On 2007-02-22 04:22:20 -0800, Frederico Madeira [EMAIL PROTECTED] said:
Hi guys,
My asterisk is show me some errors on line registration.
This message appear on console: Request to schedule in the past?!?!
What it mean ?
Thanks.
I see this message all the time on my lowely powerPC mac
Hi,
For a customer, I am looking for a good and reliable Asterisk based system.
Five servers will be installed at different locations and will be linked
together with each other. This system will work as a call center as well. It
has to be a stable and reliable. Customer also needs GUIs for
Ok, thank you very much.
I've just installed on the pc the trixbox Linux distribution.
The PBX now is ready?
I think that i have only to configure the other lans' pcs, didn't i?
Please answer me...
Have a nice day
Andrea
2007/2/24, Tzafrir Cohen [EMAIL PROTECTED]:
On Sat, Feb 24, 2007 at
You could set a dialplan variable in the AGI so that it's pretty easy to
tell what happened in the AGI.
About the code 0, the funny part is that you see AGI Script completed,
returning 0 even if the AGI does not exist, or is not executable. This
should be a good candidate for improvement
hi,
i was able to get this working with google talk.
i entered [EMAIL PROTECTED] using the gtalk2voip.com website's invite
box, and as a result, saw a request from [EMAIL PROTECTED] to be added
as a buddy in my google talk contact list. i accepted the request.
in my asterisk dialplan, i have
Zoa wrote:
So does asterisk (Albeit with a commercial package)
http://www.attractel.com/t38.html
I'll do it for 30% less than they quote. :-)
Lee Howard wrote:
Matt Riddell [NZ] wrote:
Does OpenPBX do a T.38 gateway then?
Yes, it does.
Lee.
Steve
On Fri, 2 Mar 2007, Al Bochter wrote:
I don't see why the cost to send SMS is around .15 each. What does the
gateway know that I don't know about sending the SMS.
I just think .15 for each SMS send is high. Or am I just over looking
something?
You're missing nothing; The telcos have us by
The best option would be to use AstManProxy and connect your event manager
to it.
Jesus
On 3/2/07, Doug Garstang [EMAIL PROTECTED] wrote:
Ok, so I ain't much of a Java programmer, but...
Can the Asterisk Java API be written with threads? Ie, I need to connect
to multiple Asterisk systems
On Sat, Mar 03, 2007 at 12:01:58PM +, Gordon Henderson wrote:
You're missing nothing; The telcos have us by the short curlys. For
them, it's money for old rope. They probably (in the UK at least) make
many times more money through TXT messages than voice. The base rate
here is about
Gordon Henderson wrote:
On Fri, 2 Mar 2007, Al Bochter wrote:
I don't see why the cost to send SMS is around .15 each. What does
the gateway know that I don't know about sending the SMS.
I just think .15 for each SMS send is high. Or am I just over
looking something?
You're missing
Jesus Mogollon wrote:
The best option would be to use AstManProxy and connect your event
manager to it.
why would adding a new system in between be better than directly
connecting to multiple Asterisk servers?
=Stefan
--
reuter network consulting
Neusser Str. 110
50760 Koeln
Germany
Telefon:
Good Idea, but when the user has to do nothing is better for my users!
Thanks
JAson
Mojo with Horan Company, LLC wrote:
Another option is to have the user hit the forward button on their
phone and manually type in their cellphone number when they're going
to be out of the office.
Jason
From what i was told via the GT guys at bell
The customer should send
I 1C Standard Facility Length = 21
9F Serv Discrim Networking Extensions
PDU Component Begins (hex)
8B0100A10F020101...
instead of:
I 28 Display Length = 9
E Display .JIMBOB
so we
On Sat, Mar 03, 2007 at 11:52:38AM -0500, Mike Lynchfield wrote:
From what i was told via the GT guys at bell
The customer should send
I 1C Standard Facility Length = 21
9F Serv Discrim Networking Extensions
PDU Component Begins (hex)
8B0100A10F020101...
I'll second that,
CentOS 4.4 + FreePBX 2.1.3 + Asterisk 1.2.13 + Sangoma A200D + Aastra
9133i's running 4 months without a reboot and no memory leaks fielding about
150 calls a day. Everyone loves the system. These are normal users used to
tradtitional phone systems.
I would not go as far as
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On 3 Mar 2007, at 15:02, Steve Totaro wrote:
Text messaging is not that big in the US for some reason. Well
anyways, on my T-Mobile phone, I have an unlimited text message
package that cost $15/mo. I am not sure how many constitutes
hey sorry for caps my bad.
and yes i will , sub to -dev however i tought that for that 1 post i
shouldn't go trough the troubles of doing that. and that someone could
already be on the -dev form this thread ,
i also assumed threads where like sub channels where people could isolate
from the
Thank you.
Please help me with two more issues.
tzieleniewski wrote:
1. In many docs on the web there is an info to make asterisk by invoking
#make mpg123 in order to have the mp3 support, when I try to do it this way
make informs that there are no such make rules.
Those documents are
On Sat, Mar 03, 2007 at 06:24:29PM +0100, tzieleniewski wrote:
Thank you.
Please help me with two more issues.
tzieleniewski wrote:
right now I don't have any digium cards installed in my PC so the only
thing I need at the moment is the ztdummy module for the purpose of
timer
Can one do an in-place update to 1.4.1 from 1.4.0 ? (Just compile and
install)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Asterisk
Development Team
Sent: Friday, March 02, 2007 7:04 PM
To: undisclosed-recipients:
Subject: [asterisk-users] Asterisk
Wanting to connect my asterisk box off of 2 unused analog extensions on
the non* PBX system.
Can I bring those lines in to * on XP100 FXO cards?
Any special wiring/open loop issues to watch for?
The desired config is inbound PRI - non* PBX, attendant picks up and if
caller wants services on
Hi,
I'm writing asterisk application in C language. I need to know what is
state of my asterisk user, so I have found command: ast_device_state(data);
. So if my IP phone is reachable I get status 1
From: Joseph [EMAIL PROTECTED]
Date: Sat, 03 Mar 2007 00:30:42 -0700
I would like to implement enum lookup in my dial plan but searching for
solution / implementation I'm getting confused what is current
standard.
On some pages I read that the ENUMLOOKUP is not in development anymore
and
From: Lenz [EMAIL PROTECTED]
Date: Sat, 03 Mar 2007 11:37:29 +0100
You could set a dialplan variable in the AGI so that it's pretty easy to
tell what happened in the AGI.
About the code 0, the funny part is that you see AGI Script completed,
returning 0 even if the AGI does not exist, or is
On 3/3/07, Mike D'Ambrogia [EMAIL PROTECTED] wrote:
Wanting to connect my asterisk box off of 2 unused analog extensions on the
non* PBX system.
Sounds workable.
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On Sat, 3 Mar 2007, Mike D'Ambrogia wrote:
Wanting to connect my asterisk box off of 2 unused analog extensions on
the non* PBX system.
Can I bring those lines in to * on XP100 FXO cards?
Any special wiring/open loop issues to watch for?
You might be better off with one TDM400P card with the
Jason Walker wrote:
I have users in my dialplan that go from SIP to Cell
When they are at their desk and they hit reject call, it goes to the
next thing in the dialplan, thus transferring to their cell. Not what
they want. Is it possible to change the reject button to make it go to
voice
Kevin P. Fleming wrote:
Mike Lynchfield wrote:
Please note that we are available to fix the current REMOTE crash that
affects Asterisk/openpbx/trixbox and crashes these systems via a
malformed packet
please contacts use if you need a hand to patch your systems.
This list is for
Michelle Dupuis wrote:
Can one do an in-place update to 1.4.1 from 1.4.0 ? (Just compile and
install)
./configure
make update worked here.
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asterisk-users mailing list
To UNSUBSCRIBE or
D
Not sure why this works
exten = _3665[0-9],1,goto(test|${EXTEN}|1)
but this does not.
exten = _366[50-59],1,goto(test|${EXTEN}|1)
I would like to route 36650 - 36700 to a Context 'test' however I'm only
able to get 10 to work at a time. Any ideas?
Any help would be great!
Lacy Moore - Aspendora wrote:
On 3/3/07, Mike D'Ambrogia [EMAIL PROTECTED] wrote:
Wanting to connect my asterisk box off of 2 unused analog extensions
on the
non* PBX system.
Sounds workable.
Generally if you do this Asterisk will have a tough time knowing when to
hang up the line.
Hall, Eric M. wrote:
Not sure why this works
exten = _3665[0-9],1,goto(test|${EXTEN}|1)
but this does not.
exten = _366[50-59],1,goto(test|${EXTEN}|1)
I would like to route 36650 – 36700 to a Context ‘test’ however I’m only
able to get 10 to work at a time. Any ideas?
The square brackets
The second example is for a four digit extension. while the first is
for a five digit extension. everything within the brackets is meant
for just one digit.
On 3/3/07, Hall, Eric M. [EMAIL PROTECTED] wrote:
D
Not sure why this works
exten = _3665[0-9],1,goto(test|${EXTEN}|1)
but this
Mike there aint such thing as outbound caller id name that you send
and a pots subscriber sees. so drop it.
On 3/3/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Sat, Mar 03, 2007 at 11:52:38AM -0500, Mike Lynchfield wrote:
From what i was told via the GT guys at bell
The customer should send
Thanks I was just about to say this. You CAN'T send caller-id-name. To be
able to set name you need to set it with Telcordia or whomever manages
numbers in your country.
On 3/3/07, C F [EMAIL PROTECTED] wrote:
Mike there aint such thing as outbound caller id name that you send
and a pots
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