Dear,
is any solution for replacing .call files into the database?
best
___
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Hi
Can I use Asterisk as an announcement server. We want to build announcement
server with ISDN PRI card terminating on our server and announcement being
fed on the incoming calls.
Also is there any ISDN card available for Laptop.
regards
Sandesh
___
Hi,I need a bit of help regarding setting up asterisk. I am trying to
setup a simple PBX for a small office we have. We just need 4 extensions. I
would like to spent as less as possible.
The below are possible solutions i can think off.
1.) I use a Digium Card and connect my PSTN line to
On Sat, Jul 26, 2008 at 12:50:21AM -0700, Pezhman Lali wrote:
Dear,
is any solution for replacing .call files into the database?
This is not likely to work. It would require Asterisk to constantly poll
the database.
But then again, you can easily originate calls from the manager
interface if
If you want to connect you pstn only, nothing more, and dont forget that FXO
is for the lines. FXS for the Phones
2008/7/26 Preetish Kakkar [EMAIL PROTECTED]
Hi,I need a bit of help regarding setting up asterisk. I am trying to
setup a simple PBX for a small office we have. We just need 4
Yes you can. As for the bri and laptop cards you have xorcom with usb
http://www.xorcom.com/index.php/products/astribank/astribank_models__1/bri_astribank_models
and various bri--sip gates on the market.
2008/7/26 ballamudi madhulika [EMAIL PROTECTED]
Hi
Can I use Asterisk as an announcement
ballamudi madhulika wrote:
Can I use Asterisk as an announcement server. We want to build
announcement server with ISDN PRI card terminating on our server and
announcement being fed on the incoming calls.
With the right dialplan and scripts or AGIs, I don't see why not.
Shouldn't be a
Why not compile from source ?
2008/7/25 Andrew Joakimsen [EMAIL PROTECTED]
Does anyone know who maintains the asterisk packages in the openSUSE
buildservice? They are not updating Zaptel with their kernel updates
and I want to get that matter corrected.
I submitted to them a bug report but
On Sat, Jul 26, 2008 at 02:32:28PM +0200, Grygoriy Dobrovolskyy wrote:
Why not compile from source ?
So it works well as a packagee. So it has proper dependencies on the
kernel package isntalled. So you could upgrade kernel and relatively
easily upgrade Zaptel. So you could use 'rpm -V' to check
On Fri, Jul 25, 2008 at 12:17:55PM -0400, Andrew Joakimsen wrote:
Does anyone know who maintains the asterisk packages in the openSUSE
buildservice? They are not updating Zaptel with their kernel updates
and I want to get that matter corrected.
I submitted to them a bug report but they seem
ballamudi madhulika schrieb:
Can I use Asterisk as an announcement server. We want to build announcement
server with ISDN PRI card terminating on our server and announcement being
fed on the incoming calls.
Yes.
Also is there any ISDN card available for Laptop.
Recently I discovered a cool
I need to have different gain settings on each channel. Is this easy to
achieve?
txgain, rxgain and many other parameters are defined on a per-channel
basis in zapata.conf, they're not global. Each channel definition
channel = x
assumes previous definitions of such parameters.
Lol crackup.
Having said that here is some help.
Don't even think of using a laptop that's just dumb.
Next - check out www.voip-info.org you'll find what you need there.
Regards,
Dean Collins
+1-212-203-4357 (Direct)
+61-2-9016-5642 (Sydney in-dial)
http://www.Cognation.net
-Original
Hi List,
We are trying to make a click 2 call button, we have a PHP script that
passes the 1st phone number of the 1st leg to a manager script, that then
dials the 1st call, then the 2nd call gets placed inside of Asterisk using a
normal dial command. Problem is, we get no status codes, we cannot
Philipp Kempgen schrieb:
ballamudi madhulika schrieb:
Can I use Asterisk as an announcement server. We want to build announcement
server with ISDN PRI card terminating on our server and announcement being
fed on the incoming calls.
Yes.
Also is there any ISDN card available for Laptop.
On Fri, Jul 25, 2008 at 3:12 PM, Joseph L. Casale
[EMAIL PROTECTED] wrote:
The question you haven't answered yet, Joseph, is how does your
Meridian connect to the PSTN?
Is it a T-1 now, or analog?
Sorry Jay,
I ended up in an offline conversation with someone regarding this.
Its on an analogue
On Fri, Jul 25, 2008 at 10:57:49PM +0300, Tzafrir Cohen wrote:
On Fri, Jul 25, 2008 at 01:54:21PM -0400, Jay R. Ashworth wrote:
On Thu, Jul 24, 2008 at 04:35:33PM +0300, Tzafrir Cohen wrote:
What's wrong with plain old Zap/NN ?
[test]
exten =
On Sat, Jul 26, 2008 at 01:09:16AM +0300, Tzafrir Cohen wrote:
On Fri, Jul 25, 2008 at 05:52:20PM -0400, Al Baker wrote:
Quote
seems like a dial-by-span syntax.
What is Dial-by-span ?
Zap/span-num-channel-in-span
Hmmm.
Zap/2 here means the second Zap timeslot on the machine, as does
I am having problems with CME transferring calls that are busy or noan to
voicemail which is on Asterisk. I have used the no supplementary-service sip
moved-temporarily
and no supplementary-service sip refer commands but when an outside call is
transferred to voicemail it just goes to a busy
On Sat, Jul 26, 2008 at 01:14:14PM -0400, Jay R. Ashworth wrote:
On Sat, Jul 26, 2008 at 01:09:16AM +0300, Tzafrir Cohen wrote:
On Fri, Jul 25, 2008 at 05:52:20PM -0400, Al Baker wrote:
Quote
seems like a dial-by-span syntax.
What is Dial-by-span ?
Zap/span-num-channel-in-span
On Sat, Jul 26, 2008 at 01:12:42PM -0400, Jay R. Ashworth wrote:
On Fri, Jul 25, 2008 at 10:57:49PM +0300, Tzafrir Cohen wrote:
On Fri, Jul 25, 2008 at 01:54:21PM -0400, Jay R. Ashworth wrote:
On Thu, Jul 24, 2008 at 04:35:33PM +0300, Tzafrir Cohen wrote:
What's wrong with plain old
But how would my calls be transferred to extension phones from asterisk
server. Would i need to connect those phones to Digium card as well. What i
mean is would digium card have a main extension where i would connect main
pstn line and other 3 port where i would connect another phone line which
Al,
I managed a CCM/Unity setup for more than 5 years and saying that
Asterisk is less reliable is in my experience a stretch. I will agree
that Cisco makes excellent gateways, routers and switches and their
pstn gateways are top notch but I was never a fan of CCM and
especially Unity. We were
On Sat, Jul 26, 2008 at 09:34:29PM +0300, Tzafrir Cohen wrote:
On Sat, Jul 26, 2008 at 01:12:42PM -0400, Jay R. Ashworth wrote:
On Fri, Jul 25, 2008 at 10:57:49PM +0300, Tzafrir Cohen wrote:
On Fri, Jul 25, 2008 at 01:54:21PM -0400, Jay R. Ashworth wrote:
On Thu, Jul 24, 2008 at
On Sat, Jul 26, 2008 at 09:32:03PM +0300, Tzafrir Cohen wrote:
Zap/2 here means the second Zap timeslot on the machine, as does
Zap/2-1, using all PRI's on Digium and Sangoma cards.
I would have *expected* that it might behave the way you suggest, but
it appears not to. Unless it has
On Sat, Jul 26, 2008 at 03:28:10PM -0400, Jay R. Ashworth wrote:
On Sat, Jul 26, 2008 at 09:34:29PM +0300, Tzafrir Cohen wrote:
On Sat, Jul 26, 2008 at 01:12:42PM -0400, Jay R. Ashworth wrote:
On Fri, Jul 25, 2008 at 10:57:49PM +0300, Tzafrir Cohen wrote:
On Fri, Jul 25, 2008 at
On Sat, Jul 26, 2008 at 10:38:07PM +0300, Tzafrir Cohen wrote:
[ quoting me ]
Chanunavail/Congestion.
Here, let me go get the exact message...
==88
-- Executing AGI(SIP/101cathy-b7619990,
On Sat, Jul 26, 2008 at 03:32:34PM -0400, Jay R. Ashworth wrote:
On Sat, Jul 26, 2008 at 09:32:03PM +0300, Tzafrir Cohen wrote:
Zap/2 here means the second Zap timeslot on the machine, as does
Zap/2-1, using all PRI's on Digium and Sangoma cards.
I would have *expected* that it might
On Sat, Jul 26, 2008 at 03:46:28PM -0400, Jay R. Ashworth wrote:
On Sat, Jul 26, 2008 at 10:38:07PM +0300, Tzafrir Cohen wrote:
[ quoting me ]
Chanunavail/Congestion.
Here, let me go get the exact message...
==88
Preetish Kakkar wrote:
But how would my calls be transferred to extension phones from
asterisk server. Would i need to connect those phones to Digium card
as well. What i mean is would digium card have a main extension where
i would connect main pstn line and other 3 port where i would
On Sat, Jul 26, 2008 at 10:48:54PM +0300, Tzafrir Cohen wrote:
Except that that is what Asterisk is giving *us*:
-- Local/[EMAIL PROTECTED],1 answered Zap/73-1
-- IAX2/VICIast26-19 answered Zap/73-1
-- Zap/11-1 is ringing
-- Zap/11-1 answered SIP/101cathy-0824cda0
On Sat, Jul 26, 2008 at 3:53 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Sat, Jul 26, 2008 at 03:46:28PM -0400, Jay R. Ashworth wrote:
On Sat, Jul 26, 2008 at 10:38:07PM +0300, Tzafrir Cohen wrote:
[ quoting me ]
Chanunavail/Congestion.
Here, let me go get the exact message...
I just stumbled across this on youtube.
Does any on the list us it? This is the first I've heard over it.
http://www.youtube.com/watch?v=H1j5OrgL1og
Regards,
Dean Collins
[EMAIL PROTECTED]
+1-212-203-4357 (New York)
+61-2-9016-5642 (Sydney)
http://www.Cognation.net
On Sat, Jul 26, 2008 at 11:40 AM, Preetish Kakkar
[EMAIL PROTECTED] wrote:
But how would my calls be transferred to extension phones from asterisk
server. Would i need to connect those phones to Digium card as well. What i
mean is would digium card have a main extension where i would connect
The something is generated by Asterisk at the time the call is
created. You should never add it, since you don't control that call
instance info. In fact, you should almost never care about the call
instance string. The -1 means first instance of a call on this
channel, a -2 would be seen
On Sat, Jul 26, 2008 at 6:12 PM, randulo [EMAIL PROTECTED] wrote:
On Sat, Jul 26, 2008 at 11:40 AM, Preetish Kakkar
[EMAIL PROTECTED] wrote:
But how would my calls be transferred to extension phones from asterisk
server. Would i need to connect those phones to Digium card as well. What i
mean
On Sat, Jul 26, 2008 at 7:05 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
On Sat, Jul 26, 2008 at 6:12 PM, randulo [EMAIL PROTECTED] wrote:
On Sat, Jul 26, 2008 at 11:40 AM, Preetish Kakkar
[EMAIL PROTECTED] wrote:
But how would my calls be transferred to extension phones from asterisk
server.
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