date:20061004

[asterisk-users] Call Forwarding not working for extension in queue, why?

2006-10-04 Thread Zeeshan Zakaria

Extension 200 is member of a queue. At night time, it is forwarded to a different number. Now when this extension is dialed directly, call forwarding works, but when a call comes into the queue, ext. 200 keeps on ringing and doesn't get forwarded. Why is that and how to fix it?
-- Zeeshan A Zakaria 
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Re: [asterisk-users] Screen pop based on incoming DID

2006-10-04 Thread Wolfgang Lumpp

Hi,

Am Dienstag, 3. Oktober 2006 14:43 schrieb Greg Delgado:
 I want to pop up a web page when a queue member phone
 rings but, instead of displaying the clid, I want to
 display the DID number the call came in. Any ideas how
 to best implement this?
you can try FOP with such an entry in your extensions.conf
exten = 1,2,UserEvent(FOP_Popup|
URL:testpage.php?clid=${CALLERIDNUM}^Target:temp)
You can set any variable to parse by testpage.php

Regards
Wolfgang
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[asterisk-users] Re: extensions.conf strangeness

2006-10-04 Thread Martin Joseph


On 2006-10-02 13:55:15 -0700, Brian Candler [EMAIL PROTECTED] said:


On Sun, Oct 01, 2006 at 03:55:45PM +0100, Marco Mouta wrote:


[invalid]

exten = _X!,1,Answer()
exten = _X!,2,Background(pbx-invalid)

Are you sure that your invalid context is correctly written?
I've never heard about this pattern match _X!
As far as i know the wild card is the .
So your invalid context should be:
[invalid]
exten = _X.,1,Answer()
exten = _X.,2,Background(pbx-invalid)
This may be the cause


_X! means match the pattern as soon as it possibly could. If you use _X.
then a timeout has to take place to see whether some other pattern might
match.

But your explanation still doesn't go into why it works differently in one
context than another. I guess I'm going to have to assume that Asterisk
dialplans are non-deterministic :-(

Are there any debug tools which can show the thought process as a
dial-plan is processed - for example, what patterns are tried and in what
order?


You can say show dialplan from the command line...

Don't know if this helps?

Marty


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[asterisk-users] DISA and legacy PBX

2006-10-04 Thread James Harper

I've configured our PBX so that when a user dials 80 on the PBX
extension, it goes out an ISDN TE interface on the PBX and into an NT
interface on my asterisk machine, where it jumps into the 's' extension.

Asterisk then does a DISA(no-password|sip_provider_out) which allows the
call to go out via a sip provider, to give us cheaper calls.
Unfortunately if the user doesn't wait for DISA to give dialtone,
asterisk doesn't hear all of the digits.

When they dial '0' on the PBX there is no need to wait for dialtone, so
it is a bit confusing for the users.

Any suggestions? I'm using misdn, and 'immediate' and 'always_immediate'
in the config for that port, so maybe there is something I could do
there to take whatever digits have been dialled so far...

Thanks

James
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[asterisk-users] IVR for the called part (IVR inside out)

2006-10-04 Thread Daniel Cyt


Hello,

I'm trying to get it to work but I can't find the right way. I would be glad 
if the list could point me the right directions.


What I want: My Asterisk dialing out to a number (my mobile phone) and 
playing an IVR to the called part saying press one to accept this call. If 
the called part (my mobile) press 1 the call goes thru, otherwise it goes 
straight to asterisk voicemail.


Reason (my scenario): I'm going to setup a follow me from my extension to my 
mobile phone and I don't want people to find out they are actually rining on 
my mobile. I don't have the option to disable voicemail feature on the 
mobile company. The problem happens if I don't pick the call or I'm, for 
instance inside a tunnel, where my mobile lose signal. Asterisk will think 
my mobile voicemail is somebody answering, and whoever called me will her 
the mobile voicemail.


I've been searching for a while before emaiil the list but I could not find 
anything like it.


Thank you very much

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http://messenger.msn.com.br


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Re: [asterisk-users] Call Forwarding not working for extension in queue, why?

2006-10-04 Thread Dovid B




What do you mean by that it is forwarded. Is it set 
on the phone or do you have it set in que memeber.

  - Original Message - 
  From: 
  Zeeshan 
  Zakaria 
  To: Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Wednesday, October 04, 2006 8:56 
  AM
  Subject: [asterisk-users] Call Forwarding 
  not working for extension in queue, why?
  Extension 200 is member of a queue. At night time, it is 
  forwarded to a different number. Now when this extension is dialed directly, 
  call forwarding works, but when a call comes into the queue, ext. 200 keeps on 
  ringing and doesn't get forwarded. Why is that and how to fix it? -- Zeeshan A Zakaria 
  
  

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Re: [asterisk-users] t1 voip to failover pri

2006-10-04 Thread adebayo omo-dare

Yes SDSL lines do hook up to the DSLAM. I do not know if the DSLAM itselfis a fear for you, but if it is, you can look at it in this way -All it does is aggregate signals for transmission through a switch and on to a high speed backbone. This, invariably, at some point or another, is the same backbone used to transfer all traffic. With the way things are/are going, at some point, everything, in anyway related to the loop,hooks up toa common point, in some way or another. Possibly you could be more specific about your concern.With telecom companies current strategies, (and here I am thinking most especially of MPLS, Metro Ethernet, etc), each technology is being hardened in order to meet common requirements all based aroundthedelivery of tripple play over a common format (IP).So the fears you may have, may not necessarily be those that I see.
As I am sure you already know, this is an extremely wide and complex environment when you want to get things right. It is also very difficult to speak about without knowing exactly what it is you are seeking to get right in an environment where there is always a solution to every problem and always a problem in every solution.I hope the above does in some manner help and all the best.Bayo stan ford [EMAIL PROTECTED] wrote:if i went with an SDSL line, don't those lines hook up to a common point, the DSLAM?i do like this idea of faling over not to a pri but another cheaper high speed line.adebayo omo-dare [EMAIL PROTECTED] wrote:I wouldn't first presume that there is any law that states you have to run VoIP over a PRI. Other technologies exist such as SDSL - with some providers speaking of crazy prices such as £65pcm for2Mbs5:1 contention ration and £100 for 1:1 (U.K) - should be even less in the states if that's where you are. One option, but not the only one, would be to drop your pri when your contract ends and take up SDSL - and voila an initial saving, inyour case,of a 000 or more in the year.You could also have two SDSL lines for a little less than the price of the PRI. Both lineswould not only serve for High Availability -possibly even better availabilitythan single PRI- but could also, actively, both switch traffic, giving you 4Mbps of
bandwidth for your VoIP, or if you choose, some other requirement while not required as failover- all for the price of less than one PRI.Then there is compression - 64k non negotiable, per channel forPRI, and flexible -i.e., less the 64k- for VoIP (International high quality Calls are transported at 16k),giving you the capacity to potentially service more traffic with less initial outlay.Other real cost efficiencies come in the form of the fact that IP-to-IP (local/national/international) calls are free. So if you have a lot of inter-branch communications, or communications you can switch on to IP,you can totally erradicate this cost - unlike with the PRI where you will still be subject to payment. Think like this - say I have two offices - one in london and the other New York. How much will I save by moving my calls on to VoIP with no per-time or call setup
charges.Features related to OAMP,can also be faster and cheaper with you having a lot more power in your hands.In real senses, and with regards to reliability, you should take in to consideration the great moves currently being made by telecom companies (incumbents most especially), with regards to a complete shift to NGNs, which have a strong focus on ToIP. With new fiber (FTTP), new technology, etc, a lot of networks are highly reliable at the present moment - I guess this would also depend on where you are.The thing about it is that complete IP networks in terms of telecom now look inevitable. And whether you do it yourself or it is done for you - it is the way things, many expect, are going to be in the next 5 or so years. stan ford [EMAIL PROTECTED] wrote:I'm confused with something, maybe someone can explain to me. if your currently on a pri and are considering moving over to VOIP, that means you would have to purchase a t1 or fractional t1 for a your voip connections. but then, voip connections aren't as reliable as PRI. so then you would probbaly have to get a PRI failover. but then having a PRI failover means that you now have to pay 400 for a T1, then another 400 for your PRI line. wouldn't have you have just defeated the cause of savig money by now having to havea PRI on standby? now costing you 800 a month? wouldn't it almost be the same price to stick with the PRI only?is anyone out there, using a VOIP only with no failover?
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