of blogger.
As a legal note, I make absolutely no claims or warranties that this
application actually works or the following blogs display accurate
data from the CRTC or FCC's site.
Here they are:
for the CRTC: http://crtc.gulfpearl.com
for the FCC: http://fcc.gulfpearl.com
---
Shidan Gouran
I saw that already. It's not a listing of the FCCs headlines. It's
just a very lame, unusable, unordered list of a few snippets from
random FCC meetings. Check the data in my feed and compare it that and
I think the answer to what about this becomes obvious.
Cheers,
Shidan
On Fri, Jan 16, 2009
or the problem was, I haven't had a need
to modify the box in almost two years apart from the odd security
patch and yum upgrade.
---
Shidan
On 4/16/07, Voip Asterisk [EMAIL PROTECTED] wrote:
Awesome, any chance you can share your resource specs?
Thanks
Miles
Asterisk works great with openvz
Asterisk works great with openvz. Ive run 4 VE's with combined average
around 32 simultaneous calls at any time and you wouldn't know the
difference.
On 4/13/07, Voip Asterisk [EMAIL PROTECTED] wrote:
Anyone here running asterisk on openvz, if so what are your experiences?
Right now we are
I think at best its a bug with the safe_asterisk script and at worst
could be a bug with asterisk itself. I can't see how this is a
configuration issue with freepbx.
---
Shidan
On 10/11/06, Dinesh Nair [EMAIL PROTECTED] wrote:
On 10/11/06 21:15 Joseph said the following:
I quits on my
Hi so with my setup of asterisk 1.4 and installing freepbx on it, I
have everything working fine now except one thing, the remote console
keeps crashing after a reload. Any ideas?
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experienced
this problem? Also, I'm running this in the cusom confs for
freepbx.
Just to save time, I'm not looking for a solution that doesn't
use the Disa app.
Cheers,
Shidan Gouran
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asterisk
from indications.conf ( but the file is
correct).
So if it's found to be a more general problem thats something to look at.
Thanks for all your responses.
---
Shidan Gouran
On 9/29/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Make sure you have a /etc/asterisk/indications.conf
Shidan
First off, I don't work for Ericsson, but it is my impression that normally the
call control is done in Erlang and the media processing is done ineither hardware orC.I never said you work in Ericsson ;)
I You really are interested I recommend You to ask on[EMAIL PROTECTED]I did but they stopped
I don't know if I responded to the original poster before but if you are looking for a python fastAGI server, there already is one, its called starpy.Anders, since you know Erlang, do you know of any media processig libraries in Erlang, do the ericsson softswitches do the media processing
np, but in general its well worth the learn tho if you like python ;)
On 8/3/06, shawn bright [EMAIL PROTECTED] wrote:
Thanks for the reply Shidan,
i have looked at this package before, but was not exactly excited about
having to learn twisted, get twisted up and running, etc... just to get
Check out http://starpy.sourceforge.net/ if you have any questions let me know.
---
Shidan
On 8/2/06, shawn bright [EMAIL PROTECTED] wrote:
Lo there,
i have an app that needs to initiate a phone call on a zap channel.
i have been able to test it out ok with the method of dropping a call
developed a
zaptel interface. It's a shame, they are amazing cards.
---
Shidan
On 5/11/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
What makes you say that?
Nabeel Jafferali wrote:
That list is obviously not complete ;)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
If you want someone would will give cheap termination to end users, go
use voipjet or whatever you want.
If, on the other hand, you want some reliable cheap wholesale
termination, go check out voipjet.
glad to know theres options out there !
-
Shidan
A bit old but a couple of interesting articles in here:
http://www.acmqueue.com/modules.php?name=Contentpa=list_pages_issuesissue_id=16
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Asterisk-Users mailing list
To UNSUBSCRIBE or update
Sorry about the non asterisk thread, sent it to the wrong list
On 2/10/06, Shidan [EMAIL PROTECTED] wrote:
A bit old but a couple of interesting articles in here:
http://www.acmqueue.com/modules.php?name=Contentpa=list_pages_issuesissue_id=16
easy to
customize your pbx interface with Destar, and that can translate to
$$$ if you are selling pbx's or if you are a service provider, adding
an interface for functionalities specific to your offerings. Hopefully
by this time next year more people will be talking of it.
---
Shidan Gouran
On 1
How can I get * to listen on IP Aliases for sip, seems to not be
working for me for bindaddr=0.0.0.0.
Another question, in a channel can you define more than 1 bindaddr and
would this resolve the problem with IP Aliases. I'd test it out but
right now I have 10 other things to experiment with!
How hard would it
be to turn the Manager into an RSS feed if XML were an option? (stupid
example but you get my idea)
How hard is it now for a particular task, about 20 lines of code, in
less than an hour. Writing any sort of xml proxy to the manager
interface is really trivial for any
How hard would it
be to turn the Manager into an RSS feed if XML were an option? (stupid
example but you get my idea)
How hard is it now for a particular task, about 20 lines of code, in
less than an hour. Writing any sort of xml proxy to the manager
interface is really trivial for any
that's your problem because it wasn't
documented, I would really like to know where this official
documentation is apart from reading the source code. Most successful
Asterisk based operations are novel and creative hacks, not structured
formulas that follow any official documentation.
Shidan Gouran
I don't get the whole concept of FastAGI. It's nothing special. Asterisk just
opens a connection to a TCP port instead of executing a binary.
Which is exactly what would solve your supposed problem with AGI
scripts, there is no overhead with starting up the interpreter so you
can just write a
you
heard of fastcgi after 14 years of Linux? So either you are clueless
about any sort of net/web/admin development with perl or you drank
alot of caugh syroup half way through this thread.
Shidan
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Hi I have a client who needs T38 Faxing capabilities. We have tried
using T38Modem with Hylafax on the same box with * to no avail. Are
there any T38 gateways that are relatively inexpensive or is there an
* solution anyone has used successfully in production.
Cheers,
Shidan Gouran
and * doesn't support these yet.
---
Shidan
http://www.nuovotel.com
On 7/14/05, Angel Diaz [EMAIL PROTECTED] wrote:
Well, I mean Instant messaging between two SIP users registered on
Asterisk-sip server.
The thing is, some sip phones supports instant messaging but, how can I get
this feature work
*.
As to does Asterisk support MESSAGE requests with a plain/text MIME
type, you can use the ap. SendText() , look it up on the wiki
Regards,
Shidan
http://www.nuovotel.com
On 7/13/05, Angel Diaz [EMAIL PROTECTED] wrote:
Hi,
Is there a way to send and receive SMS over SIP protocol
Hi noticing the cvs updates of late, I'm wondering if there is support
for fifo/shell commands in the extended dialplan language? can it
fully replace agi scripts? Looks really interesting...
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I have a client requirement that multiple phones can be dialed,
however they don't want the pstn phone to pick up automatically
because of voicemail etc, nothing can be changed on the phones, how
can I handle this requirement, by the way no zap channels are
involved, all the pstn phones are
there was an option for the user to hit * to accept the call but I
think thats only with ZAP, anyone know of a solution to this problem
or something similar for SIP.
Shidan
On 6/8/05, Jennifer Hales [EMAIL PROTECTED] wrote:
If you want to dial a number of phones at the same time do exten =
5000,1,Dial(SIP
Is there an easy way to make Astcc silent, so that it does not tell
the user how much money he has and the cost to a location, but rather
does call control silently. Whats the general consensus of astcc vs
areski. Thanks for your advice in advance.
Shidan
Your talking garbage. Some of the most highly concurrent and cost
intensive programs have been written in Python for its amazing support
of co-routines and generators and asynchronous event handling. Even
Yahoo chose PHP over Java for their website and that has more than 20
users believe it or
Java compiles into bitcode, that a bitcode interpreter runs. Its kinda
in a state of flux between the two :)
Again, shows his ignorance
Fourth, Scheme and Lisp are very similar to OCaml (LISP family of
languages). Scheme and Lisp are interpreted, while OCaml is compiled.
Actually,
A programming language is one of the languages you wish a program had been
written in when the perl code you've been given to look at got too big and
unmanageable.
If things didnt break no one would pay to fix them, think of the
bright side of perl
Hi, does anyone know of * being used with icecast in any way. Does *
support vorbis at all? can anyone who is working on this give me a
shout.
Shidan
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Yes they do, they've had it since the K6, and compiling with
mmx-support enabled will make a real difference with them as well.
Do AMD processors have MMX as well? I seem to recall that they did, but
can't remember for sure?
Thanks,
Adam
--
--
Adam Goryachev
Website Managers
Ph:
, but
regardless you can go to the skype forums and request it as a beta
tester.
Regards,
Shidan
On Mon, 07 Feb 2005 16:45:53 +0100, HBK [EMAIL PROTECTED] wrote:
Hi
Any solution for connecting Asterisk to Skype without using fsx/fxo
hardware ?
Best Regards
HB
Norway
is becoming the venue for contact as
most Toronto respondents have signed up here. Can I request that the
remainder who responded to the Toronto call signup?
Shidan is the organizer and has proposed a date. I'd hate to confirm it
without the remainder having input - especially those with travel
I'm interested in working on this project, please contact me if you
plan or are actually working on this and we can probably coordinate
something here.
Regards,
Shidan
shidan at gmail
On Fri, 17 Dec 2004 09:05:13 -0500, Noah Miller [EMAIL PROTECTED] wrote:
I wonder if there is an application
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