Sometimes my process just drops the call and leaves the client in silence!
it happens probably 1 out of 25 times... no meaningful loads happening, just
me prototyping my (Ruby)AGI...
On Wed, May 7, 2008 at 12:40 PM, Robert Norton - SophTelecom LLC
[EMAIL PROTECTED] wrote:
Hi Marcelo,
Sorry,
Hi. I am using the 'get_data' function from an AGI, and i find that
sometimes when users call in, it won't play the full gsm soundfile, and when
i try to press a number (or pound, or star), nothing will happen - it just
hangs there...
anyone else experience this?
- Dominic Son
Could you please call it and confirm with me it's not working for you
either? I should probably transfer my DID number anyways, if I could only
get them to respond! Does anyone have a suggestion as to where to go in this
situation? Possibly a place with high capacity concurrent incoming calls...
Hi. I'm still a bit of a newb to linux, I see this in my messages log:
Asterisk init: Id ax respawning too fast: disabled for 5 minutes
What does this mean?
and how severe is it?
--
Anything else, let me know.
- Dominic
It is not the force of a stroke that makes fine art
I was previous using Asterisk 1.2.9.1 and decided to get some real servers
outside of my house. It was time for Asterisk 1.4.4.
I figured since all the conf files were in /etc/asterisk form the old box,
i'd just copy tha directory over to the new server. My SIP DID AGI stuff
worked, except
awesome. it worked. thanks guys.
On 10/20/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Sat, Oct 20, 2007 at 11:57:05PM +0530, Jaswinder Singh wrote:
astrundir = /var/run
Change this to astrundir = /var/run/asterisk on 1.4 server and chmod
/var/run/asterisk to 777 . make sure u create
Hi. In the Asterisk TFOT book, under installing Asterisk, it says as an option:
Uncomment the following line in your Asterisk Makefile to enable GSM
codec optimizations
on x86 CPU architectures that support MMX instructions:
#K6OPT = -DK6OPT
Does installing from the basic Trixbox CD possibly do
Ok, this is what worked:
EXEC System rm -rf /var/lib/asterisk/sounds/blah.gsm
the -rf eliminates the hassle.. a dream come true it worked !
On 10/13/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Sat, Oct 13, 2007 at 05:45:26PM -0700, Dominic Son wrote:
Hi.
You mean to use the AGI funtion
Uuugh..for the life of me, i cannot delete sound files using
EXEC System(rm /var/lib/asterisk/sounds/blah.gsm)
through AGI
the AGI debug log indicates the command executes successful ( equals 0)
but my files are clearly still there.
If i try System(rm ...) in my extensions.conf diaplan it'll
it if they choose to.
so there's a 'RECORD FILE', but not a 'DELETE FILE' .. : T
On 10/13/07, Philipp Kempgen [EMAIL PROTECTED] wrote:
Dominic Son wrote:
Uuugh..for the life of me, i cannot delete sound files using
EXEC System(rm /var/lib/asterisk/sounds/blah.gsm)
through AGI
agi show
Hi.
You mean to use the AGI funtion in the particular programming
language? yeah. i tried, same results.. : T
i guess i'll have to put it in a database, and flag it to remove
manually for now...
On 10/13/07, Philipp Kempgen [EMAIL PROTECTED] wrote:
Dominic Son wrote:
tried both
Hi. I'd like for my sound files to be exposed through http.
You know, the ones located in var/lib/asterisk/sounds.
This is probably an apache thing i'd have to configure or is
accessible through some asterisk http routing?
1. how one would configure this?
2. what are the security costs of doing
Hi.
Was the AGI Server to write dialplans in any programming language in
Asterisk assumed to be configured for the apache web server?
Or should it not matter what web server you have (in my case NGINX)?
- Dominic
The ability to simplify means to eliminate the unnecessary so that the
necessary
(Asterisk default media ports).They'll contact yo when a call comes in. You'll accept the call and atthe same time tell them which port to send the incoming audio to.They'll also tell you where to send your outgoing audio.
Hope that helps.MarkOn Fri, 2006-08-11 at 15:45 -0700, Dominic Son wrote: Hi. Can
Hi. Can someone explain to a right brained person what is going on with In/out bound trunks, how it connects to my Trixbox..1. i get issued a free NY phone number from a voip service like stanaphone . 2. i then call this number, it connects to the stanaphone voicemail
3. i turn off the voicemail
.-Dominic Son
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