Re: [Freeswitch-users] if using centos you should read this

:-)! On Fri, Jul 31, 2009 at 7:36 AM, Muhammad Shahzadshaherya...@googlemail.com wrote: Please read my email as, CentOS has been a trusted platfrom for me from last 3+ years. I have developed and deployed many FS and Asterisk solutions on it, 9 out of 13 FS boxes, and 27 out of 49 Asterisk

Re: [Freeswitch-users] Sangoma a101

Thanks for help. Got my alarms sorted - telco had problem there. As for my Freeswitch problem, chaning to Q931 clears the error message. I was asked by telco to specifically use telco, will using Q931 cause any issues for me? Thanks again, Niall. 2009/7/31 Peter P GMX prometheus...@gmx.net:

[Freeswitch-users] FreeSwitch doesn't play music on hold for briged channel

The scenario is the following: FS User A dial an extension Extention opens outbound socket channel to my application My application bridges the call to FS User B The application check for CHANNEL_BRIDGED event and stores Other-leg-unique-id The application sends hold to the bridged channel using

Re: [Freeswitch-users] Canceling att_xfer?

Anthony Minessale wrote: i think you dial # instead of 0 There appears to be a bug, when I try this; - A continues to hear moh - B can't send any more DTMF or audio to C (but can still hear C) - C can only send audio to B I've posted the debug cli log on pastebin:

Re: [Freeswitch-users] pocketsphinx

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello, I spent a few days working on my problem. I played around with voxforge's data, read some of the train docs of CMU, and debugged the sphinxbase, pocketsphinx and mod_pocketsphinx. Results: - -I found a way to use the voxforge data as training

Re: [Freeswitch-users] FreeSwitch doesn't play music on hold for briged channel

I don't see the variable hold_music ... did you remove it? /b On Jul 31, 2009, at 5:24 AM, Kozak Vladimir wrote: The scenario is the following: FS User A dial an extension Extention opens outbound socket channel to my application My application bridges the call to FS User B The application

Re: [Freeswitch-users] Sangoma a101

I was asked by telco to specifically use telco, will using Q931 cause any issues for me? Sorry, I do not not understand that question. Best regards Peter Niall Crosby schrieb: Thanks for help. Got my alarms sorted - telco had problem there. As for my Freeswitch problem, chaning to Q931

Re: [Freeswitch-users] pocketsphinx

On Jul 31, 2009, at 8:18 AM, Helmut Kuper wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello, I spent a few days working on my problem. I played around with voxforge's data, read some of the train docs of CMU, and debugged the sphinxbase, pocketsphinx and mod_pocketsphinx.

Re: [Freeswitch-users] Async JS functions?

Hi Nicolas, I wonder if session.execute( sched_broadcast, +1 /path/file.wav aleg) followed immediately by your 'new Session' and 'bridge' would do the trick ? Not sure if/how sched_broadcast functions when the call has not yet been bridged though... Let us know.. Best Regards

[Freeswitch-users] Language of the speech

Hi again, does anybody know why my freeswitch doesn't play German speech-files? Here is my construct: I'm generating a random number in my lua script: rand = math.random(11, 1000); The I want that the freeswitch says the random number: session:execute(say, de name_spelled iterated

[Freeswitch-users] LUA: Independent control of each call leg.

Hi, I'm trying to develop an application using lua and need to control the inbound and outbound legs independently, even when they are switched together. I can initiate the outbound session but I can't seem to bridge without losing control of the script. For example, if I use: local

Re: [Freeswitch-users] Sangoma a101

euro and q931 are the same thing on most devices. Europe uses pure q931 where most other places use enhanced versions of it. On Fri, Jul 31, 2009 at 9:45 AM, Niall Crosby niall.cro...@gmail.comwrote: Sorry, my mistake. I was asked by telco to specifically use Euro dialect, will using Q931

Re: [Freeswitch-users] Fwd: IVR Problem

Like I said you set the event lock to true right away when you create the connection, Then all your sendmsg events will be queued and processed one at a time instead of as received On Fri, Jul 31, 2009 at 12:04 AM, Thangappan.M thangappan...@gmail.comwrote: If I missed out the async mode

[Freeswitch-users] Customized voicemail greeting

Is it possible to define a custom voicemail greeting (wav file) on an extension-by-extension basis? I have read the docs on mod_voicemail and searched previous emails, but I don't know where to begin. Thanks, Lars ___ FreeSWITCH-users mailing

Re: [Freeswitch-users] Async JS functions?

Thank you Keith, I'll try that as well and let you know. On Fri, Jul 31, 2009 at 9:51 AM, Keith Laaks kei...@voxtelecom.co.zawrote: Hi Nicolas, I wonder if session.execute( sched_broadcast”, “+1 /path/file.wav aleg”) followed immediately by your ‘new Session’ and ‘bridge’ would do the

Re: [Freeswitch-users] Language of the speech

we don;t have any german sound files? On Jul 31, 2009, at 10:31 AM, Rudolf Denert wrote: Hi again, does anybody know why my freeswitch doesn't play German speech-files? Here is my construct: I'm generating a random number in my lua script: rand = math.random(11, 1000); The I want that