respond and hangup in the dialplan sent by lua) the channel cleanup would be done by the lua garbage collector? Or maybe I am understanding it wrong? The dialplan is served by a lua script. Thanks

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mod_spy on Windows
by Kees Varekamp 02 Sep '12

02 Sep '12

Hi there, I'd like to build/run mod_spy on Windows but there isn't a VS project file for that app. Is it hard to make one? Are there known problems with this module on Windows? Any help appreciated kees

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