Thank you all,
It just so turns out that it was a bad zaptel module. We saw
another post on digiums site where someone was having the exact same
problem with several versions of zaptel. We changed to the one that he
said worked (1.2.21), and all is well now. (And asterisk is now parsing
the
think you can get ANI on EM Wink trunks, how about feature group d?
-Jon
- Original Message -
From: Dan Casey [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, November 02, 2007 9:47 AM
Subject: [asterisk-users] Route an incoming call by ANI*DNIS
does anyone
does anyone know how to route a call coming in with ANI*DNIS*
Oct 31 12:47:41 VERBOSE[30056] logger.c: -- Executing
Set(Zap/49-1, DID=1231234*4812*) in new stack
I tried making a route for _.*4812* but that matched everything rather
then just the dnis i wanted.. any ideas?
I would
wrote:
Dan Casey wrote:
does anyone know how to route a call coming in with ANI*DNIS*
Oct 31 12:47:41 VERBOSE[30056] logger.c: -- Executing
Set(Zap/49-1, DID=1231234*4812*) in new stack
I tried making a route for _.*4812* but that matched everything rather
then just the dnis i wanted
Hello,
I am replacing an exisiting call center with a new asterisk based
solution. This will initially consist of to phone servers. The first
being the main PBX, and the second being a predictive dialer. The
dialer will have sip extensions for all the agents, while the main pbx
will hand
Is there a way to hang up on a sip channel. One of my phones is saying
it's busy while it's not (even after rebooting it).
I logged into asterisk, and did a sip show channel 232, and sure enough
it thinks it's on a call.
How can I force it to close?
I tried to do sip over vpn with with a linksys router handle just one
phone. When I tried it, it worked fine. Once i shipped it out we had
all types of problems.
at first it was fine, then 1 out of 5 calls would sound like cell
phones. Now I can call him be he can't hear anything. Everything
I am trying to have a SIP extension that will dial an outside phone
number (ie: cell phone) using a zap channel.
I am using the following hack, which doesn't technically works, but not
nicely. What i want to do is have it pick an available trunk from zap1
to zap20.
I have tried using
.
--
--
Steven
http://www.glimasoutheast.org
"Dan Casey" [EMAIL PROTECTED]
wrote in message news:[EMAIL PROTECTED]...
I actually have it semi-working. My trunks were set up improperly.
Now i can do it, but only if i specify a specific zap channel.
exten = 299,1,Mac
I gave that a try but had no luck. I keep getting all circuits busy.
Perhaps there is another way.
I think it is having trouble when transfering zap to zap.
but no matter what i do i can't get it.
I made a sip number to try from, but its not working
[ext-local-custom]
;exten =
I actually have it semi-working. My trunks were set up improperly.
Now i can do it, but only if i specify a specific zap channel.
exten = 299,1,Macro(dialout,2,1914304,,)
the 2 takes me to zap 1. I tried to replace that with "s" but no luck..
any idea how to do this where it will pick
Greg Broiles wrote:
On 8/3/06, Dan Casey [EMAIL PROTECTED] wrote:
I actually have it semi-working. My trunks were set up improperly.
Now i can do it, but only if i specify a specific zap channel.
exten = 299,1,Macro(dialout,2,1914304,,)
the 2 takes me to zap 1. I tried to replace
Using [EMAIL PROTECTED] 1.1 and 1.3
ip phones are on the same network as asterisk. I can call another
extension and talk/listen w/ no problem.
If i dial my own extension (or do anything that makes asterisk play
sounds back to me ) I cannot hear anything.
this is what shows up in the CLI, when
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