[asterisk-users] Lucent TNT Help

2006-11-02 Thread Corey Frang
I'm looking for someone familiar with setting up some of the more 
advanced features of the Lucent TNT, preferably someone with knowledge 
of Trunk Groups and choosing outgoing PRI channels based on call type 
and perhaps NPA-NXX


We currently have 8 PRI's.  7 of them are for our dialup pool, the 8th 
is for our voip. We currently run the dialup PRI's to a seperate TNT


We want to merge these all on to one TNT.

I found out how to do dnis-or-voip for the call type on the voip line 
which allows me to set based on dialed number if its going to go to a 
modem or voip call, however, I'm trying to figure out
how to set up the TNT to have voip origination use a certain PRI in the 
pool as the primary and then fail over to the other PRI's.  I think it 
will probally involve setting up trunk groups, but I'm not entirely sure 
how I would set the trunk group for origination. Can anyone give me some 
friendly advice to try to figure this out?


-Corey
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] Asterisk X100P - Interrupt a call?

2006-05-15 Thread Corey Frang
So, We want to be able to put a fax machine on the line port of the 
X100P in our asterisk server.  We however also want to use this card for 
911 calling.  We need some sort of mechanisim to "disable" the line out 
port on the x100p by software to "interrupt a call" on the line.


Anyone done anything like this?
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] ACD calls to busy agents

2005-10-17 Thread Corey Frang




PauseQueueMember works fine, they just never get unpaused if they hang
up the call...

BJ Weschke wrote:
You must specify to pqm and upqm what interface you're
attempting to pause and unpause. It will not figure that out based on
what interface the channel that called the app in the dial plan might
have used.
  
  On 10/17/05, Corey Frang <[EMAIL PROTECTED]> wrote:
  So,
I'm looking into using PauseQueueMember and unpause queuemember

How the heck to you get Unpause to run, no matter what, after the call

is over?

The "g" argument to Dial only works when the >called< party hangs
up.

Using the "h" extension appears to be doing nothing...

Is there any way we could add a feature to the "pausequeuemember" that

basically says "As long as this channel is open, this member is paused"
so  that way when they hang up they are unpaused automatically?

Julian Lyndon-Smith wrote:

> Have you tried the "PauseQueueMember" application in the dialplan
?

>
> If the agent makes an outbound call, before the dial() call
> PauseQueuemember - and UnPauseQueuemember when the call is
complete.
> The system should not then send any agent calls through, but all
other

> calls (direct / internal) should come through.
>
> This is in 1.2b1 and CVS-HEAD.
>
> HTH
>
> Julian.
>
> Tom Rymes wrote:
>
>> I don't know how to make this happen, and I don't even think
it is

>> really possible given the current Queue app, but this would be
a
>> very  nice feature to have. The queue shouldn't pass a call to
an
>> agent if  they are already on a call from the queue, but an
incoming

>> call from  another internal extension, or even a DID ought to
be able
>> to get  through.
>>
>> Consider this a feature request?
>>
>> Tom
>>
>> On Oct 15, 2005, at 10:04 PM, J Thomas wrote:

>>
>>> One of my friends is facing this problems and I could not
find any
>>> solution to that. Hence this post.
>>>
>>> In her Asterisk PBX, she has programmed about 10 agents,
and

>>> strategy is
>>> rrmemory. Everything works fine. When an agent has
received an ACD
>>> call,
>>> another call is not presented to him as long as he is on
the ACD call.
>>>
>>> However when an agent has made an outgoing call, he is
still presented
>>> another ACD call when his turn comes. This results in
unnecessary
>>> delay
>>> in answering that call.

>>>
>>> Taking out call waiting is not an option, as an agent can
also get a
>>> direct dialed call, and he should be able to pick up that
call even
>>> when
>>> he is on another call.

>>>
>>> Is there a way so that a busy agent (whether busy because
of an
>>> incoming
>>> call, or outgoing call) is not presented another ACD call?
>>>
>>> Thanks,

>>> -- jt
>>>
>>> ___
>>> --Bandwidth and Colocation sponsored by Easynews.com --
>>>

>>> Asterisk-Users mailing list
>>> Asterisk-Users@lists.digium.com
>>> 
http://lists.digium.com/mailman/listinfo/asterisk-users
>>> To UNSUBSCRIBE or update options visit:
>>>http://lists.digium.com/mailman/listinfo/asterisk-users

>>>
>>
>> ___
>> --Bandwidth and Colocation sponsored by Easynews.com --
>>
>> Asterisk-Users mailing list

>> Asterisk-Users@lists.digium.com
>> http://lists.digium.com/mailman/listinfo/asterisk-users

>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
>
> ___
> --Bandwidth and Colocation sponsored by Easynews.com --
>
> Asterisk-Users mailing list
> 
Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>


--
Rock River InternetCorey Frang

202 W. State St, 8th Floor [EMAIL PROTECTED]
Rockford, IL 61101  815-968-9888 Ext. 2205
USA   fax 968-6888


___
--Bandwidth and Colocation sponsored by Easynews.com
--

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  

Re: [Asterisk-Users] ACD calls to busy agents

2005-10-17 Thread Corey Frang

So, I'm looking into using PauseQueueMember and unpause queuemember

How the heck to you get Unpause to run, no matter what, after the call 
is over?


The "g" argument to Dial only works when the >called< party hangs up.

Using the "h" extension appears to be doing nothing...

Is there any way we could add a feature to the "pausequeuemember" that 
basically says "As long as this channel is open, this member is paused" 
so  that way when they hang up they are unpaused automatically?


Julian Lyndon-Smith wrote:


Have you tried the "PauseQueueMember" application in the dialplan ?

If the agent makes an outbound call, before the dial() call 
PauseQueuemember - and UnPauseQueuemember when the call is complete. 
The system should not then send any agent calls through, but all other 
calls (direct / internal) should come through.


This is in 1.2b1 and CVS-HEAD.

HTH

Julian.

Tom Rymes wrote:

I don't know how to make this happen, and I don't even think it is  
really possible given the current Queue app, but this would be a 
very  nice feature to have. The queue shouldn't pass a call to an 
agent if  they are already on a call from the queue, but an incoming 
call from  another internal extension, or even a DID ought to be able 
to get  through.


Consider this a feature request?

Tom

On Oct 15, 2005, at 10:04 PM, J Thomas wrote:


One of my friends is facing this problems and I could not find any
solution to that. Hence this post.

In her Asterisk PBX, she has programmed about 10 agents, and  
strategy is
rrmemory. Everything works fine. When an agent has received an ACD  
call,

another call is not presented to him as long as he is on the ACD call.

However when an agent has made an outgoing call, he is still presented
another ACD call when his turn comes. This results in unnecessary  
delay

in answering that call.

Taking out call waiting is not an option, as an agent can also get a
direct dialed call, and he should be able to pick up that call even  
when

he is on another call.

Is there a way so that a busy agent (whether busy because of an  
incoming

call, or outgoing call) is not presented another ACD call?

Thanks,
-- jt

___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users



___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users




___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users






--
Rock River InternetCorey Frang
202 W. State St, 8th Floor [EMAIL PROTECTED]
Rockford, IL 61101  815-968-9888 Ext. 2205
USA   fax 968-6888

___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] Dial DTMF after bridging call

2005-10-12 Thread Corey Frang
I don't think that its the D() dialing before the call is bridged, I 
just tested it on Asterisk 1.0.7 and CVS HEAD


Both times I did:

Dial(SIP/[EMAIL PROTECTED],20,D(ww1234))

both times i picked up the phone, it waited about 1 second, dialed 1, 
then stopped alltogether.


This might be an actual bug...

Interestingly on CVS HEAD i tried

Dial(SIP/[EMAIL PROTECTED],20,D(ww1234:ww1234))

I heard the DTMF's on the calling phone... I'm wondering if there is 
some issue with how its writing the DTMF to the outgoing SIP channels?


The lucentbox is a MaxTNT.

Interestingly, I started playing with the numbers on my phone after the 
dial messed up, and I could get the DTMF tones "stuck" playing one tone 
for a long time.  If i took the D() out of it It didn't have that 
problem.


On Aug 25, 2005, at 15:04, Joseph wrote:


Is there a way to dial DTMF after bridging the call.
The current option D() in Dial will dial DTMF before the call is 
bridged

and this doesn't do the job.
I need to dial DTMF after the call is bridged and the message is played
with "Background"

--
#Joseph
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users



___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users