date:20050430

I would know if they real-time charged for that... Although I normally only
have 10-12 calls going, I watch pretty close and dispute any
supposed-to-be-free-but-not calls!

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of trixter
http://www.0xdecafbad.com
Sent: Sunday, May 01, 2005 1:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Broadvoice limits???

On Sat, 2005-04-30 at 23:19 -0700, Kerry Garrison wrote:
> There is no physical limitation that I am aware of right now. Be sure
> and check you end user agreement but I think its pretty vauge. They
> told me once "its one call per account" but when I mentioned call
> waiting they said "ok well two calls". We have actually tested it with
> four to see if there was a limit.
>  

pay particular attention to section 1.3.1 with broadvoice.  I however
accidentally looped my dialplan - had stanaphone (free NY number) used
the whole number including the '1' as its extension.  Set up the
broadvoice dialplan (which what is on broadvoices page seems
imncomplete, never checked to see what else is unlimited but missing but
found one country not there) which defaults to 1NX to go through
broadvoice.  Called my stanaphone number to test, when the call came it
it dialed my stanaphone number. 

Stanaphone didnt like this (and shut my free account off, I got a new
onewith a better number so oh well : but broadovice did not care.  I had
20-30 ports in use at the same time.

One thing broadvoice said in a news article a while back was that they
charge 3.9 cents/min for people who make calls at the same time,
excluding 3-way.  When I did the 30 call loop they didnt charge me
extra, I *think* this only applies if you have multiple IPs register and
place calls at the same time (they do allow multiple IPs to register
according to that same article and all will ring when a call comes in or
something).

I dont have a cite for that article I found it on google, I think it was
a voxilla article though, but it was a while back, and since it was not
an article from today they may have changed their policies onthe 3.9
cents/min.

-- 
Trixter http://www.0xdecafbad.com
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378

___
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] Broadvoice limits???









Broadvoice…

Seems to be no limit on inbound, but I
found any channels after 5 outbounds would get an immediate disco. Guess I’ll
have to stick to Vonage to blast into the local radio shows.  Or maybe 5
on BV, 5 on Vonage, and X on the PRI…

 

 









From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kerry Garrison
Sent: Sunday, May 01, 2005 1:20 AM
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Subject: RE: [Asterisk-Users]
Broadvoice limits???



 

There is no physical limitation that I am
aware of right now. Be sure and check you end user agreement but I think its
pretty vauge. They told me once "its one call per account" but when I
mentioned call waiting they said "ok well two calls". We have
actually tested it with four to see if there was a limit.

 

 







From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Tim Connolly
Sent: Saturday, April 30, 2005
11:09 PM
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Subject: [Asterisk-Users]
Broadvoice limits???

   
Where are the limitations on the Broadvoice service? I saw a mention on the
list saying two inbound/outbound calls, and Kerry just mentioned it during the
radio interview… I don’t see that 2 call limitation with my BYOD
World Plus account. Am I lucky, or just missing where the limitation is?






___
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] Broadvoice limits???

2005-04-30 Thread trixter http://www.0xdecafbad.com

On Sat, 2005-04-30 at 23:19 -0700, Kerry Garrison wrote:
> There is no physical limitation that I am aware of right now. Be sure
> and check you end user agreement but I think its pretty vauge. They
> told me once "its one call per account" but when I mentioned call
> waiting they said "ok well two calls". We have actually tested it with
> four to see if there was a limit.
>  

pay particular attention to section 1.3.1 with broadvoice.  I however
accidentally looped my dialplan - had stanaphone (free NY number) used
the whole number including the '1' as its extension.  Set up the
broadvoice dialplan (which what is on broadvoices page seems
imncomplete, never checked to see what else is unlimited but missing but
found one country not there) which defaults to 1NX to go through
broadvoice.  Called my stanaphone number to test, when the call came it
it dialed my stanaphone number. 

Stanaphone didnt like this (and shut my free account off, I got a new
onewith a better number so oh well : but broadovice did not care.  I had
20-30 ports in use at the same time.

One thing broadvoice said in a news article a while back was that they
charge 3.9 cents/min for people who make calls at the same time,
excluding 3-way.  When I did the 30 call loop they didnt charge me
extra, I *think* this only applies if you have multiple IPs register and
place calls at the same time (they do allow multiple IPs to register
according to that same article and all will ring when a call comes in or
something).

I dont have a cite for that article I found it on google, I think it was
a voxilla article though, but it was a while back, and since it was not
an article from today they may have changed their policies onthe 3.9
cents/min.

-- 
Trixter http://www.0xdecafbad.com
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378

signature.asc
Description: This is a digitally signed message part
___
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] T1 Technology and VoIP Gateway Primer

2005-04-30 Thread Callum McGillivray





Hi Adam,

Unfortunatley we are located in Australia and our chosen provider does
not provide this service.

In the future as our client bae grows larger, we may need to look at
implmenting other carriers that provide this kind of service, but in
the meantime we will be using PRI's.

Cheers,

Callum

Adam Robins wrote:

  Why would you use gateways and PRI's when several of the major carriers
(AT&T, Global Crossing, etc.) also have products that can interface
directly with SIP for the same per minute cost?

We have a multisite Asterisk call center application and are routing all
calls over private VPN to one central Asterisk location from where we
have multiple point-to-point T1's going straight into Global Crossing.
They are accepting the traffic as SIP g.729a and are handling the
gateway themselves.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of Callum
McGillivray
Sent: Friday, April 29, 2005 1:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] T1 Technology and VoIP Gateway Primer

Hi Matt & everyone else,

We have also been steering toward using a gateway for our large
installation.

Ours differs from your significantly in as much as our setup will
involve 8 apartment buildings located throughout the CBD.  Each
apartment building will have as many as 600 extensions (rooms) with an
Asterisk Server in the comms room in the basement.

Incoming and Outgoing calls are going to be trunked from the Asterisk
box along a fiber link back to our core exchange, where the calls will
be handed off to a gateway machine (Cisco?) which will have an
impressively large number of PRI's plugged into the back of it.

My (very vague) examination so far tells me that I can use something
along the lines of a Cisco AS5400 (a couple of which I have kicking
around here in the office).

Has anyone had experience in handing off / receiving calls from a Cisco
AS5400 with Asterisk ? 

How is it done ?

Matt, is this similar to the idea that you have for your project ?  What
Cisco hardware have you looked at so far ?  How many E1/T1 lines are you
going to have terminating on your setup ?

Cheers,

Callum

Matt Roth wrote:

  
  
Michael,



  Have you decided which PSTN-VoIP gateway you'll use?
  



Not yet, but our preference is a Cisco gateway.  Lucent, Quintum, and 
AudioCodes also make TDM-VoIP gateways.

Prior to purchasing any hardware, our entire layout will be posted to 
this list in detail for review.

Matthew Roth
http://voip-info.org/tiki-index.php?page=Running%20Asterisk%20on%20Deb
ian ___
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

  
  

___
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

The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any.


___
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
  




___
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] Asterisk on Radio Tonight

True! I was asking on the irc channel to talk more about Asterisk vs. Cisco
and Avaya solutions... But like you said.. Well... What can We do. 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Kerry Garrison
|Sent: Domingo, 01 de Mayo de 2005 01:04 a.m.
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: RE: [Asterisk-Users] Asterisk on Radio Tonight
|
|Thanks! I didnt like their spin on trying to make it for home 
|users as much as they did, but oh well, I did what I could.
|-Kerry
| 
|
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Anton Krall
|Sent: Saturday, April 30, 2005 9:58 PM
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: RE: [Asterisk-Users] Asterisk on Radio Tonight
|
|You can grab Kerry's Radio Interview from my website
|
|http://www.intruder.com.mx/kerry.mp3
|
|Hope this helps. 
|
||-Original Message-
||From: [EMAIL PROTECTED]
||[mailto:[EMAIL PROTECTED] On Behalf Of Jim 
||Sturtevant
||Sent: Sábado, 30 de Abril de 2005 10:53 p.m.
||To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
||Subject: RE: [Asterisk-Users] Asterisk on Radio Tonight
||
||Kerry, can you put an archive of the audio up on your web 
|site or do we 
||need to record the whole 3hr show.  Also, the schedule on their web 
||site shows 5am EST and other repeats.
||I'd love to hear the program.
||
||
||
||-Original Message-
||From: [EMAIL PROTECTED]
||[mailto:[EMAIL PROTECTED] On Behalf Of Kerry 
||Garrison
||Sent: Saturday, April 30, 2005 2:28 PM
||To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
||Subject: [Asterisk-Users] Asterisk on Radio Tonight
||
||Kerry Garrison from The Geek Gazette (http://geekgazette.com) will be 
||interviewed tonight on Mick Mick Williams' Cyber Line radio 
|program at 
||9:00PM PST. The show is broadcast on the USA Radio network. If you do 
||not have a channel in your area, you can listen listen live online 
||. The show will cover the 
||basic of what the Asterisk PBX is all about and what it takes to 
||implement a system.
||
||-Kerry
||
||
||___
||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
||
||___
||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
||
||
|
|___
|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
|
|
|
|___
|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
|
|

___
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] Broadvoice limits???

2005-04-30 Thread Kerry Garrison




There is no physical limitation that I am aware of right 
now. Be sure and check you end user agreement but I think its pretty vauge. They 
told me once "its one call per account" but when I mentioned call waiting they 
said "ok well two calls". We have actually tested it with four to see if there 
was a limit.
 


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Tim 
ConnollySent: Saturday, April 30, 2005 11:09 PMTo: 
'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: 
[Asterisk-Users] Broadvoice limits???


    
Where are the limitations on the Broadvoice service? I saw a mention on the list 
saying two inbound/outbound calls, and Kerry just mentioned it during the radio 
interview… I don’t see that 2 call limitation with my BYOD World Plus account. 
Am I lucky, or just missing where the limitation 
is?
___
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

[Asterisk-Users] Broadvoice limits???









    Where
are the limitations on the Broadvoice service? I saw a mention on the list
saying two inbound/outbound calls, and Kerry just mentioned it during the radio
interview… I don’t see that 2 call limitation with my BYOD World
Plus account. Am I lucky, or just missing where the limitation is?






___
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] Send DTMF AFTER channels are bridged

2005-04-30 Thread Peter Svensson

On Sun, 1 May 2005, Matt Riddell wrote:

> > Someone to know how can I send a DTMF after the channels are bridged?
> > I need something like the "D" option of the Dial application, but this
> > option sends the DTMF before the channels are bridged. In fact I want the
> > caller and the callee to receive the DTMF. Please help :)
> 
> If using a codec with inband DTMF, you could always use the option to 
> play an audio file once connected, and just put the DTMF in there.

I think the Dial application was modified recently to allow dtmf to be 
sent to both the caller and callee.

   D([called][:calling])'  -- Send DTMF strings *after* called party 
has answered, but before the call gets bridged. The 'called' 
DTMF string is sent to the called party, and the 'calling' 
DTMF string is sent to the calling party. Both parameters 
can be used alone.\n"

Peter

___
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] Asterisk on Radio Tonight

2005-04-30 Thread Kerry Garrison

Thanks! I didnt like their spin on trying to make it for home users as much
as they did, but oh well, I did what I could.
-Kerry

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: Saturday, April 30, 2005 9:58 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Asterisk on Radio Tonight

You can grab Kerry's Radio Interview from my website

http://www.intruder.com.mx/kerry.mp3

Hope this helps. 

|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of Jim 
|Sturtevant
|Sent: Sábado, 30 de Abril de 2005 10:53 p.m.
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: RE: [Asterisk-Users] Asterisk on Radio Tonight
|
|Kerry, can you put an archive of the audio up on your web site or do we 
|need to record the whole 3hr show.  Also, the schedule on their web 
|site shows 5am EST and other repeats.
|I'd love to hear the program.
|
|
|
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Kerry Garrison
|Sent: Saturday, April 30, 2005 2:28 PM
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: [Asterisk-Users] Asterisk on Radio Tonight
|
|Kerry Garrison from The Geek Gazette (http://geekgazette.com)  
|will be interviewed tonight on Mick Mick Williams' Cyber Line 
|radio program at 9:00PM PST. The show is broadcast on the USA 
|Radio network. If you do not have a channel in your area, you 
|can listen listen live online 
|. The show will cover 
|the basic of what the Asterisk PBX is all about and what it 
|takes to implement a system.
|
|-Kerry
|
|
|___
|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
|
|___
|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
|
|

___
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

___
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] Asterisk@Home bug

2005-04-30 Thread Steve Prior

I disabled serial port login by commenting out the ttys0 line in
/etc/inittab.  Then after I rebooted the machine the message stopped.
There is a way to make this take effect without rebooting the machine,
but I was too lazy at the time...
I don't know what the real cause is though.  If you actually did need
to login through the serial port you couldn't use my fix.
Steve
Manny A. Wise wrote:
After installation of [EMAIL PROTECTED] v1, I have an annoying message in 
the screen, anyone know how to fix it

 

INIT: Id “s0” respawning too fast: disable for 5 minutes
___
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] Asterisk on Radio Tonight

You can grab Kerry's Radio Interview from my website

http://www.intruder.com.mx/kerry.mp3

Hope this helps. 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Jim Sturtevant
|Sent: Sábado, 30 de Abril de 2005 10:53 p.m.
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: RE: [Asterisk-Users] Asterisk on Radio Tonight
|
|Kerry, can you put an archive of the audio up on your web site 
|or do we need to record the whole 3hr show.  Also, the 
|schedule on their web site shows 5am EST and other repeats.  
|I'd love to hear the program.
|
|
|
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Kerry Garrison
|Sent: Saturday, April 30, 2005 2:28 PM
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: [Asterisk-Users] Asterisk on Radio Tonight
|
|Kerry Garrison from The Geek Gazette (http://geekgazette.com)  
|will be interviewed tonight on Mick Mick Williams' Cyber Line 
|radio program at 9:00PM PST. The show is broadcast on the USA 
|Radio network. If you do not have a channel in your area, you 
|can listen listen live online 
|. The show will cover 
|the basic of what the Asterisk PBX is all about and what it 
|takes to implement a system.
|
|-Kerry
|
|
|___
|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
|
|___
|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
|
|

___
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

[Asterisk-Users] Re: Hotel CDR Software

2005-04-30 Thread tgj

Hi,

IPSwitchBoard wil handle the billing, rates and check in/out part.

Download FREE: http://ipswitchboard.thorben.dk



"Anton Krall" <[EMAIL PROTECTED]> skrev i en meddelelse 
news:[EMAIL PROTECTED]
> Guys.
>
> Anybody knows of asterisk compliant cdr software for Hotel that will let 
> you
> enter diff. rates, checkin and out that will create the extension and 
> setup
> voicemail for the room, etc?
>
>
> ___
> 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
> 



___
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] Asterisk on Radio Tonight

2005-04-30 Thread Jim Sturtevant

That would be great, thanks

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: Saturday, April 30, 2005 9:42 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Asterisk on Radio Tonight

If anybody wants it, I recorded the part of the show and will put it up on
my website in about 30 mins. 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Jim Sturtevant
|Sent: Sábado, 30 de Abril de 2005 10:53 p.m.
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: RE: [Asterisk-Users] Asterisk on Radio Tonight
|
|Kerry, can you put an archive of the audio up on your web site 
|or do we need to record the whole 3hr show.  Also, the 
|schedule on their web site shows 5am EST and other repeats.  
|I'd love to hear the program.
|
|
|
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Kerry Garrison
|Sent: Saturday, April 30, 2005 2:28 PM
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: [Asterisk-Users] Asterisk on Radio Tonight
|
|Kerry Garrison from The Geek Gazette (http://geekgazette.com)  
|will be interviewed tonight on Mick Mick Williams' Cyber Line 
|radio program at 9:00PM PST. The show is broadcast on the USA 
|Radio network. If you do not have a channel in your area, you 
|can listen listen live online 
|. The show will cover 
|the basic of what the Asterisk PBX is all about and what it 
|takes to implement a system.
|
|-Kerry
|
|
|___
|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
|
|___
|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
|
|

___
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

___
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] Asterisk on Radio Tonight

If anybody wants it, I recorded the part of the show and will put it up on
my website in about 30 mins. 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Jim Sturtevant
|Sent: Sábado, 30 de Abril de 2005 10:53 p.m.
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: RE: [Asterisk-Users] Asterisk on Radio Tonight
|
|Kerry, can you put an archive of the audio up on your web site 
|or do we need to record the whole 3hr show.  Also, the 
|schedule on their web site shows 5am EST and other repeats.  
|I'd love to hear the program.
|
|
|
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Kerry Garrison
|Sent: Saturday, April 30, 2005 2:28 PM
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: [Asterisk-Users] Asterisk on Radio Tonight
|
|Kerry Garrison from The Geek Gazette (http://geekgazette.com)  
|will be interviewed tonight on Mick Mick Williams' Cyber Line 
|radio program at 9:00PM PST. The show is broadcast on the USA 
|Radio network. If you do not have a channel in your area, you 
|can listen listen live online 
|. The show will cover 
|the basic of what the Asterisk PBX is all about and what it 
|takes to implement a system.
|
|-Kerry
|
|
|___
|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
|
|___
|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
|
|

___
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] SIP over IAX2

Thanks. That's what I needed.
- Daniel
On Apr 30, 2005, at 8:00 PM, Tim Connolly wrote:
Asterisk Box 2 (agents register)
extensions.conf
[agents-context]
exten => 1234,1,Dial(SIP/[EMAIL PROTECTED])
exten => 1234,2,Hangup
Asterisk Box 1
Sip.conf
[ab1]
type=friend
host=
context=incoming
canreinvite=yes
qualify=yes
extension.conf
[incoming]
Exten => 1234etc...
-Original Message-
From: Daniel Salama [mailto:[EMAIL PROTECTED]
Sent: Saturday, April 30, 2005 6:50 PM
To: Tim Connolly
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SIP over IAX2
I understand and I guess I know how to do that within a single box.
If I have the following:
Asterisk Box 1 (no agents)
extensions.conf
[test-ivr]
exten => s,1,AGI(play_ivr)
exten => s,2,Hangup
Asterisk Box 2 (agents register)
extensions.conf
[agents-context]
exten => 1234,1,Dial(?)
exten => 1234,2,Hangup
Question is, when the agents dial 1234, how do I tell the application
to connect to the agent with context test-ivr of Asterisk_1?
Thanks,
Daniel
On Apr 30, 2005, at 7:12 PM, Tim Connolly wrote:
Maybe I'm missing something, but as long as you have the entension
defined
on the agent box to dial the extension on the IVR, you should be okay.
Just
make sure the default SIP context on the IVR has that extension
defined, or
define the IVR box as a SIP peer.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Daniel
Salama
Sent: Saturday, April 30, 2005 5:57 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] SIP over IAX2
I have two asterisk boxes. I'm running an IVR script in one of them 
and
I have agents registered on the second box.

I wish to create an extension on the * box where the agents are
registered, so that when dialed, it will connect the agent to the IVR
script on the other * box. However, I'd like for the connection to be
done using SIP instead of IAX. Can anyone help me, if at all possible,
write this configuration?
Thanks,
Daniel
___
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


Daniel Salama
[EMAIL PROTECTED]
Voice: (954) 655-8051
Fax  : (954) 252-3988

This e-mail contains information which may be confidential and
privileged. Unless you are the addressee (or authorized to
receive for the addressee), you may not use, copy or disclose
to anyone the message or any information contained in the
message.  If you have received the message in error, please
advise the sender by reply e-mail to [EMAIL PROTECTED] or
tel. +1-954-655-8051 and delete the material from any computer.
___
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] Asterisk on Radio Tonight

2005-04-30 Thread Jim Sturtevant

Kerry, can you put an archive of the audio up on your web site or do we need
to record the whole 3hr show.  Also, the schedule on their web site shows
5am EST and other repeats.  I'd love to hear the program.



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kerry Garrison
Sent: Saturday, April 30, 2005 2:28 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Asterisk on Radio Tonight

Kerry Garrison from The Geek Gazette (http://geekgazette.com)  will be
interviewed tonight on Mick Mick Williams' Cyber Line radio program at
9:00PM PST. The show is broadcast on the USA Radio network. If you do not
have a channel in your area, you can listen listen live online
. The show will cover the basic of
what the Asterisk PBX is all about and what it takes to implement a system.

-Kerry


___
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

___
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] Cisco 7960s and skinny

2005-04-30 Thread Julien Goodwin

On Sat, Apr 30, 2005 at 01:00:18PM -0500, Andy Hamilton arranged a set of bits 
into the following:
> I'll be able to get back to you Sunday night about specifics; the
> phone is not where I am right now. Using chan_sccp, (I think November
> 2004 or so CVS Head) I know I can receive calls, place calls, etc. It
> is a rather low volume phone, so I don't know off hand about specific
> keys; I'll check those later.
Generally if the phone supports the function, and support is in
chan_sccp for that function it will work for all phones.

> Additionally, I have not yet tried a new copy from CVS.
 
> Occasionally, I think the chan_sccp driver blips out in Asterisk (it
> may be the phone; I've had it apart several times because the on/off
> hook switch membrane is a little sketchy). I have dealt with this by
That's one of the big things that causes problems, both with chan_sccp
and the phones themselves, both get a little confused. However several
other crash issues have been recently fixed, so running CVS_HEAD is
advised.

> restarting Asterisk. The only other thing I can say right now about
> the 7910 is that it and my Cisco FastHub don't get along. At all. I
> have the 7910 plugged into my 7960.
That's odd, the only time I've ever had ethernet incompatabilities was
with a very cheap switch.

> Overall, I would say that if you have a non-critical system and would
> like to use a 7910, chan_sccp should be able to handle it fine. 
> However, if you budget permits, the 7960 and 7940 phones are quite
> nice (use SIP with those -- it's far more reliable. I must say,
> though, that my 7960 has frozen/crashed a handful of time when running
> the SIP image. That was the phone itself, Asterisk was fine.) I have
> yet to purchase a 7905 or 7912, but I've played around with some
> 7912's on a CCM system -- they seem quite nice and I think they take
> SIP.
Yep, they do. (Don't know about the 7902, but really can't see why
anyone would buy one)

>The 7920 is also nice because it's wireless. However, I don't
> think Cisco has anything but a Skinny image for it [yet].
No they don't, and forget the yet, if a phone isn't announced with SIP
support it probably never will have it (witness: 7935/6, 7970)

> I would stick with SIP wherever you can.
And I agree

Thanks,
Julien
chan_sccp project lead


pgpnrcHALcJbk.pgp
Description: PGP signature
___
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] Asterisk@Home bug

2005-04-30 Thread Mike

This is not that AAH mailing list, check out the fourms.

On Sat, 30 Apr 2005, Manny A. Wise wrote:
After installation of [EMAIL PROTECTED] v1, I have an annoying message in the
screen, anyone know how to fix it

INIT: Id "s0" respawning too fast: disable for 5 minutes

Thanks

___
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] Programing a call forward feature to cel phones

When I call outside using zaps, everything works fine and sounds good except
for the feedback from time to time.

Here is my full zapata.conf

[channels]
language=sp
signalling=fxs_ks
usecallerid=yes
cidsignalling=bell
cidstart=ring
hidecallerid=no
callwaiting=yes
usecallingpres=yes
;sendcalleridafter=1
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=no

echocancel=yes
;echocancelwhenbridged=yes
echocancelwhenbridged=no
echotraining=yes
echotraining=800

;relaxdtmf=yes
rxgain=6.0
txgain=6.0
group=1
callgroup=1
pickupgroup=1
immediate=no
callerid=asreceived
amaflags=default
;accountcode=lss0101
busydetect=yes
busycount=4
faxdetect=both
faxdetect=incoming
faxdetect=outgoing
faxdetect=yes
musiconhold=default
context=casa
channel => 1

context=intruder
language=sp
channel => 2
 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Tim Connolly
|Sent: Sábado, 30 de Abril de 2005 08:52 p.m.
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: RE: [Asterisk-Users] Programing a call forward 
|feature to cel phones
|
|Also, if the call goes back out on a zap device, check your gain.
|
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Tim Connolly
|Sent: Saturday, April 30, 2005 8:50 PM
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: RE: [Asterisk-Users] Programing a call forward 
|feature to cel phones
|
|Sounds like a echo/feedback which overmodulates the call once 
|it gets going.
|Make sure you have echotraining turned on, and maybe vad too.
|
|
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Anton Krall
|Sent: Saturday, April 30, 2005 8:40 PM
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: RE: [Asterisk-Users] Programing a call forward 
|feature to cel phones
|
|Here it is...  You can notice the problem after the voice and 
|some tones, then it goes bad.
|
|http://www.intruder.com.mx/badcall.wav
|
| 
|
||-Original Message-
||From: [EMAIL PROTECTED]
||[mailto:[EMAIL PROTECTED] On Behalf Of Tim 
||Connolly
||Sent: Sábado, 30 de Abril de 2005 08:33 p.m.
||To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
||Subject: RE: [Asterisk-Users] Programing a call forward 
|feature to cel 
||phones
||
||That would be entertaining at worst! Send a link to it please!
||
||
||-Original Message-
||From: [EMAIL PROTECTED]
||[mailto:[EMAIL PROTECTED] On Behalf Of Anton 
||Krall
||Sent: Saturday, April 30, 2005 8:29 PM
||To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
||Subject: RE: [Asterisk-Users] Programing a call forward 
|feature to cel 
||phones
||
||No, problem is not delay, problem is the sound it kicks in... 
||If you want I can forward you a recording of what happened.
|| 
||
|||-Original Message-
|||From: [EMAIL PROTECTED]
|||[mailto:[EMAIL PROTECTED] On Behalf 
|Of Gregory 
|||Wiktor - ADCom Corp.
|||Sent: Sábado, 30 de Abril de 2005 08:18 p.m.
|||To: Asterisk Users Mailing List - Non-Commercial Discussion
|||Subject: RE: [Asterisk-Users] Programing a call forward
||feature to cel
|||phones
|||
|||I know I saw something about not using GSM codecs when on
||cell phones,
|||could this be the case?
|||The 2 second delay, well unfortunately all cell's have about a
|||.5 second delay on their own, so that may be what you are
||hearing.  You
|||just need to learn how to talk like you are on an
||international call...
|||
|||Greg
|||
|||-Original Message-
|||From: [EMAIL PROTECTED]
|||[mailto:[EMAIL PROTECTED] On Behalf Of Anton 
|||Krall
|||Sent: Saturday, April 30, 2005 8:47 PM
|||To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|||Subject: RE: [Asterisk-Users] Programing a call forward
||feature to cel
|||phones
|||
|||Jejejejeje I didn't know how to put the sound it does... Its like an 
|||intermitent sound like when you are to lose a cel phone connection.
|||
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt 
Riddell
Sent: Sábado, 30 de Abril de 2005 07:15 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Programing a call forward
|||feature to cel
phones

Anton Krall wrote:
> Problem I have is that if somebody using a cel phone calls
in and gets
> directed to my extension which in turn is directed to my
||cel phone,
> the call comes thru but after 2 seconds, the call gets all
garbled and
> with a sound like b and the caller cant be
|||heard anymore.

Maybe the caller is cold?

:)

Sorry.

--
Cheers,

Matt Riddell
___

http://www.sineapps.com/news.php (Daily Asterisk News - html) 
http://www.sineapps.com/rssfeed.php (Daily Asterisk New

RE: [Asterisk-Users] Programing a call forward feature to cel phones

This problem also happens from time to time when calling using a zap line
(x100).
 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Tim Connolly
|Sent: Sábado, 30 de Abril de 2005 08:50 p.m.
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: RE: [Asterisk-Users] Programing a call forward 
|feature to cel phones
|
|Sounds like a echo/feedback which overmodulates the call once 
|it gets going.
|Make sure you have echotraining turned on, and maybe vad too.
|
|
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Anton Krall
|Sent: Saturday, April 30, 2005 8:40 PM
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: RE: [Asterisk-Users] Programing a call forward 
|feature to cel phones
|
|Here it is...  You can notice the problem after the voice and 
|some tones, then it goes bad.
|
|http://www.intruder.com.mx/badcall.wav
|
| 
|
||-Original Message-
||From: [EMAIL PROTECTED]
||[mailto:[EMAIL PROTECTED] On Behalf Of Tim 
||Connolly
||Sent: Sábado, 30 de Abril de 2005 08:33 p.m.
||To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
||Subject: RE: [Asterisk-Users] Programing a call forward 
|feature to cel 
||phones
||
||That would be entertaining at worst! Send a link to it please!
||
||
||-Original Message-
||From: [EMAIL PROTECTED]
||[mailto:[EMAIL PROTECTED] On Behalf Of Anton 
||Krall
||Sent: Saturday, April 30, 2005 8:29 PM
||To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
||Subject: RE: [Asterisk-Users] Programing a call forward 
|feature to cel 
||phones
||
||No, problem is not delay, problem is the sound it kicks in... 
||If you want I can forward you a recording of what happened.
|| 
||
|||-Original Message-
|||From: [EMAIL PROTECTED]
|||[mailto:[EMAIL PROTECTED] On Behalf 
|Of Gregory 
|||Wiktor - ADCom Corp.
|||Sent: Sábado, 30 de Abril de 2005 08:18 p.m.
|||To: Asterisk Users Mailing List - Non-Commercial Discussion
|||Subject: RE: [Asterisk-Users] Programing a call forward
||feature to cel
|||phones
|||
|||I know I saw something about not using GSM codecs when on
||cell phones,
|||could this be the case?
|||The 2 second delay, well unfortunately all cell's have about a
|||.5 second delay on their own, so that may be what you are
||hearing.  You
|||just need to learn how to talk like you are on an
||international call...
|||
|||Greg
|||
|||-Original Message-
|||From: [EMAIL PROTECTED]
|||[mailto:[EMAIL PROTECTED] On Behalf Of Anton 
|||Krall
|||Sent: Saturday, April 30, 2005 8:47 PM
|||To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|||Subject: RE: [Asterisk-Users] Programing a call forward
||feature to cel
|||phones
|||
|||Jejejejeje I didn't know how to put the sound it does... Its like an 
|||intermitent sound like when you are to lose a cel phone connection.
|||
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt 
Riddell
Sent: Sábado, 30 de Abril de 2005 07:15 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Programing a call forward
|||feature to cel
phones

Anton Krall wrote:
> Problem I have is that if somebody using a cel phone calls
in and gets
> directed to my extension which in turn is directed to my
||cel phone,
> the call comes thru but after 2 seconds, the call gets all
garbled and
> with a sound like b and the caller cant be
|||heard anymore.

Maybe the caller is cold?

:)

Sorry.

--
Cheers,

Matt Riddell
___

http://www.sineapps.com/news.php (Daily Asterisk News - html) 
http://www.sineapps.com/rssfeed.php (Daily Asterisk News -
rss) ___
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


|||
|||___
|||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
|||___
|||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
|||
|||
||
||___
||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.d

RE: [Asterisk-Users] Programing a call forward feature to cel phones

My zapata reads:

echocancel=yes
;echocancelwhenbridged=yes
echocancelwhenbridged=no
echotraining=yes
echotraining=800

Vmd?? 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Tim Connolly
|Sent: Sábado, 30 de Abril de 2005 08:50 p.m.
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: RE: [Asterisk-Users] Programing a call forward 
|feature to cel phones
|
|Sounds like a echo/feedback which overmodulates the call once 
|it gets going.
|Make sure you have echotraining turned on, and maybe vad too.
|
|
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Anton Krall
|Sent: Saturday, April 30, 2005 8:40 PM
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: RE: [Asterisk-Users] Programing a call forward 
|feature to cel phones
|
|Here it is...  You can notice the problem after the voice and 
|some tones, then it goes bad.
|
|http://www.intruder.com.mx/badcall.wav
|
| 
|
||-Original Message-
||From: [EMAIL PROTECTED]
||[mailto:[EMAIL PROTECTED] On Behalf Of Tim 
||Connolly
||Sent: Sábado, 30 de Abril de 2005 08:33 p.m.
||To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
||Subject: RE: [Asterisk-Users] Programing a call forward 
|feature to cel 
||phones
||
||That would be entertaining at worst! Send a link to it please!
||
||
||-Original Message-
||From: [EMAIL PROTECTED]
||[mailto:[EMAIL PROTECTED] On Behalf Of Anton 
||Krall
||Sent: Saturday, April 30, 2005 8:29 PM
||To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
||Subject: RE: [Asterisk-Users] Programing a call forward 
|feature to cel 
||phones
||
||No, problem is not delay, problem is the sound it kicks in... 
||If you want I can forward you a recording of what happened.
|| 
||
|||-Original Message-
|||From: [EMAIL PROTECTED]
|||[mailto:[EMAIL PROTECTED] On Behalf 
|Of Gregory 
|||Wiktor - ADCom Corp.
|||Sent: Sábado, 30 de Abril de 2005 08:18 p.m.
|||To: Asterisk Users Mailing List - Non-Commercial Discussion
|||Subject: RE: [Asterisk-Users] Programing a call forward
||feature to cel
|||phones
|||
|||I know I saw something about not using GSM codecs when on
||cell phones,
|||could this be the case?
|||The 2 second delay, well unfortunately all cell's have about a
|||.5 second delay on their own, so that may be what you are
||hearing.  You
|||just need to learn how to talk like you are on an
||international call...
|||
|||Greg
|||
|||-Original Message-
|||From: [EMAIL PROTECTED]
|||[mailto:[EMAIL PROTECTED] On Behalf Of Anton 
|||Krall
|||Sent: Saturday, April 30, 2005 8:47 PM
|||To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|||Subject: RE: [Asterisk-Users] Programing a call forward
||feature to cel
|||phones
|||
|||Jejejejeje I didn't know how to put the sound it does... Its like an 
|||intermitent sound like when you are to lose a cel phone connection.
|||
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt 
Riddell
Sent: Sábado, 30 de Abril de 2005 07:15 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Programing a call forward
|||feature to cel
phones

Anton Krall wrote:
> Problem I have is that if somebody using a cel phone calls
in and gets
> directed to my extension which in turn is directed to my
||cel phone,
> the call comes thru but after 2 seconds, the call gets all
garbled and
> with a sound like b and the caller cant be
|||heard anymore.

Maybe the caller is cold?

:)

Sorry.

--
Cheers,

Matt Riddell
___

http://www.sineapps.com/news.php (Daily Asterisk News - html) 
http://www.sineapps.com/rssfeed.php (Daily Asterisk News -
rss) ___
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


|||
|||___
|||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
|||___
|||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
|||
|||
||
||___
||Asterisk-Users mailing list
||Asterisk-Users@lists.digium.com
||http://lists.digium.com/mailman/listinfo/asterisk-users
||To UNSUBSCRIBE o

[Asterisk-Users] Asterisk@Home bug

2005-04-30 Thread Manny A. Wise









After installation of [EMAIL PROTECTED] v1, I have an annoying message
in the screen, anyone know how to fix it

 

INIT: Id “s0” respawning too fast: disable for 5
minutes

 

Thanks






___
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] Problem with Sangoma/Adtran 600 installation

2005-04-30 Thread mattf

Hello,

is this how you are starting up:

1. modprobe zaptel
2. wanrouter start
3. ztcfg -v
4. asterisk -vvvgc

Also, what version of the wanroute driver software are you using?

MATT---

-Original Message-
From: Chris Mason (Lists) [mailto:[EMAIL PROTECTED]
Sent: Saturday, April 30, 2005 5:59 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Problem with Sangoma/Adtran 600 installation


I have installed Asterisk on a CentOS4 box and then installed Asterisk from
CVS.
I installed a Sangoma A101 and connected it to a Adtran 600 using a T1
Crossover cable. The 600 has 12 x FXS, 12 x FXO interfaces.

I ran through the wanpipe install instructions and configured it, now I can
run 

[EMAIL PROTECTED] asterisk]#  wanrouter hwprobe

---
| Wanpipe Hardware Probe Info |
---
1 . AFT-A101u  : SLOT=1 : BUS=1 : IRQ=209 : CPU=A : PORT=PRI

Card Cnt: S508=0  S514X=0  S518=0  A101-2=1  A104=0  A300=0 

So I know the card is there OK.

My /etc/zaptel.conf looks like:
span=1,1,0,esf,b8zs
loadzone = us
defaultzone=us
fxsls=1-12

I am only trying to get half to load for now to make it simple.

[EMAIL PROTECTED] asterisk]# ztcfg
Notice: Configuration file is /etc/zaptel.conf
line 0: Unable to open master device '/dev/zap/ctl'

1 error(s) detected

When I run "service zaptel restart" I get:

Waiting for zap to come online...Error: missing /dev/zap!

Wha am I doing wrong?


Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (646)722-0001 Fax: (815)301-9759 
(305) 704-7249
Yahoo IM: [EMAIL PROTECTED] 
Skype ID: netconcepts

___
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
___
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] Programing a call forward feature to cel phones

Also, if the call goes back out on a zap device, check your gain.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim Connolly
Sent: Saturday, April 30, 2005 8:50 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Programing a call forward feature to cel
phones

Sounds like a echo/feedback which overmodulates the call once it gets going.
Make sure you have echotraining turned on, and maybe vad too.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: Saturday, April 30, 2005 8:40 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Programing a call forward feature to cel
phones

Here it is...  You can notice the problem after the voice and some tones,
then it goes bad.

http://www.intruder.com.mx/badcall.wav

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Tim Connolly
|Sent: Sábado, 30 de Abril de 2005 08:33 p.m.
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: RE: [Asterisk-Users] Programing a call forward 
|feature to cel phones
|
|That would be entertaining at worst! Send a link to it please!
|
|
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Anton Krall
|Sent: Saturday, April 30, 2005 8:29 PM
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: RE: [Asterisk-Users] Programing a call forward 
|feature to cel phones
|
|No, problem is not delay, problem is the sound it kicks in... 
|If you want I can forward you a recording of what happened.
| 
|
||-Original Message-
||From: [EMAIL PROTECTED]
||[mailto:[EMAIL PROTECTED] On Behalf Of Gregory 
||Wiktor - ADCom Corp.
||Sent: Sábado, 30 de Abril de 2005 08:18 p.m.
||To: Asterisk Users Mailing List - Non-Commercial Discussion
||Subject: RE: [Asterisk-Users] Programing a call forward 
|feature to cel 
||phones
||
||I know I saw something about not using GSM codecs when on 
|cell phones, 
||could this be the case?
||The 2 second delay, well unfortunately all cell's have about a
||.5 second delay on their own, so that may be what you are 
|hearing.  You 
||just need to learn how to talk like you are on an 
|international call...
||
||Greg
||
||-Original Message-
||From: [EMAIL PROTECTED]
||[mailto:[EMAIL PROTECTED] On Behalf Of Anton 
||Krall
||Sent: Saturday, April 30, 2005 8:47 PM
||To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
||Subject: RE: [Asterisk-Users] Programing a call forward 
|feature to cel 
||phones
||
||Jejejejeje I didn't know how to put the sound it does... Its like an 
||intermitent sound like when you are to lose a cel phone connection.
||
|||-Original Message-
|||From: [EMAIL PROTECTED]
|||[mailto:[EMAIL PROTECTED] On Behalf Of Matt 
|||Riddell
|||Sent: Sábado, 30 de Abril de 2005 07:15 p.m.
|||To: Asterisk Users Mailing List - Non-Commercial Discussion
|||Subject: Re: [Asterisk-Users] Programing a call forward
||feature to cel
|||phones
|||
|||Anton Krall wrote:
|||> Problem I have is that if somebody using a cel phone calls
|||in and gets
|||> directed to my extension which in turn is directed to my 
|cel phone, 
|||> the call comes thru but after 2 seconds, the call gets all
|||garbled and
|||> with a sound like b and the caller cant be
||heard anymore.
|||
|||Maybe the caller is cold?
|||
|||:)
|||
|||Sorry.
|||
|||--
|||Cheers,
|||
|||Matt Riddell
|||___
|||
|||http://www.sineapps.com/news.php (Daily Asterisk News - html) 
|||http://www.sineapps.com/rssfeed.php (Daily Asterisk News -
|||rss) ___
|||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
|||
|||
||
||___
||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
||___
||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
||
||
|
|___
|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
|
|___
|Asterisk-Users mailing list
|Asterisk-Users@lists.digium.com
|http://lists.digium.co

RE: [Asterisk-Users] Programing a call forward feature to cel phones

Sounds like a echo/feedback which overmodulates the call once it gets going.
Make sure you have echotraining turned on, and maybe vad too.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: Saturday, April 30, 2005 8:40 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Programing a call forward feature to cel
phones

Here it is...  You can notice the problem after the voice and some tones,
then it goes bad.

http://www.intruder.com.mx/badcall.wav

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Tim Connolly
|Sent: Sábado, 30 de Abril de 2005 08:33 p.m.
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: RE: [Asterisk-Users] Programing a call forward 
|feature to cel phones
|
|That would be entertaining at worst! Send a link to it please!
|
|
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Anton Krall
|Sent: Saturday, April 30, 2005 8:29 PM
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: RE: [Asterisk-Users] Programing a call forward 
|feature to cel phones
|
|No, problem is not delay, problem is the sound it kicks in... 
|If you want I can forward you a recording of what happened.
| 
|
||-Original Message-
||From: [EMAIL PROTECTED]
||[mailto:[EMAIL PROTECTED] On Behalf Of Gregory 
||Wiktor - ADCom Corp.
||Sent: Sábado, 30 de Abril de 2005 08:18 p.m.
||To: Asterisk Users Mailing List - Non-Commercial Discussion
||Subject: RE: [Asterisk-Users] Programing a call forward 
|feature to cel 
||phones
||
||I know I saw something about not using GSM codecs when on 
|cell phones, 
||could this be the case?
||The 2 second delay, well unfortunately all cell's have about a
||.5 second delay on their own, so that may be what you are 
|hearing.  You 
||just need to learn how to talk like you are on an 
|international call...
||
||Greg
||
||-Original Message-
||From: [EMAIL PROTECTED]
||[mailto:[EMAIL PROTECTED] On Behalf Of Anton 
||Krall
||Sent: Saturday, April 30, 2005 8:47 PM
||To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
||Subject: RE: [Asterisk-Users] Programing a call forward 
|feature to cel 
||phones
||
||Jejejejeje I didn't know how to put the sound it does... Its like an 
||intermitent sound like when you are to lose a cel phone connection.
||
|||-Original Message-
|||From: [EMAIL PROTECTED]
|||[mailto:[EMAIL PROTECTED] On Behalf Of Matt 
|||Riddell
|||Sent: Sábado, 30 de Abril de 2005 07:15 p.m.
|||To: Asterisk Users Mailing List - Non-Commercial Discussion
|||Subject: Re: [Asterisk-Users] Programing a call forward
||feature to cel
|||phones
|||
|||Anton Krall wrote:
|||> Problem I have is that if somebody using a cel phone calls
|||in and gets
|||> directed to my extension which in turn is directed to my 
|cel phone, 
|||> the call comes thru but after 2 seconds, the call gets all
|||garbled and
|||> with a sound like b and the caller cant be
||heard anymore.
|||
|||Maybe the caller is cold?
|||
|||:)
|||
|||Sorry.
|||
|||--
|||Cheers,
|||
|||Matt Riddell
|||___
|||
|||http://www.sineapps.com/news.php (Daily Asterisk News - html) 
|||http://www.sineapps.com/rssfeed.php (Daily Asterisk News -
|||rss) ___
|||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
|||
|||
||
||___
||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
||___
||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
||
||
|
|___
|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
|
|___
|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
|
|

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

RE: [Asterisk-Users] Programing a call forward feature to cel phones

Here it is...  You can notice the problem after the voice and some tones,
then it goes bad.

http://www.intruder.com.mx/badcall.wav

 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Tim Connolly
|Sent: Sábado, 30 de Abril de 2005 08:33 p.m.
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: RE: [Asterisk-Users] Programing a call forward 
|feature to cel phones
|
|That would be entertaining at worst! Send a link to it please!
|
|
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Anton Krall
|Sent: Saturday, April 30, 2005 8:29 PM
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: RE: [Asterisk-Users] Programing a call forward 
|feature to cel phones
|
|No, problem is not delay, problem is the sound it kicks in... 
|If you want I can forward you a recording of what happened.
| 
|
||-Original Message-
||From: [EMAIL PROTECTED]
||[mailto:[EMAIL PROTECTED] On Behalf Of Gregory 
||Wiktor - ADCom Corp.
||Sent: Sábado, 30 de Abril de 2005 08:18 p.m.
||To: Asterisk Users Mailing List - Non-Commercial Discussion
||Subject: RE: [Asterisk-Users] Programing a call forward 
|feature to cel 
||phones
||
||I know I saw something about not using GSM codecs when on 
|cell phones, 
||could this be the case?
||The 2 second delay, well unfortunately all cell's have about a
||.5 second delay on their own, so that may be what you are 
|hearing.  You 
||just need to learn how to talk like you are on an 
|international call...
||
||Greg
||
||-Original Message-
||From: [EMAIL PROTECTED]
||[mailto:[EMAIL PROTECTED] On Behalf Of Anton 
||Krall
||Sent: Saturday, April 30, 2005 8:47 PM
||To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
||Subject: RE: [Asterisk-Users] Programing a call forward 
|feature to cel 
||phones
||
||Jejejejeje I didn't know how to put the sound it does... Its like an 
||intermitent sound like when you are to lose a cel phone connection.
||
|||-Original Message-
|||From: [EMAIL PROTECTED]
|||[mailto:[EMAIL PROTECTED] On Behalf Of Matt 
|||Riddell
|||Sent: Sábado, 30 de Abril de 2005 07:15 p.m.
|||To: Asterisk Users Mailing List - Non-Commercial Discussion
|||Subject: Re: [Asterisk-Users] Programing a call forward
||feature to cel
|||phones
|||
|||Anton Krall wrote:
|||> Problem I have is that if somebody using a cel phone calls
|||in and gets
|||> directed to my extension which in turn is directed to my 
|cel phone, 
|||> the call comes thru but after 2 seconds, the call gets all
|||garbled and
|||> with a sound like b and the caller cant be
||heard anymore.
|||
|||Maybe the caller is cold?
|||
|||:)
|||
|||Sorry.
|||
|||--
|||Cheers,
|||
|||Matt Riddell
|||___
|||
|||http://www.sineapps.com/news.php (Daily Asterisk News - html) 
|||http://www.sineapps.com/rssfeed.php (Daily Asterisk News -
|||rss) ___
|||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
|||
|||
||
||___
||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
||___
||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
||
||
|
|___
|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
|
|___
|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
|
|

___
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] Programing a call forward feature to cel phones

The forward is done like this

You enter the number to use as forward into astdb.

When a call comes in for that extension, its reads the number from astdb and
then does this:

[macro-stdexten];
exten => s,1,GotoIf($["${ARG1}" = ""]?s,110:s,2)
exten => s,2,DBGet(DNDFWD=DNDFWD/${ARG2})
exten => s,3,NoOp(DNDFWD: ${DNDFWD})
exten => s,4,GotoIf($["${DNDFWD}" = "${ARG2}"]?s-NOANSWER,1:s,5)
exten => s,5,GotoIf($[${LEN(${DNDFWD})} >= 8]?s,6:s,120)
exten =>
s,6,SetVar(CALLFILENAME=${TIMESTAMP}-from${CALLERIDNUM}-to${ARG2}-forwarded-
to-${DNDFWD})
exten => s,7,Monitor(wav,${CALLFILENAME},m)
exten => s,8,Playback(${SONIDOS}/bridgingcall)
exten => s,9,Macro(zapdial,${DNDFWD})
exten => s,10,Playback(vm-goodbye)
exten => s,11,Hangup
exten => s,103,System(/bin/echo -n -e "'From: ${CALLERIDNAME} -
${CALLERIDNUM} To: ${ARG2}'" | nc -w 1 10.0.0.2 10629)
exten =>
s,104,SetVar(CALLFILENAME=${TIMESTAMP}-from${CALLERIDNUM}-to${ARG2})
exten => s,105,Monitor(wav,${CALLFILENAME},m)
exten => s,106,Dial(${ARG1},${ARG3},${ARG4})
exten => s,107,Goto(s-${DIALSTATUS},1)
exten => s,110,Background(${SONIDOS}/incorrecta)
exten => s,111,Playback(vm-goodbye)
exten => s,112,Hangup
exten => s,120,GotoIf($[${LEN(${DNDFWD})} = 3]?s,121:s-NOANSWER,1)
exten =>
s,121,SetVar(CALLFILENAME=${TIMESTAMP}-from${CALLERIDNUM}-to${ARG2}-forwarde
d-to-${DNDFWD})
exten => s,122,Monitor(wav,${CALLFILENAME},m)
exten => s,123,System(/bin/echo -n -e "'From: ${CALLERIDNAME} -
${CALLERIDNUM} To: ${ARG2}'" | nc -w 1 10.0.0.2 10629)
exten => s,124,Dial(${exten${DNDFWD}},${ARG3},${ARG4})
exten => s,125,Goto(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,Voicemail(u${ARG2})
exten => s-NOANSWER,2,Playback(vm-goodbye)
exten => s-NOANSWER,3,Hangup
exten => s-BUSY,1,Voicemail(b${ARG2})
exten => s-BUSY,2,Playback(vm-goodbye)
exten => s-BUSY,3,Hangup
exten => _s-.,1,Goto(s-NOANSWER,1) 

This check the forwarding number and if its the same as the extension
number, send the caller to voicemail, if = 3 digits, then forward to an
internal extension and if >= 8 digits, then dial outside using zap.



|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Race Vanderdecken
|Sent: Sábado, 30 de Abril de 2005 08:24 p.m.
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: RE: [Asterisk-Users] Programing a call forward 
|feature to cel phones
|
|Curious,
|
|How did you do the forward? Was it a script or programming in C?
|
|Any output from debug?
|
|Race "The Tyrant" Vanderdecken
|
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Anton Krall
|Sent: Saturday, April 30, 2005 8:00 PM
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: [Asterisk-Users] Programing a call forward feature to 
|cel phones
|
|Guys.
|
|I just programed a feature that allows any extension to be 
|forwarded to any outside number, for example, forward your 
|extension 201 to any number outside (via zap) so that if 
|somebody calls your extension either from inside out outside 
|(using another zap we have) it gets directed.
|
|Problem I have is that if somebody using a cel phone calls in 
|and gets directed to my extension which in turn is directed to 
|my cel phone, the call comes thru but after 2 seconds, the 
|call gets all garbled and with a sound like b and 
|the caller cant be heard anymore.
|
|Anybody has any idea why this is happening?
|
|
|___
|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
|
|
|___
|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
|
|

___
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] Programing a call forward feature to cel phones

That would be entertaining at worst! Send a link to it please!

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: Saturday, April 30, 2005 8:29 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Programing a call forward feature to cel
phones

No, problem is not delay, problem is the sound it kicks in... If you want I
can forward you a recording of what happened.

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Gregory Wiktor - ADCom Corp.
|Sent: Sábado, 30 de Abril de 2005 08:18 p.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: RE: [Asterisk-Users] Programing a call forward 
|feature to cel phones
|
|I know I saw something about not using GSM codecs when on cell 
|phones, could this be the case?
|The 2 second delay, well unfortunately all cell's have about a 
|.5 second delay on their own, so that may be what you are 
|hearing.  You just need to learn how to talk like you are on 
|an international call...
|
|Greg 
|
|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Anton Krall
|Sent: Saturday, April 30, 2005 8:47 PM
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: RE: [Asterisk-Users] Programing a call forward 
|feature to cel phones
|
|Jejejejeje I didn't know how to put the sound it does... Its 
|like an intermitent sound like when you are to lose a cel 
|phone connection. 
|
||-Original Message-
||From: [EMAIL PROTECTED]
||[mailto:[EMAIL PROTECTED] On Behalf Of Matt 
||Riddell
||Sent: Sábado, 30 de Abril de 2005 07:15 p.m.
||To: Asterisk Users Mailing List - Non-Commercial Discussion
||Subject: Re: [Asterisk-Users] Programing a call forward 
|feature to cel 
||phones
||
||Anton Krall wrote:
||> Problem I have is that if somebody using a cel phone calls
||in and gets
||> directed to my extension which in turn is directed to my cel phone, 
||> the call comes thru but after 2 seconds, the call gets all
||garbled and
||> with a sound like b and the caller cant be 
|heard anymore.
||
||Maybe the caller is cold?
||
||:)
||
||Sorry.
||
||--
||Cheers,
||
||Matt Riddell
||___
||
||http://www.sineapps.com/news.php (Daily Asterisk News - html) 
||http://www.sineapps.com/rssfeed.php (Daily Asterisk News -
||rss) ___
||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
||
||
|
|___
|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
|___
|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
|
|

___
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

___
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] Programing a call forward feature to cel phones

No, problem is not delay, problem is the sound it kicks in... If you want I
can forward you a recording of what happened.
 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Gregory Wiktor - ADCom Corp.
|Sent: Sábado, 30 de Abril de 2005 08:18 p.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: RE: [Asterisk-Users] Programing a call forward 
|feature to cel phones
|
|I know I saw something about not using GSM codecs when on cell 
|phones, could this be the case?
|The 2 second delay, well unfortunately all cell's have about a 
|.5 second delay on their own, so that may be what you are 
|hearing.  You just need to learn how to talk like you are on 
|an international call...
|
|Greg 
|
|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Anton Krall
|Sent: Saturday, April 30, 2005 8:47 PM
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: RE: [Asterisk-Users] Programing a call forward 
|feature to cel phones
|
|Jejejejeje I didn't know how to put the sound it does... Its 
|like an intermitent sound like when you are to lose a cel 
|phone connection. 
|
||-Original Message-
||From: [EMAIL PROTECTED]
||[mailto:[EMAIL PROTECTED] On Behalf Of Matt 
||Riddell
||Sent: Sábado, 30 de Abril de 2005 07:15 p.m.
||To: Asterisk Users Mailing List - Non-Commercial Discussion
||Subject: Re: [Asterisk-Users] Programing a call forward 
|feature to cel 
||phones
||
||Anton Krall wrote:
||> Problem I have is that if somebody using a cel phone calls
||in and gets
||> directed to my extension which in turn is directed to my cel phone, 
||> the call comes thru but after 2 seconds, the call gets all
||garbled and
||> with a sound like b and the caller cant be 
|heard anymore.
||
||Maybe the caller is cold?
||
||:)
||
||Sorry.
||
||--
||Cheers,
||
||Matt Riddell
||___
||
||http://www.sineapps.com/news.php (Daily Asterisk News - html) 
||http://www.sineapps.com/rssfeed.php (Daily Asterisk News -
||rss) ___
||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
||
||
|
|___
|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
|___
|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
|
|

___
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] Programing a call forward feature to cel phones

:)
No problem dialing another cell phone from asterisk or incoming from cel
phone, etc.

Console says nothing.

The forwarded call is been directed using zap (x100)

So nothing looks wrong... But still...cant figure out why forwarding the
call to a cel phone via zap gets those weird sounds after 2 seconds of
talking and why this happens just when redirecting to a cel phone. Seems
that if you redirect to a land line is ok.

Also, sometimes, when in a call, any call (cel, land line, etc) sometimes a
weird sound much like the one I mentioned kicks in the call and I cant get
the caller because of the sound and he cant listen to me, so I need to hit
flash and then flash again and the call continues without the sounds...
Anybody seen that before? Could it be asterisk or the x100? Maybe worth
mentioning, that I use Monitor to records all calls... Could that be it ? 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Matt Riddell
|Sent: Sábado, 30 de Abril de 2005 08:09 p.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] Programing a call forward 
|feature to cel phones
|
|Anton Krall wrote:
|> Jejejejeje I didnt know how to put the sound it does... Its like an 
|> intermitent sound like when you are to lose a cel phone connection.
|
|Ah...and the cellphone has range?
|
|Normally I would say that seeing as you normally hear in on a 
|cell phone it would likely be that end, but maybe they are 
|just playing back the last bit of audio data repeatedly until 
|they get another one.
|
|How are you connecting to the cell phone?
|
|What do you see in the Asterisk console?
|
|How is the call getting to your cellphone?
|
|What happens if you dial straight from your Asterisk box to 
|your Cellphone?
|
|What happens if you dial another cellphone?
|
|Do you have any problems when you dial in from you cellphone 
|to Asterisk?
|
|What happens if you send the call to another land line instead?
|
|--
|Cheers,
|
|Matt Riddell
|___
|
|http://www.sineapps.com/news.php (Daily Asterisk News - html) 
|http://www.sineapps.com/rssfeed.php (Daily Asterisk News - 
|rss) ___
|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
|

___
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] Programing a call forward feature to cel phones

2005-04-30 Thread Race Vanderdecken

Curious,

How did you do the forward? Was it a script or programming in C?

Any output from debug?

Race "The Tyrant" Vanderdecken

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton
Krall
Sent: Saturday, April 30, 2005 8:00 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Programing a call forward feature to cel
phones

Guys.

I just programed a feature that allows any extension to be forwarded to
any
outside number, for example, forward your extension 201 to any number
outside (via zap) so that if somebody calls your extension either from
inside out outside (using another zap we have) it gets directed.

Problem I have is that if somebody using a cel phone calls in and gets
directed to my extension which in turn is directed to my cel phone, the
call
comes thru but after 2 seconds, the call gets all garbled and with a
sound
like b and the caller cant be heard anymore.

Anybody has any idea why this is happening?


___
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


___
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] How to bridge 2 calls

2005-04-30 Thread Gregory Wiktor - ADCom Corp.

I just made an extension 390 that calls my cell, so people can hold,
then send to 390 and hangup.

Greg 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton
Krall
Sent: Saturday, April 30, 2005 7:47 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] How to bridge 2 calls

Guys.

I have some dialing rules defined for my internal extensions but I am
now defning a call forward option that allow an extension to be
forwarded to an outside number, right now Im using Dial cmds but I was
wondering if ther is a way to do this but using the dialing rules that I
have also defined for the internal extensions? For exaple, like DISA
does...

Any ideas?

___
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
___
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] Programing a call forward feature to cel phones

2005-04-30 Thread Gregory Wiktor - ADCom Corp.

I know I saw something about not using GSM codecs when on cell phones, could 
this be the case?
The 2 second delay, well unfortunately all cell's have about a .5 second delay 
on their own, so that may be what you are hearing.  You just need to learn how 
to talk like you are on an international call...

Greg 

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: Saturday, April 30, 2005 8:47 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Programing a call forward feature to cel phones

Jejejejeje I didn't know how to put the sound it does... Its like an 
intermitent sound like when you are to lose a cel phone connection. 

|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of Matt 
|Riddell
|Sent: Sábado, 30 de Abril de 2005 07:15 p.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] Programing a call forward feature to cel 
|phones
|
|Anton Krall wrote:
|> Problem I have is that if somebody using a cel phone calls
|in and gets
|> directed to my extension which in turn is directed to my cel phone, 
|> the call comes thru but after 2 seconds, the call gets all
|garbled and
|> with a sound like b and the caller cant be heard anymore.
|
|Maybe the caller is cold?
|
|:)
|
|Sorry.
|
|--
|Cheers,
|
|Matt Riddell
|___
|
|http://www.sineapps.com/news.php (Daily Asterisk News - html) 
|http://www.sineapps.com/rssfeed.php (Daily Asterisk News -
|rss) ___
|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
|
|

___
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
___
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] A good SIP receptionist phone

2005-04-30 Thread Gregory Wiktor - ADCom Corp.

Hello Sean,
I thought the Polycom's had some kind of BLF Feature don't they?

I am thinking of getting two of them, so it would be nice to know,
otherwise I would get 2 more 7960's. (which are great phones)

Greg 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sean
Kennedy
Sent: Saturday, April 30, 2005 9:03 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] A good SIP receptionist phone

Jason Brown wrote:

>I have a problem. The average person is too freaking stupid to use a
VOIP phone. My experience has so far been that if it doesn't have 20
buttons with little red LED's on it, the user cannot comprehend call
parking, attended transfer, blind transfer, DND, and navigating through
a voicemail menu.
>
>I need a good receptionist phone that works with Asterisk. It basically
needs to act like an avaya partner phone, I don't need 20 buttons with
little red LED's...what I do need is for the phone to register multiple
extensions to my asterisk server and act like each SIP extension is a
line, so if the idiot receptionist has a call ringing in on line 1, she
can pick it up, look at the buttons, see a call ringing in on line 2
(and the phone ringer rings), put call 1 on hold without hanging the
caller up, and hit the little "I am an idiot and need a line 2 button"
to pick up line 2, so on and so forth.
>
>I love VOIP systems and all the functionality they bring and features I
get. Unfortunately, the average person in this country anymore is
apparently completely stupid and cannot understand how to juggle calls
without hanging up on people.
>
>
>
>So seriously does anyone have a recommendation for a good receptionist
phone? I tried the Snom today and I can't get the programmable buttons
to do this, even by following the manual. So please, any suggestions
would be great, before I get fired at my dayjob for everyone else's
idiocy.
>

1) I suggest you learn to live and like those "idiots".  I also suggest
you tone down that attitude and adjust it.  Those idiots contribute to
YOUR pay.

2) There isn't anything like what you want.  I know, I want the same
thing.  There is no phone out there that will do this with any protocol
that asterisk uses.  This is the one major failing of asterisk ( and
voip in general.  I smell an oportunity for a phone manufacture ), and
what keeps it out of a lot of places.

I can see this being implemented with a phone that speaks to *'s manager
interface.  Who wants to talk to polycom or cisco about it?  :)

Sean
___
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
___
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] Programing a call forward feature to cel phones

2005-04-30 Thread Matt Riddell

Anton Krall wrote:
Jejejejeje I didn’t know how to put the sound it does... Its like an
intermitent sound like when you are to lose a cel phone connection. 
Ah...and the cellphone has range?
Normally I would say that seeing as you normally hear in on a cell phone 
it would likely be that end, but maybe they are just playing back the 
last bit of audio data repeatedly until they get another one.

How are you connecting to the cell phone?
What do you see in the Asterisk console?
How is the call getting to your cellphone?
What happens if you dial straight from your Asterisk box to your Cellphone?
What happens if you dial another cellphone?
Do you have any problems when you dial in from you cellphone to Asterisk?
What happens if you send the call to another land line instead?
--
Cheers,
Matt Riddell
___
http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)
___
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] A good SIP receptionist phone

2005-04-30 Thread Sean Kennedy

Jason Brown wrote:
I have a problem. The average person is too freaking stupid to use a VOIP 
phone. My experience has so far been that if it doesn't have 20 buttons with 
little red LED's on it, the user cannot comprehend call parking, attended 
transfer, blind transfer, DND, and navigating through a voicemail menu.
I need a good receptionist phone that works with Asterisk. It basically needs to act like 
an avaya partner phone, I don't need 20 buttons with little red LED's...what I do need is 
for the phone to register multiple extensions to my asterisk server and act like each SIP 
extension is a line, so if the idiot receptionist has a call ringing in on line 1, she 
can pick it up, look at the buttons, see a call ringing in on line 2 (and the phone 
ringer rings), put call 1 on hold without hanging the caller up, and hit the little 
"I am an idiot and need a line 2 button" to pick up line 2, so on and so forth.
I love VOIP systems and all the functionality they bring and features I get. 
Unfortunately, the average person in this country anymore is apparently 
completely stupid and cannot understand how to juggle calls without hanging up 
on people.

So seriously does anyone have a recommendation for a good receptionist phone? I 
tried the Snom today and I can't get the programmable buttons to do this, even 
by following the manual. So please, any suggestions would be great, before I 
get fired at my dayjob for everyone else's idiocy.
1) I suggest you learn to live and like those "idiots".  I also suggest 
you tone down that attitude and adjust it.  Those idiots contribute to 
YOUR pay.

2) There isn't anything like what you want.  I know, I want the same 
thing.  There is no phone out there that will do this with any protocol 
that asterisk uses.  This is the one major failing of asterisk ( and 
voip in general.  I smell an oportunity for a phone manufacture ), and 
what keeps it out of a lot of places.

I can see this being implemented with a phone that speaks to *'s manager 
interface.  Who wants to talk to polycom or cisco about it?  :)

Sean
___
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] Programing a call forward feature to cel phones

Jejejejeje I didnt know how to put the sound it does... Its like an
intermitent sound like when you are to lose a cel phone connection. 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Matt Riddell
|Sent: Sábado, 30 de Abril de 2005 07:15 p.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] Programing a call forward 
|feature to cel phones
|
|Anton Krall wrote:
|> Problem I have is that if somebody using a cel phone calls 
|in and gets 
|> directed to my extension which in turn is directed to my cel phone, 
|> the call comes thru but after 2 seconds, the call gets all 
|garbled and 
|> with a sound like b and the caller cant be heard anymore.
|
|Maybe the caller is cold?
|
|:)
|
|Sorry.
|
|--
|Cheers,
|
|Matt Riddell
|___
|
|http://www.sineapps.com/news.php (Daily Asterisk News - html) 
|http://www.sineapps.com/rssfeed.php (Daily Asterisk News - 
|rss) ___
|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
|
|

___
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] Programing a call forward feature to cel phones

2005-04-30 Thread Matt Riddell

Anton Krall wrote:
Problem I have is that if somebody using a cel phone calls in and gets
directed to my extension which in turn is directed to my cel phone, the call
comes thru but after 2 seconds, the call gets all garbled and with a sound
like b and the caller cant be heard anymore.
Maybe the caller is cold?
:)
Sorry.
--
Cheers,
Matt Riddell
___
http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)
___
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] Send DTMF AFTER channels are bridged

2005-04-30 Thread Matt Riddell

Shady wrote:
Someone to know how can I send a DTMF after the channels are bridged?
I need something like the "D" option of the Dial application, but this
option sends the DTMF before the channels are bridged. In fact I want the
caller and the callee to receive the DTMF. Please help :)
If using a codec with inband DTMF, you could always use the option to 
play an audio file once connected, and just put the DTMF in there.

--
Cheers,
Matt Riddell
___
http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)
___
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] Problems with TDM400P card

2005-04-30 Thread Andrew Kohlsmith

On April 30, 2005 10:23 am, Kim Culhan wrote:
> > If so, what do see if you run 'vmstat 1' and let it run for about
> > twenty seconds?  Do you see the cpu utilization going to about 100%
> > every five or six seconds?
>
> Negative:

That's interesting; so that can potentially narrow the problematic code down 
to any bits specific to Linux and not BSD.  This is very helpful!

Thank you Richard for thinking to ask this, and thank you Kim for responding!

-A.
___
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] Zaptel and Boostringer

2005-04-30 Thread Andrew Kohlsmith

On April 30, 2005 02:56 pm, Ian Pattison wrote:
> For some time now I've had issues with ringing voltages on my TDM400P.
> Numerous folks have told me that using "modprobe wcfxs boostringer=1"  when
> loading the module will force the driver to use boosted ring voltage. For
> some strange reason this has never worked for me. Today I got creative...
> another way to do it is to edit wcfxs.c (in the zaptel CVS) and find the
> following block of declarations:

It would do you well to figure out WHY your system is not passing parameters 
properly; 'boostringer=1' is supposed to set that boostringer variable.  The 
fact that it isn't indicates a deeper seated problem with your kernel, 
modprobe, or your distribution.

-A.
___
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] SIP over IAX2

Asterisk Box 2 (agents register)
extensions.conf
[agents-context]
exten => 1234,1,Dial(SIP/[EMAIL PROTECTED])
exten => 1234,2,Hangup

Asterisk Box 1
Sip.conf
[ab1]
type=friend
host=
context=incoming
canreinvite=yes
qualify=yes

extension.conf
[incoming]
Exten => 1234etc...

-Original Message-
From: Daniel Salama [mailto:[EMAIL PROTECTED] 
Sent: Saturday, April 30, 2005 6:50 PM
To: Tim Connolly
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SIP over IAX2

I understand and I guess I know how to do that within a single box.

If I have the following:

Asterisk Box 1 (no agents)
extensions.conf
[test-ivr]
exten => s,1,AGI(play_ivr)
exten => s,2,Hangup

Asterisk Box 2 (agents register)
extensions.conf
[agents-context]
exten => 1234,1,Dial(?)
exten => 1234,2,Hangup

Question is, when the agents dial 1234, how do I tell the application 
to connect to the agent with context test-ivr of Asterisk_1?

Thanks,
Daniel

On Apr 30, 2005, at 7:12 PM, Tim Connolly wrote:

> Maybe I'm missing something, but as long as you have the entension 
> defined
> on the agent box to dial the extension on the IVR, you should be okay. 
> Just
> make sure the default SIP context on the IVR has that extension 
> defined, or
> define the IVR box as a SIP peer.
>
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Daniel 
> Salama
> Sent: Saturday, April 30, 2005 5:57 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] SIP over IAX2
>
> I have two asterisk boxes. I'm running an IVR script in one of them and
> I have agents registered on the second box.
>
> I wish to create an extension on the * box where the agents are
> registered, so that when dialed, it will connect the agent to the IVR
> script on the other * box. However, I'd like for the connection to be
> done using SIP instead of IAX. Can anyone help me, if at all possible,
> write this configuration?
>
> Thanks,
> Daniel
>
> ___
> 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


___
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

[Asterisk-Users] Programing a call forward feature to cel phones

Guys.

I just programed a feature that allows any extension to be forwarded to any
outside number, for example, forward your extension 201 to any number
outside (via zap) so that if somebody calls your extension either from
inside out outside (using another zap we have) it gets directed.

Problem I have is that if somebody using a cel phone calls in and gets
directed to my extension which in turn is directed to my cel phone, the call
comes thru but after 2 seconds, the call gets all garbled and with a sound
like b and the caller cant be heard anymore.

Anybody has any idea why this is happening?


___
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] SIP over IAX2

I understand and I guess I know how to do that within a single box.
If I have the following:
Asterisk Box 1 (no agents)
extensions.conf
[test-ivr]
exten => s,1,AGI(play_ivr)
exten => s,2,Hangup
Asterisk Box 2 (agents register)
extensions.conf
[agents-context]
exten => 1234,1,Dial(?)
exten => 1234,2,Hangup
Question is, when the agents dial 1234, how do I tell the application 
to connect to the agent with context test-ivr of Asterisk_1?

Thanks,
Daniel
On Apr 30, 2005, at 7:12 PM, Tim Connolly wrote:
Maybe I'm missing something, but as long as you have the entension 
defined
on the agent box to dial the extension on the IVR, you should be okay. 
Just
make sure the default SIP context on the IVR has that extension 
defined, or
define the IVR box as a SIP peer.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Daniel 
Salama
Sent: Saturday, April 30, 2005 5:57 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] SIP over IAX2

I have two asterisk boxes. I'm running an IVR script in one of them and
I have agents registered on the second box.
I wish to create an extension on the * box where the agents are
registered, so that when dialed, it will connect the agent to the IVR
script on the other * box. However, I'd like for the connection to be
done using SIP instead of IAX. Can anyone help me, if at all possible,
write this configuration?
Thanks,
Daniel
___
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
___
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

[Asterisk-Users] How to bridge 2 calls

Guys.

I have some dialing rules defined for my internal extensions but I am now
defning a call forward option that allow an extension to be forwarded to an
outside number, right now Im using Dial cmds but I was wondering if ther is
a way to do this but using the dialing rules that I have also defined for
the internal extensions? For exaple, like DISA does...

Any ideas?

___
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] Kernel 2.4 or 2.6

2005-04-30 Thread Ian Pattison

I believe it is Zaptel only that becomes a problem. I'm running asterisk on a 
2.6 kernel... the only concession I had to make was to use "make linux26" when 
I compiled Zaptel.

Thanks,

Ian

>>> [EMAIL PROTECTED] 30/04/2005 19:10 >>>
I was reading on the wiki about the supported kernels and I __THINK__ 
the main issues with the kernel versions have more to do with Zaptel 
driver and not necessarily Asterisk itself. Is this correct?

Thanks,
Daniel

___
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

___
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] budgetphone

2005-04-30 Thread Michiel van Baak

On 01:11, Sun 01 May 05, Bert Haverkamp wrote:
> Dear all,
> 
> I'm trying to get asterisk to register to budgetphone.nl. In several threads
>  I saw people who got this to work:
> http://www.voip-info.org/wiki-Talkin2ya

Hi,

The directions on the page there are working like a charm.
The one who made this page helped me too. We did this setup
at the same time, and I was able to get 50% done without his
help and after his help all worked fine.

Did you setup the /etc/hosts file ?
That was the thing I needed to do to get it all working.
My config now is (ip's and telephone numbers replaced with
bogus values):

/etc/hosts:
# Host Database
81.23.228.150   budgetphone.nl

/etc/asterisk/sip.conf:
[general]
context=from-sip ; Default context for incoming calls
realm=vanbaak   ; Realm for digest authentication
port=5060   ; UDP Port to bind to (SIP standard port is 
5060)
bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes   ; Enable DNS SRV lookups on outbound calls
maxexpirey=3600 ; Max length of incoming registration we allow
defaultexpirey=120  ; Default length of incoming/outoing 
registration
allow=all   ; This may also be set for individual 
users/peers
language=en ; Default language setting for all users/peers
relaxdtmf=yes   ; Relax dtmf handling
rtptimeout=60   ; Terminate call if 60 seconds of no RTP 
activity
rtpholdtimeout=300  ; Terminate call if 300 seconds of no RTP 
activity
useragent=Asterisk  ; Allows you to change the user agent string
nat=no  ; NAT settings 
externip=XXX.XXX.XXX.XXX
localnet=192.168.2.0/255.255.255.0
promiscredir = no  ; If yes, allows 302 or REDIR to non-local SIP address
register => 31X:[EMAIL PROTECTED]/31X

[budgetphone]
canreinvite=no
context=from-budgetphone
fromuser=31
fromdomain=budgetphone.nl
host=budgetphone.nl
insecure=very
nat=yes
;qualify=yes
secret=my_passwd
type=friend
username=31X

/etc/asterisk/extensions.conf:
[general]
static=yes
writeprotect=no
[globals]
VMBOX=michiel ; the VM box

[outgoing-budgetphone]
exten => _0X,1,SetAccount(outgoing-budgetphone)
exten => _0X,2,SetCallerID(31X)
exten => _0X,3,SetCIDName(Michiel en Nancy van Baak)
exten => _0X,4,SetCIDNum(31)
exten => _0X,5,Dial(SIP/budgetphone/${EXTEN},50,Tr)
exten => _0X,6,Congestion
exten => _0X,106,Busy

[from-budgetphone]
exten => 31X,1,SetCIdNum(0${CALLERIDNUM:2})
exten => 31X,2,LookupCIDName
exten => 31X,3,Macro(stdexten,michiel,SIP/michiel)


We are using this setup for 3 months now and the KPN line is
already history.

-- 
Michiel van Baak
http://lunteren.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D

"Two of the most famous products of Berkeley are LSD and BSD. I don't think 
that this is a coincidence."

___
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

[Asterisk-Users] budgetphone

2005-04-30 Thread Bert Haverkamp

Dear all,

I'm trying to get asterisk to register to budgetphone.nl. In several threads
 I saw people who got this to work:
http://www.voip-info.org/wiki-Talkin2ya
http://lists.digium.com/pipermail/asterisk-users/2005-March/092850.html

But I've spent a whole saturday on it now and didn't get any further.
I also have a granstrema handytone 486. This thing manages to register.

I've tried to look into the differences between the sip messages with
ethereal. In ethereal I see the following sip conversation for the handytone:
192.168.0.60->81.23.228.150   SIP Request: REGISTER sip:budgetphone.nl
81.23.228.150->192.168.0.60  SIP Status: 401 Unauthorized(0 bindings)
192.168.0.60->81.23.228.150  SIP Request: REGISTER sip:budgetphone.nl
81.23.228.150->192.168.0.60  SIP Status: 200 OK(1 bindings)

In the first message the handytone tries to register, but it gets a request
for authentication (second packet) with a challenge. The third packet is a
retry to register, but this time with the response to the challenge. The
fourth packet is then the confirmation that all went well.

When I do the same with asterisk I get the following
192.168.0.35->81.23.228.150   SIP Request: REGISTER sip:budgetphone.nl
81.23.228.150->192.168.0.35  SIP Status: 401 Unauthorized(0 bindings)
192.168.0.35->81.23.228.150  SIP Request: REGISTER sip:budgetphone.nl
192.168.0.35->81.23.228.150  SIP Request: REGISTER sip:budgetphone.nl
192.168.0.35->81.23.228.150  SIP Request: REGISTER sip:budgetphone.nl
...
Asterisk gives a response to the challenge, but never gets an answer back.
What is going wrong?
Hope someone can shed some light here..

; SIP Configuration for Asterisk
;
[general]
context=default ; Default context for incoming calls
recordhistory=yes ; Record SIP history by default
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
language=en ; Default language setting for all users/peers
nat=no
defaultexpirey=1200
disallow=all
allow=g729
allow=gsm
allow=ulaw
allow=alaw

register => 31437110310:[EMAIL PROTECTED]/31437110310

[31437110310]
type=friend
context=from-budgetphone
host=budgetphone.nl
callerid="John Doe"
fromuser=31437110310
fromdomain=budgetphone.nl
username=31437110310
insecure=very
secret=PASSWD
qualify=no
canreinvite=no
nat=yes
port=5060

---
___
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

[Asterisk-Users] Kernel 2.4 or 2.6

I was reading on the wiki about the supported kernels and I __THINK__ 
the main issues with the kernel versions have more to do with Zaptel 
driver and not necessarily Asterisk itself. Is this correct?

Thanks,
Daniel
___
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

[Asterisk-Users] SIP over IAX2

I have two asterisk boxes. I'm running an IVR script in one of them and 
I have agents registered on the second box.

I wish to create an extension on the * box where the agents are 
registered, so that when dialed, it will connect the agent to the IVR 
script on the other * box. However, I'd like for the connection to be 
done using SIP instead of IAX. Can anyone help me, if at all possible, 
write this configuration?

Thanks,
Daniel
___
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] A good SIP receptionist phone

2005-04-30 Thread Rusty Shackleford


> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Michael Welter
> Sent: Saturday, April 30, 2005 12:53 PM
> To: [EMAIL PROTECTED]; Asterisk Users Mailing List - 
> Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] A good SIP receptionist phone
> 
> In a multi-tenant environment, is there a way to display, on 
> the phone, 
> which DID (which tenant) is being called?


Yes. We've done this by simply prepending a meaningful string onto the
front of the CIDName. It's a total kludge, but it works.

-- 
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.308 / Virus Database: 266.11.0 - Release Date: 04/29/2005
 

___
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

[Asterisk-Users] Problem with Sangoma/Adtran 600 installation

2005-04-30 Thread Chris Mason (Lists)

I have installed Asterisk on a CentOS4 box and then installed Asterisk from
CVS.
I installed a Sangoma A101 and connected it to a Adtran 600 using a T1
Crossover cable. The 600 has 12 x FXS, 12 x FXO interfaces.

I ran through the wanpipe install instructions and configured it, now I can
run 

[EMAIL PROTECTED] asterisk]#  wanrouter hwprobe

---
| Wanpipe Hardware Probe Info |
---
1 . AFT-A101u  : SLOT=1 : BUS=1 : IRQ=209 : CPU=A : PORT=PRI

Card Cnt: S508=0  S514X=0  S518=0  A101-2=1  A104=0  A300=0 

So I know the card is there OK.

My /etc/zaptel.conf looks like:
span=1,1,0,esf,b8zs
loadzone = us
defaultzone=us
fxsls=1-12

I am only trying to get half to load for now to make it simple.

[EMAIL PROTECTED] asterisk]# ztcfg
Notice: Configuration file is /etc/zaptel.conf
line 0: Unable to open master device '/dev/zap/ctl'

1 error(s) detected

When I run "service zaptel restart" I get:

Waiting for zap to come online...Error: missing /dev/zap!

Wha am I doing wrong?


Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (646)722-0001 Fax: (815)301-9759 
(305) 704-7249
Yahoo IM: [EMAIL PROTECTED] 
Skype ID: netconcepts

___
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

[Asterisk-Users] Problem with PSTN

2005-04-30 Thread Salina Jain

Hi,
I am a new user of Linux and Asterisk. I bought Digium TDM400P card and now 
want to setup my dial plan. With some help from the suggestions given online 
I have been able to configure the two SIP phones to interact with each 
other.

I want to use this to call on to a Telecom line(PSTN) and vice versa. I read 
somewhere that we need to use some provider for it like FWD or iconnect, do 
we need to use them to make outgoing and incoming calls to PSTN lines or we 
can do it without them. I can post my .conf files if anybody needs them to 
help me out with this. I don't know what should I put in the .conf files so 
that it enables these calls.

Any amount of help or suggestions would really be appreciated.
Thanx,
Salina
_
News, views and gossip. http://www.msn.co.in/Cinema/ Get it all at MSN 
Cinema!

___
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

[Asterisk-Users] Asterisk on Radio Tonight

2005-04-30 Thread Kerry Garrison

Kerry Garrison from The Geek Gazette (http://geekgazette.com)  will be
interviewed tonight on Mick Mick Williams' Cyber Line radio program at
9:00PM PST. The show is broadcast on the USA Radio network. If you do not
have a channel in your area, you can listen listen live online
. The show will cover the basic of
what the Asterisk PBX is all about and what it takes to implement a system.

-Kerry


___
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] A good SIP receptionist phone

2005-04-30 Thread Chris Mason (Lists)

> Take a look at the Polycom IP 600

I just added one to my desk as a test unit, I can't image you would need
anything more. We have Mitel 4015/4025/Superset for the office pbx I will be
replacing with *, and the Polycom 600 is a much better unit by far.

Chris Mason
www.anguillaguide.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] Pattern Matching









Like this:

 

[dids]

Exten => 2145550001,1,dial(SIP/6001)

Exten => 2145550002,1,dial(SIP/6002)

Exten => 2145550003,1,dial(SIP/6003)

Include => default-did

 

[default-did]

Exten => _.,1,dial(SIP/6000)

 

 

Seems pretty simple. I used this method of least/highest cost routing
to choose my LD carrier. Should work the same though.

 

 

http://www.voip-info.org/tiki-index.php?page=Asterisk%20least%20cost%20routing%20using%20broadvoice

 

 

 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mojo Jojo
Sent: Saturday, April 30, 2005 3:08 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Pattern Matching

 

Not sure what you mean exactly... Can you give me a hint?

 

 

Private Label Wholesale Internet Access!

http://www.YourOwnISP.com

 

- Original Message - 

From: "Michael D Schelin" <[EMAIL PROTECTED]>

To: "Asterisk Users Mailing List - Non-Commercial Discussion"




Sent: Friday, April 29, 2005 10:10 PM

Subject: Re: [Asterisk-Users] Pattern Matching

 

 

> Hey Mojo, I'm thinking you might try using priorty 's to set some
kind 

> routing. just a thought..

> 

> 

> 

> Mojo Jojo wrote:

> 

>> We recently had our PRI installed, we currently have 100
toll-free's 

>> pointing to it.

>> 

>> I have almost everything working great but..

>> 

>> I have setup the first few numbers we want to use coming in
from the PRI 

>> and they work great, but..

>> 

>> What I want to do is setup an extension with pattern matching
to answer 

>> for any numbers called that are pointed to our system and PRI
but not yet 

>> in use/configured.

>> 

>> I have been successful at setting up pattern matching as a
catch all for 

>> 98 or so numbers not in use yet and I have been successful
setting up the 

>> 2 numbers I want to make use of for now.

>> 

>> Problem is, I can't use both at the same time!

>> 

>> If I turn on the pattern matching then my greeting plays for
the 

>> configured number, then the message plays for the invalid
number 

>> (basically executing the extension with the pattern matching).

>> 

>> I have read about sorting with pattern matching by using an
include, I 

>> did this but it's not really helping.

>> 

>> I have set a response timeout after the first extension plays
it's 

>> greeting, I would think it should wait until it times out but
it doesn't, 

>> it just immediately moves to the pattern matched extension.

>> 

>> I must be missing something big here..

>> 

>> Any help is appreciated..

>> 

>> 

>> -- 

>> Private Label Wholesale Internet Access!

>> http://www.YourOwnISP.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

>> 

>> 

> 

> ___

> 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

> 

 

 

___

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






___
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] A good SIP receptionist phone

Can you show us an example of using the callerID for this purpose?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Florian
Overkamp
Sent: Saturday, April 30, 2005 3:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] A good SIP receptionist phone

Hi,

Citeren Michael Welter <[EMAIL PROTECTED]>:

> In a multi-tenant environment, is there a way to display, on the phone,
> which DID (which tenant) is being called?

We use the callerID name for that purpose.

Florian
___
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

___
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

[Asterisk-Users] Call-park timeouts..









    If a person parks a call, the call hits the
timeout exten for that context after the park expires.. Is there any way to
make it ring back to the person who parked the call instead of using the
timeout?

 






___
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

[Asterisk-Users] Hotel CDR Software

Guys.

Anybody knows of asterisk compliant cdr software for Hotel that will let you
enter diff. rates, checkin and out that will create the extension and setup
voicemail for the room, etc?


___
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

[Asterisk-Users] Install Asterisk on CCM MCS-7835 Server

2005-04-30 Thread Walid Azab


Hi All,

I am replacing Cisco Call Manager with Asterisk. As you know CCM
is on a MCS 7835 Server which comes with a custom version of
Windows. Does any one know how to install Linux on that H/W. My
guess is that someone must have tried the same thing before. I
know how to install Linux however I cannot get passed the H/W
limitation. 



Walid
___
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] help with compiling addons for cdr

2005-04-30 Thread Jim Sturtevant

:-) ok... so I feel foolish... Mathew thanks a lot, worked like a charm.

Jim


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm
Sent: Saturday, April 30, 2005 11:16 AM
To: Asterisk Users
Subject: Re: [Asterisk-Users] help with compiling addons for cdr

Well for some reason, you decided to use the stable version of asterisk but
also decided not to use the stable version of addons. Hmm...interesting
decisions.

rm -rf asterisk-addons/
cvs co addons -r v1.0.7

Then it will work.

-Matthew

> From: forums <[EMAIL PROTECTED]>
> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Date: Sat, 30 Apr 2005 10:24:24 -0700
> To: 
> Subject: [Asterisk-Users] help with compiling addons for cdr
> 
> I'm running Asterisk 1.0.7.  I've checked out using "cvs checkout
> asterisk-addons".
> 
>  
> 
> When I make install I get the following errors:
> 
>  
> 
> app_addon_sql_mysql.c:162:36: macro "AST_LIST_REMOVE" requires 4
arguments,
> but only 3 given
> 
>  
> 
> I'm using the default FC3 mysql:
> 
>  
> 
> mysql-server-3.23.58-16.FC3.1
> 
> perl-DBD-MySQL-2.9003-5
> 
> mysql-3.23.58-16.FC3.1
> 
> mysql-devel-3.23.58-16.FC3.1
> 
> php-mysql-4.3.11-2.4
> 
> libdbi-dbd-mysql-0.6.5-9
> 
> MySQL-python-0.9.2-4
> 
>  
> 
> I've search wiki, etal and have found a couple of references with a
proposed
> patch file, but the patch file fails too.
> 
>  
> 
> I would appreciate any assistance.
> 
>  
> 
>  
> 
> ___
> 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


___
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

___
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] Cisco 7960s and skinny

Thank you for the detailed description Andy.

Please let me know how about the specs when you can.

My client has legacy 7610 but I am trying to suggest swithcing to native sip
phones like grandstream or better in order to make everything 100% asterisk
compliant.

Plus, Cisco charging for the sip images and such ($150) doesnt look good,
for that price you can get some SIP phones.

 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Andy Hamilton
|Sent: Sábado, 30 de Abril de 2005 01:00 p.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] Cisco 7960s and skinny
|
|Anton:
|
|I'll be able to get back to you Sunday night about specifics; 
|the phone is not where I am right now. Using chan_sccp, (I 
|think November
|2004 or so CVS Head) I know I can receive calls, place calls, 
|etc. It is a rather low volume phone, so I don't know off hand 
|about specific keys; I'll check those later.
|Additionally, I have not yet tried a new copy from CVS.
|
|Occasionally, I think the chan_sccp driver blips out in 
|Asterisk (it may be the phone; I've had it apart several times 
|because the on/off hook switch membrane is a little sketchy). 
|I have dealt with this by restarting Asterisk. The only other 
|thing I can say right now about the 7910 is that it and my 
|Cisco FastHub don't get along. At all. I have the 7910 plugged 
|into my 7960.
|
|Overall, I would say that if you have a non-critical system 
|and would like to use a 7910, chan_sccp should be able to 
|handle it fine. 
|However, if you budget permits, the 7960 and 7940 phones are 
|quite nice (use SIP with those -- it's far more reliable. I 
|must say, though, that my 7960 has frozen/crashed a handful of 
|time when running the SIP image. That was the phone itself, 
|Asterisk was fine.) I have yet to purchase a 7905 or 7912, but 
|I've played around with some 7912's on a CCM system -- they 
|seem quite nice and I think they take SIP. The 7920 is also 
|nice because it's wireless. However, I don't think Cisco has 
|anything but a Skinny image for it [yet].
|
|I would stick with SIP wherever you can.
|
|-Andy
|
|
|
|On 4/30/05, Anton Krall <[EMAIL PROTECTED]> wrote:
|> Andy
|> 
|> How did the 7910 worked with skinny under *? Did all the keys on the 
|> phone worked? Ive seen sometimes the forward key or 
|something does not 
|> fully do what you would excpect.
|> 
|> What are the drawbacks from using skinny vs sip under *?
|>
|___
|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
|
|

___
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

[Asterisk-Users] Send DTMF AFTER channels are bridged

2005-04-30 Thread Shady

Someone to know how can I send a DTMF after the channels are bridged?
I need something like the "D" option of the Dial application, but this
option sends the DTMF before the channels are bridged. In fact I want the
caller and the callee to receive the DTMF. Please help :)


___
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] Pattern Matching

2005-04-30 Thread Mojo Jojo

Not sure what you mean exactly... Can you give me a hint?
Private Label Wholesale Internet Access!
http://www.YourOwnISP.com
- Original Message - 
From: "Michael D Schelin" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Friday, April 29, 2005 10:10 PM
Subject: Re: [Asterisk-Users] Pattern Matching

Hey Mojo, I'm thinking you might try using priorty 's to set some kind 
routing. just a thought..

Mojo Jojo wrote:
We recently had our PRI installed, we currently have 100 toll-free's 
pointing to it.

I have almost everything working great but..
I have setup the first few numbers we want to use coming in from the PRI 
and they work great, but..

What I want to do is setup an extension with pattern matching to answer 
for any numbers called that are pointed to our system and PRI but not yet 
in use/configured.

I have been successful at setting up pattern matching as a catch all for 
98 or so numbers not in use yet and I have been successful setting up the 
2 numbers I want to make use of for now.

Problem is, I can't use both at the same time!
If I turn on the pattern matching then my greeting plays for the 
configured number, then the message plays for the invalid number 
(basically executing the extension with the pattern matching).

I have read about sorting with pattern matching by using an include, I 
did this but it's not really helping.

I have set a response timeout after the first extension plays it's 
greeting, I would think it should wait until it times out but it doesn't, 
it just immediately moves to the pattern matched extension.

I must be missing something big here..
Any help is appreciated..
--
Private Label Wholesale Internet Access!
http://www.YourOwnISP.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

___
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

___
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] A good SIP receptionist phone

2005-04-30 Thread Florian Overkamp

Hi,

Citeren Michael Welter <[EMAIL PROTECTED]>:

> In a multi-tenant environment, is there a way to display, on the phone,
> which DID (which tenant) is being called?

We use the callerID name for that purpose.

Florian
___
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] A good SIP receptionist phone

2005-04-30 Thread Florian Overkamp

Hi,

Citeren Jason Brown <[EMAIL PROTECTED]>:

> I need a good receptionist phone that works with Asterisk. It basically needs
> to act like an avaya partner phone, I don't need 20 buttons with little red
> LED's...what I do need is for the phone to register multiple extensions to my
> asterisk server and act like each SIP extension is a line, so if the idiot
> receptionist has a call ringing in on line 1, she can pick it up, look at the
> buttons, see a call ringing in on line 2 (and the phone ringer rings), put
> call 1 on hold without hanging the caller up, and hit the little "I am an
> idiot and need a line 2 button" to pick up line 2, so on and so forth.

Take a look at the SNOM220 phone. They come with an optional side panel to add
line or speed dial keys.

Florian
___
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] A good SIP receptionist phone

2005-04-30 Thread Michael Welter

So seriously does anyone have a recommendation for a good receptionist phone? I tried the Snom today and I can't get the programmable buttons to do this, even by following the manual. So please, any suggestions would be great, before I get fired at my dayjob for everyone else's idiocy.

How many lines do you need?
The Cisco 7960 gives you 6, with call waiting you can get 2 calls on
each line.
You have to trade off between incoming lines and speed dials, unless you
can train the monkey^w receptionist (sorry, unfair to simians there) to
use the directories.
Seriously, you may need to look deeper here on the human side. Could
this be a people problem in that the receptionist does not want to
learn/is a friend/relative of a PBX supplier who is being usurped?  Have
you made an enemy of this person?
We have just switched over to Asterisk with 7960s. We have had a few
little problems but have not lost a call yet.  OK, we have left a few
callers on hold a bit longer than we intended, once or twice ;)
In a multi-tenant environment, is there a way to display, on the phone, 
which DID (which tenant) is being called?

___
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] A good SIP receptionist phone

2005-04-30 Thread Mike Clark

Jason Brown wrote:
I have a problem. The average person is too freaking stupid to use a VOIP
phone. My experience has so far been that if it doesn't have 20 buttons with
little red LED's on it, the user cannot comprehend call parking, attended
transfer, blind transfer, DND, and navigating through a voicemail menu.
I need a good receptionist phone that works with Asterisk. It basically needs to act like
an avaya partner phone, I don't need 20 buttons with little red LED's...what I do need is
for the phone to register multiple extensions to my asterisk server and act like each SIP
extension is a line, so if the idiot receptionist has a call ringing in on line 1, she
can pick it up, look at the buttons, see a call ringing in on line 2 (and the phone
ringer rings), put call 1 on hold without hanging the caller up, and hit the little
"I am an idiot and need a line 2 button" to pick up line 2, so on and so forth.
I love VOIP systems and all the functionality they bring and features I get.
Unfortunately, the average person in this country anymore is apparently
completely stupid and cannot understand how to juggle calls without hanging up
on people.

So seriously does anyone have a recommendation for a good receptionist phone? I tried the Snom today and I can't get the programmable buttons to do this, even by following the manual. So please, any suggestions would be great, before I get fired at my dayjob for everyone else's idiocy.

Take a look at the Polycom IP 600
Mike Clark
___
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] A good SIP receptionist phone

2005-04-30 Thread Ron Wellsted

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Jason Brown wrote:
> I have a problem. The average person is too freaking stupid to use a VOIP 
> phone. My experience has so far been that if it doesn't have 20 buttons with 
> little red LED's on it, the user cannot comprehend call parking, attended 
> transfer, blind transfer, DND, and navigating through a voicemail menu.
> 
> I need a good receptionist phone that works with Asterisk. It basically needs 
> to act like an avaya partner phone, I don't need 20 buttons with little red 
> LED's...what I do need is for the phone to register multiple extensions to my 
> asterisk server and act like each SIP extension is a line, so if the idiot 
> receptionist has a call ringing in on line 1, she can pick it up, look at the 
> buttons, see a call ringing in on line 2 (and the phone ringer rings), put 
> call 1 on hold without hanging the caller up, and hit the little "I am an 
> idiot and need a line 2 button" to pick up line 2, so on and so forth.
> 
> I love VOIP systems and all the functionality they bring and features I get. 
> Unfortunately, the average person in this country anymore is apparently 
> completely stupid and cannot understand how to juggle calls without hanging 
> up on people.
> 
> 
> 
> So seriously does anyone have a recommendation for a good receptionist phone? 
> I tried the Snom today and I can't get the programmable buttons to do this, 
> even by following the manual. So please, any suggestions would be great, 
> before I get fired at my dayjob for everyone else's idiocy.

How many lines do you need?

The Cisco 7960 gives you 6, with call waiting you can get 2 calls on
each line.

You have to trade off between incoming lines and speed dials, unless you
can train the monkey^w receptionist (sorry, unfair to simians there) to
use the directories.

Seriously, you may need to look deeper here on the human side. Could
this be a people problem in that the receptionist does not want to
learn/is a friend/relative of a PBX supplier who is being usurped?  Have
you made an enemy of this person?

We have just switched over to Asterisk with 7960s. We have had a few
little problems but have not lost a call yet.  OK, we have left a few
callers on hold a bit longer than we intended, once or twice ;)

- --
Ron Wellsted
http://www.wellsted.org.uk
[EMAIL PROTECTED]
FWD:519961  Gossiptel:9309811
N 52.567623, W 2.137621
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.1 (GNU/Linux)
Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org

iQEVAwUBQnPdF0tP/KMNOfRbAQJYoggAlg84sltFMmvPrD8AZ1xW5xjXuYBMzKU0
5nGDOvqUfiBSJHJrPw1bm+vvr0SdQK0jMJRVutnnLO6T//RG0qSjKT3NQoWKyY/u
y10PhMnh56+yPd5JcTv6194IKFXMxwRvp+U1K5Zr/nccuXMjDsacYnrZM32Duo8s
zl6ISF/cw6fydzhBKaAvCPa4+oYs7GCRfhaGD5cx21nzPW6bPDnVQJlBzjvn46UX
SDbRii3xne86lTrIVkgZnBPmtGPiWwG8epIhaorMqiB0gw//9M5rO39MPgKuCvbu
GaiYaLhXGsPfC3d2Q4EPdzx6HbhSHMkj4veU4bvXGxJRSloHnqG42w==
=j6r7
-END PGP SIGNATURE-
___
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] A good SIP receptionist phone

2005-04-30 Thread Mike Dent

On 4/30/05, Jason Brown <[EMAIL PROTECTED]> wrote:
> I have a problem. The average person is too freaking stupid to use a VOIP 
> phone. My experience has so far been that if it doesn't have 20 buttons with 
> little red LED's on it, the user cannot comprehend call parking, attended 
> transfer, blind transfer, DND, and navigating through a voicemail menu.
> 
> I need a good receptionist phone that works with Asterisk. It basically needs 
> to act like an avaya partner phone, I don't need 20 buttons with little red 
> LED's...what I do need is for the phone to register multiple extensions to my 
> asterisk server and act like each SIP extension is a line, so if the idiot 
> receptionist has a call ringing in on line 1, she can pick it up, look at the 
> buttons, see a call ringing in on line 2 (and the phone ringer rings), put 
> call 1 on hold without hanging the caller up, and hit the little "I am an 
> idiot and need a line 2 button" to pick up line 2, so on and so forth.
> 
> I love VOIP systems and all the functionality they bring and features I get. 
> Unfortunately, the average person in this country anymore is apparently 
> completely stupid and cannot understand how to juggle calls without hanging 
> up on people.
> 
> 
> 
> So seriously does anyone have a recommendation for a good receptionist phone? 
> I tried the Snom today and I can't get the programmable buttons to do this, 
> even by following the manual. So please, any suggestions would be great, 
> before I get fired at my dayjob for everyone else's idiocy.
> 
> 

Hi,
the Cisco 7960 (6 SIP lines) or the 7940 (2 lines) does what you wan, i think!
I have one here which is registered with 6 different extensions on my * box.
I can switch between calls on different buttons.

Mike

> ___
> 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
> 
>
___
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] A good SIP receptionist phone

2005-04-30 Thread Eric Wieling aka ManxPower

Jason Brown wrote:
So seriously does anyone have a recommendation for a good receptionist phone? I tried the Snom today and I can't get the programmable buttons to do this, even by following the manual. So please, any suggestions would be great, before I get fired at my dayjob for everyone else's idiocy.
My suggestion is to get a good receptionist.  The receptionists at my 
customers are consistantly more technology oriented than other 
employees.

--
Always do right. This will gratify some people and astonish the rest.
Mark Twain
___
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

[Asterisk-Users] A good SIP receptionist phone

2005-04-30 Thread Jason Brown

I have a problem. The average person is too freaking stupid to use a VOIP 
phone. My experience has so far been that if it doesn't have 20 buttons with 
little red LED's on it, the user cannot comprehend call parking, attended 
transfer, blind transfer, DND, and navigating through a voicemail menu.

I need a good receptionist phone that works with Asterisk. It basically needs 
to act like an avaya partner phone, I don't need 20 buttons with little red 
LED's...what I do need is for the phone to register multiple extensions to my 
asterisk server and act like each SIP extension is a line, so if the idiot 
receptionist has a call ringing in on line 1, she can pick it up, look at the 
buttons, see a call ringing in on line 2 (and the phone ringer rings), put call 
1 on hold without hanging the caller up, and hit the little "I am an idiot and 
need a line 2 button" to pick up line 2, so on and so forth.

I love VOIP systems and all the functionality they bring and features I get. 
Unfortunately, the average person in this country anymore is apparently 
completely stupid and cannot understand how to juggle calls without hanging up 
on people.



So seriously does anyone have a recommendation for a good receptionist phone? I 
tried the Snom today and I can't get the programmable buttons to do this, even 
by following the manual. So please, any suggestions would be great, before I 
get fired at my dayjob for everyone else's idiocy.
___
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

[Asterisk-Users] X-lite and * behind Firewalls

2005-04-30 Thread Julio Zavala









Hi All, i´m new in this list.

 

I have an Asterisk behind a firewall with forwarded ports
and my SIP clients is X-Lite.

 

In local connection we don’t have problem, and the
same with VPN connection. All work fine.

 

But when I try to connect to * from Internet or from others
LANs, the connection some time is successfully but the audio from * don’t
work.

 

ASTERISK  Firewall << Internet (SIP Client)
<<< Firewall <<< Other LAN (SIP Client)

 

I have NAT=Yes in each configuration of SIP Phone in SIP.CONF

I Have externip = asterisk.mydomain.com and internalip = 172.16.10.40

 

May be I have missed somes firewalls ports,
or my setup in asterisk isn’t complete…

 

Ports Forwarded in my firewall.

 

Tipo   Internal Server  Type  External/Internal
Port

SIP   172.16.10.40     *   
5060/5060

MGCP  172.16.10.40     UDP     2727/2727

X-Lite    172.16.10.40     UDP     3478/3478

xlite  172.16.10.40     UDP     8000/8000 


xlite 2   172.16.10.40     UDP     8001/8001 


SIP 2    172.16.10.40     UDP     5061/5061 


sip 3     172.16.10.40     *   631/631

 

Ports Opened in my firewall  

Allow UDP ASTERISK Wan,* LAN,172.16.10.40  FROM
WAN/LAN, 1-2

 

Cordialmente

 


 
  
  Julio Zavala A.
   
  
  
  
  Servicios Triactivos
  
 


 



_
Servicios Triactivos Limitada - www.triactivos.cl - Proyectos - Ingeniería - Servicios - Telefonía IP - Redes



<>___
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

[Asterisk-Users] Zaptel and Boostringer

2005-04-30 Thread Ian Pattison

Hi All,

For some time now I've had issues with ringing voltages on my TDM400P. Numerous 
folks have told me that using "modprobe wcfxs boostringer=1"  when loading the 
module will force the driver to use boosted ring voltage. For some strange 
reason this has never worked for me. Today I got creative... another way to do 
it is to edit wcfxs.c (in the zaptel CVS) and find the following block of 
declarations:

static int debug = 0;
static int robust = 0;
static int timingonly = 0;
static int lowpower = 0;
static int boostringer = 0;
static int _opermode = 0;
static char *opermode = "FCC";
static int fxshonormode = 0;

set boostringer=1 instead of 0 and recompile Zaptel. The FXS ports will be 
forced to generate 89V ring signals from now on.

Now if I can just stop the FXO ports from dropping calls

Thanks,

Ian

Ian Pattison, Senior Analyst
Technology Associates Inc.
Tel: 905-459-2100 ext. 204
Mobile: 416-568-6548
E-mail: [EMAIL PROTECTED] 
WWW: http://www.technologyassociates.ca

BEGIN:VCARD
VERSION:2.1
FN:Ian Pattison
EMAIL;WORK;PREF:[EMAIL PROTECTED]
TEL;WORK:416-657-2464 ext. 204
N:Pattison;Ian
TITLE:Senior Analyst
ADR;INTL;WORK;PARCEL;POSTAL:;;9052 Creditview Rd.;Brampton;Ontario;L6V 
1A1;Canada
LABEL;INTL;WORK;PARCEL;POSTAL;ENCODING=QUOTED-PRINTABLE:Ian Pattison=0A=
9052 Creditview Rd.=0A=
Brampton, Ontario  L6V 1A1=0A=
Canada
LABEL;DOM;WORK;PARCEL;POSTAL;ENCODING=QUOTED-PRINTABLE:Ian Pattison=0A=
9052 Creditview Rd.=0A=
Brampton, Ontario  L6V 1A1
TEL;CELL:416-568-6548
TEL;PREF:416-657-2464 ext. 204
TEL;WORK:905-459-2100 ext. 204
ORG:Technology Associates Inc.
END:VCARD

___
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] Grandstream Budgetone 100 Caller ID shows extension, not incoming Caller ID

2005-04-30 Thread Brian Capouch

Amit Sharma wrote:
Hi,
Is this a known problem with Grandstream Budgetone 100, I could see 
several people complaining about this but no answers.

Details
--
1) I am simply trying to go from one SIP extension to another, so the 
zapata.conf and zaptel.conf entries are irrelavant.
2) I added a NoOp(CALLERID=${CALLERID}), to my dial plan cand could see 
the Caller ID on the console, so asterisk is aware of the caller ID.
3) All ID's are simply numbers no fancy alphanumeric strings.

I have been looking for a solution for a quite some time and seem to 
have hit a wall, any pointers would be greatly appreciated.

The BT101 doesn't display alpha characters.
B.
___
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] transfer pstn call to voip line, thus freeing up pstn line

2005-04-30 Thread Eric Wieling aka ManxPower

Joseph wrote:
On Sat, 2005-04-30 at 13:23 -0400, Time Bandit wrote:
I was wondering if there was a way to have incoming
calls to my PSTN line be "transferred" to a voip line?
I would like to make it so that as soon as the pstn
call is recieved it will switch the call to the voip
line, thus freeing up the pstn line to get more calls.
Kind of like roaming.
If you have call transfer on your line, you can do it with somethin like 
this
(from the top of my head)
exten => s,1,Answer()
exten => s,2,Flash()
exten => s,3,SendDTMF(${MYVOIPNUM})
exten => s,4,Hangup()
Basically, you hook-flash the line (giving you a dialtone), compose
the number where you want the calls to be forwarded, then hangup the
line. The calling party will be connected to the destination and your
line will be free

Correct me anybody if I'm wrong; but I think this way he will only free
his internal extension not the PSTN line.
I think the only way of doing this is to order call forward feature form
his PSTN service provider, so when the call comes and his number is busy
it will automatically redirect it to let say DID number over IP (or any
other number). 

The better way would be to order Call Forward Busy Line from their 
telco.  In the example above when the hangup() happens both legs of 
the call will be disconneced.  The person should order 
Conference/Drop/Transfer service from his/her telco, rather than the 
traditional Three-Way Calling service if they really want to transfer 
the call.

--
Always do right. This will gratify some people and astonish the rest.
Mark Twain
___
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

[Asterisk-Users] ANNOUNCEMENT: Asterisk-java 0.1 released

2005-04-30 Thread Stefan Reuter

Asterisk-java 0.1 a Java control for the Asterisk PBX has been released.

The Asterisk-java package consists of a set of Java classes that allow
you to easily build Java applications that interact with an Asterisk
PBX Server. Asterisk-java supports both interfaces that Asterisk
provides for this scenario: The FastAGI  protocol and the Manager API.

The FastAGI implementation supports all commands currently available
from Asterisk.

The Manager API implementation supports receiving events from the
Asterisk server (e.g. call progess, registered peers, channel state)
and sending actions to Asterisk (e.g. originate call,
agent login/logoff, start/stop voice recording).

Asterisk-java is available under Apache 2.0 license at
http://asterisk-java.sourceforge.net



signature.asc
Description: This is a digitally signed message part
___
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] help with compiling addons for cdr

2005-04-30 Thread Matthew Boehm

Well for some reason, you decided to use the stable version of asterisk but
also decided not to use the stable version of addons. Hmm...interesting
decisions.

rm -rf asterisk-addons/
cvs co addons -r v1.0.7

Then it will work.

-Matthew

> From: forums <[EMAIL PROTECTED]>
> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Date: Sat, 30 Apr 2005 10:24:24 -0700
> To: 
> Subject: [Asterisk-Users] help with compiling addons for cdr
> 
> I'm running Asterisk 1.0.7.  I've checked out using "cvs checkout
> asterisk-addons".
> 
>  
> 
> When I make install I get the following errors:
> 
>  
> 
> app_addon_sql_mysql.c:162:36: macro "AST_LIST_REMOVE" requires 4 arguments,
> but only 3 given
> 
>  
> 
> I'm using the default FC3 mysql:
> 
>  
> 
> mysql-server-3.23.58-16.FC3.1
> 
> perl-DBD-MySQL-2.9003-5
> 
> mysql-3.23.58-16.FC3.1
> 
> mysql-devel-3.23.58-16.FC3.1
> 
> php-mysql-4.3.11-2.4
> 
> libdbi-dbd-mysql-0.6.5-9
> 
> MySQL-python-0.9.2-4
> 
>  
> 
> I've search wiki, etal and have found a couple of references with a proposed
> patch file, but the patch file fails too.
> 
>  
> 
> I would appreciate any assistance.
> 
>  
> 
>  
> 
> ___
> 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


___
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] transfer pstn call to voip line, thus freeing up pstn line

2005-04-30 Thread Joseph

On Sat, 2005-04-30 at 13:23 -0400, Time Bandit wrote:
> > I was wondering if there was a way to have incoming
> > calls to my PSTN line be "transferred" to a voip line?
> > 
> > I would like to make it so that as soon as the pstn
> > call is recieved it will switch the call to the voip
> > line, thus freeing up the pstn line to get more calls.
> > Kind of like roaming.
> If you have call transfer on your line, you can do it with somethin like this
> (from the top of my head)
> exten => s,1,Answer()
> exten => s,2,Flash()
> exten => s,3,SendDTMF(${MYVOIPNUM})
> exten => s,4,Hangup()
> 
> Basically, you hook-flash the line (giving you a dialtone), compose
> the number where you want the calls to be forwarded, then hangup the
> line. The calling party will be connected to the destination and your
> line will be free

Correct me anybody if I'm wrong; but I think this way he will only free
his internal extension not the PSTN line.
I think the only way of doing this is to order call forward feature form
his PSTN service provider, so when the call comes and his number is busy
it will automatically redirect it to let say DID number over IP (or any
other number). 

-- 
#Joseph
___
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] Group/Broadcast Voicemail

2005-04-30 Thread Eric Wieling aka ManxPower

in app_voicemail.c in the function vm_exec set the tmp[256] to be 
tmp[4096]

Chris Stinson wrote:
I have one with 33. but I can't get the voicemail to copy to more than 
20 mailboxes.

Eric Wieling aka ManxPower wrote:
Has anyone set up Group/Broadcast Voicemail for 50 or more mailboxes?
--
Always do right. This will gratify some people and astonish the rest.
Mark Twain
___
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] Cisco 7960s and skinny

2005-04-30 Thread Andy Hamilton

Anton:

I'll be able to get back to you Sunday night about specifics; the
phone is not where I am right now. Using chan_sccp, (I think November
2004 or so CVS Head) I know I can receive calls, place calls, etc. It
is a rather low volume phone, so I don't know off hand about specific
keys; I'll check those later.
Additionally, I have not yet tried a new copy from CVS.

Occasionally, I think the chan_sccp driver blips out in Asterisk (it
may be the phone; I've had it apart several times because the on/off
hook switch membrane is a little sketchy). I have dealt with this by
restarting Asterisk. The only other thing I can say right now about
the 7910 is that it and my Cisco FastHub don't get along. At all. I
have the 7910 plugged into my 7960.

Overall, I would say that if you have a non-critical system and would
like to use a 7910, chan_sccp should be able to handle it fine. 
However, if you budget permits, the 7960 and 7940 phones are quite
nice (use SIP with those -- it's far more reliable. I must say,
though, that my 7960 has frozen/crashed a handful of time when running
the SIP image. That was the phone itself, Asterisk was fine.) I have
yet to purchase a 7905 or 7912, but I've played around with some
7912's on a CCM system -- they seem quite nice and I think they take
SIP. The 7920 is also nice because it's wireless. However, I don't
think Cisco has anything but a Skinny image for it [yet].

I would stick with SIP wherever you can.

-Andy

On 4/30/05, Anton Krall <[EMAIL PROTECTED]> wrote:
> Andy
> 
> How did the 7910 worked with skinny under *? Did all the keys on the phone
> worked? Ive seen sometimes the forward key or something does not fully do
> what you would excpect.
> 
> What are the drawbacks from using skinny vs sip under *?
>
___
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] transfer pstn call to voip line, thus freeing up pstn line

2005-04-30 Thread Robert Webb

>
> I was wondering if there was a way to have incoming
> calls to my PSTN line be "transferred" to a voip line?
>
> I would like to make it so that as soon as the pstn
> call is recieved it will switch the call to the voip
> line, thus freeing up the pstn line to get more calls.
> Kind of like roaming.
>
> Tom
>

Why not just call forward everything to your Voip line and then run it
through *. Most all providers allow for at least two incoming calls at a
time. You would then have your PSTN line free for outgoing only and tie
it into a group with your Voip and save some outgoing VoIP minutes.

Robert

P.S. - This does work very well. It is what I am using at home with my
PSTN and myphonecompany.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] Amp extensions script

2005-04-30 Thread Robert Webb

>
> Hi, Is there a script in amp for adding the extensions?  And can it be
> modified?  When adding a new extension it rewrites all of the
> information it the context blowing out my additions.

You my want to try the AMP forum. Since they are the producers of AMP,
they may have a little better info.

http://sourceforge.net/forum/?group_id=121515



___
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

[Asterisk-Users] Amp extensions script

2005-04-30 Thread Michael D Schelin

Hi, Is there a script in amp for adding the extensions?  And can it be 
modified?  When adding a new extension it rewrites all of the 
information it the context blowing out my additions.
___
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

[Asterisk-Users] help with compiling addons for cdr

2005-04-30 Thread forums









I’m running Asterisk 1.0.7.  I’ve checked
out using “cvs checkout asterisk-addons”. 


 

When I make install I get the following
errors:

 

app_addon_sql_mysql.c:162:36: macro
"AST_LIST_REMOVE" requires 4 arguments, but only 3 given

 

I’m using the default FC3 mysql:

 

mysql-server-3.23.58-16.FC3.1

perl-DBD-MySQL-2.9003-5

mysql-3.23.58-16.FC3.1

mysql-devel-3.23.58-16.FC3.1

php-mysql-4.3.11-2.4

libdbi-dbd-mysql-0.6.5-9

MySQL-python-0.9.2-4

 

I’ve search wiki, etal and have
found a couple of references with a proposed patch file, but the patch file
fails too.  

 

I would appreciate any assistance.

 

 






___
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] transfer pstn call to voip line, thus freeing up pstn line

2005-04-30 Thread Time Bandit

> I was wondering if there was a way to have incoming
> calls to my PSTN line be "transferred" to a voip line?
> 
> I would like to make it so that as soon as the pstn
> call is recieved it will switch the call to the voip
> line, thus freeing up the pstn line to get more calls.
> Kind of like roaming.
If you have call transfer on your line, you can do it with somethin like this
(from the top of my head)
exten => s,1,Answer()
exten => s,2,Flash()
exten => s,3,SendDTMF(${MYVOIPNUM})
exten => s,4,Hangup()

Basically, you hook-flash the line (giving you a dialtone), compose
the number where you want the calls to be forwarded, then hangup the
line. The calling party will be connected to the destination and your
line will be free

hth
___
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

[Asterisk-Users] transfer pstn call to voip line, thus freeing up pstn line

2005-04-30 Thread Thomas Miller

I was wondering if there was a way to have incoming
calls to my PSTN line be "transferred" to a voip line?

I would like to make it so that as soon as the pstn
call is recieved it will switch the call to the voip
line, thus freeing up the pstn line to get more calls.
Kind of like roaming.

Tom

__
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.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

[Asterisk-Users] Intel 536EP

2005-04-30 Thread Jeff

Forgive me if this has been asked before, I wasn't able to find any
clear answers in the archives.
Will the Intel 536EP function as a FXO? And if so, do I need to use a
different version of the Zaptel driver?
Any assistance would be great.
PS - that's 536EP, not 537EP.
Thanks!
___
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

[Asterisk-Users] Chan_modem_*

2005-04-30 Thread Marco Supino

Hi,
I was looking for solutions for simple FXO cards, and came across the 
two modem channels in the asterisk channels/ dir, i assume they are 
there becuase someone made these two types of modems work as FXO (or are 
they there for other purpose ?),

does anyone have any info on these channels ? anyone has them working 
with any type of modem ? (aopen or bestdata).

Marco.
___
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

[Asterisk-Users] Grandstream Budgetone 100 Caller ID shows extension, not incoming Caller ID

2005-04-30 Thread Amit Sharma

Hi,
Is this a known problem with Grandstream Budgetone 100, I could see 
several people complaining about this but no answers.

Details
--
1) I am simply trying to go from one SIP extension to another, so the 
zapata.conf and zaptel.conf entries are irrelavant.
2) I added a NoOp(CALLERID=${CALLERID}), to my dial plan cand could see 
the Caller ID on the console, so asterisk is aware of the caller ID.
3) All ID's are simply numbers no fancy alphanumeric strings.

I have been looking for a solution for a quite some time and seem to 
have hit a wall, any pointers would be greatly appreciated.

Thanks,
Amit
sip.conf

[116]  ; Extension 1
type = friend context = sip-phone   username = 
116 fromuser = 116callerid = "116" <116>
host = 10.0.1.116  nat = no   
canreinvite=yes   dtmfmode = rfc-2833
[EMAIL PROTECTED] disallow=all   
allow=ulaw allow=alaw
  

  

[117]
type = friend  ; extension 2
context = sip-phoneusername = 117 fromuser = 
117 callerid = "117" <117>
host = 10.0.1.117 nat = no   
canreinvite=yes   dtmfmode = rfc-2833   
[EMAIL PROTECTED]   disallow=all 
allow=ulawallow=alaw
 extensions.conf
--
[macro-exten]
exten => s,1,NoOp(CALLERID=${CALLERID})
exten => s,2,Dial(SIP/${ARG1},20)
exten => s,3,Voicemail(u${ARG1})
exten => s,4,Hangup

[default]
exten => 116,1,Macro(exten,116)
exten => 117,1,Macro(exten,117)

___
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

[Asterisk-Users] Re: IPSwitchBoard version 0.111 released

2005-04-30 Thread tgj

You will find the URL on my download page

thorben

"TC" <[EMAIL PROTECTED]> skrev i en meddelelse 
news:[EMAIL PROTECTED]
> what is the url for the version of the framework it wants now ?
> the dot net auto installer is busted ?
> - Original Message -
> From: "Thorben Jensen" <[EMAIL PROTECTED]>
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
> 
> Sent: Saturday, April 30, 2005 8:39 AM
> Subject: [Asterisk-Users] IPSwitchBoard version 0.111 released
>
>
>> Version 0.111 - 30. April 2005.
>>
>> * Security added, you can now specify what the user of IPS is allowed to
> do
>> such as start different programs, hang-up calls etc.
>> * Many bug fixes
>>
>> Download: http://ipswitchboard.thorben.dk
>>
>>
>>
> ___
>> IPSwitchBoard is a FREE Windows.Net application that will allow you to:
>>
>> Unattended/attended transfers.
>> Park calls and retrieve/forward them again.
>> Organize all your Zap, SIP and IAX extensions (automatically retrieved
> from
>> Asterisk).
>> Hotel/Call Shop Billing module
>> Monitor all extensions.
>> Monitor all queues.
>> Monitor Agents.
>> Monitor Parked Calls.
>> Dynamically log extensions in and out of queues.
>> Integration with CRM software on the web.
>> Record conversations.
>> Browse Call Records
>> Drop any active call.
>> Set Do Not Disturb on Extensions and give a reason.
>> Speed Dialling.
>> User selectable ring tones for IPSwitchBoard.
>> User selectable button colors.
>>
>> ___
>> 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
>
> ___
> 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
> 



___
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

[Asterisk-Users] Re: chan_capi crashes asterisk

2005-04-30 Thread Stefan Tichy

On Fri, Apr 29, 2005 at 01:44:24AM +0200, Sebastian Voitzsch wrote:
> I can?t get chan_capi to work with any version of asterisk. I tried several 
> versions, all with the same effect: the phone rings, as soon as the call gets 
> answerd, asterisk crashes.

Certainly it is chan_capi 0.3.5, but which kernel and libcapi
(capi4linux) versions are used?


-- 
Stefan Tichy   <[EMAIL PROTECTED]>
___
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] IPSwitchBoard version 0.111 released

2005-04-30 Thread TC

what is the url for the version of the framework it wants now ?
the dot net auto installer is busted ?
- Original Message -
From: "Thorben Jensen" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"

Sent: Saturday, April 30, 2005 8:39 AM
Subject: [Asterisk-Users] IPSwitchBoard version 0.111 released


> Version 0.111 - 30. April 2005.
>
> * Security added, you can now specify what the user of IPS is allowed to
do
> such as start different programs, hang-up calls etc.
> * Many bug fixes
>
> Download: http://ipswitchboard.thorben.dk
>
>
>
___
> IPSwitchBoard is a FREE Windows.Net application that will allow you to:
>
> Unattended/attended transfers.
> Park calls and retrieve/forward them again.
> Organize all your Zap, SIP and IAX extensions (automatically retrieved
from
> Asterisk).
> Hotel/Call Shop Billing module
> Monitor all extensions.
> Monitor all queues.
> Monitor Agents.
> Monitor Parked Calls.
> Dynamically log extensions in and out of queues.
> Integration with CRM software on the web.
> Record conversations.
> Browse Call Records
> Drop any active call.
> Set Do Not Disturb on Extensions and give a reason.
> Speed Dialling.
> User selectable ring tones for IPSwitchBoard.
> User selectable button colors.
>
> ___
> 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

___
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] Bouncing DTMF?

2005-04-30 Thread Jan Johansson

>What type of phone SIP or analog? What is your DTMF type set for?

It's a system phone, via PBX to a PRI to Operator to my SIP Provider to my
Asterisk box.

Sip.conf is

[general]
context=default
port=5060
bindaddr=0.0.0.0
srvlookup=yes
disallow=all
allow=alaw
allow=gsm
;allow=ilbc
realm=asterisk
;
register=085000:[EMAIL PROTECTED]/1000

[rix]
type=peer
nat=yes
username=0850007696
fromuser=0850007696
secret=6ARiAnME
host=82.96.24.7
fromdomain=82.96.24.7
insecure=very



smime.p7s
Description: S/MIME cryptographic signature
___
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

[Asterisk-Users] 7910 and Skinny

2005-04-30 Thread Mark Johnson

I just had a very successful installation of Asterisk and have a 
question.  On my 7910's using the Skinny protocol, the user does not 
hear ringing when they make another call.  I found a patch that makes 
the ringing work, but something is still wrong with it.  If I use the 
7910 to make internal Skinny to other internal Skinny or SIP phones, the 
ringing works.  Once they make an outside call, they can not hear 
ringing again until I shutdown Asterisk and start it back up.  I'm using 
1.0.7.  Anyone have any ideas?  I also tried chan_sccp and that was a 
real disaster.  Asterisk kept crashing after a period of about 30 
minutes.  It was like when the phones reregistered so many times, it 
started claiming that some of the phones were dead and that others 
couldn't be registered because they already were, then it crashed.  
Anyone have any ideas?  Below is the patch code I found.

Mark
/@@ -1715,14 +1756,17 @@
   }
   switch(ind) {
   case AST_CONTROL_RINGING:
-   if (ast->_state == AST_STATE_RINGING) {
+   ast_verbose(VERBOSE_PREFIX_3 "State AST_CONTROL_RINGINGn");
+   // if (ast->_state == AST_STATE_RINGING) {
+   ast_verbose(VERBOSE_PREFIX_3 "State AST_STATE_RINGINGn");
   if (!sub->progress) {   
   transmit_tone(s, SKINNY_ALERT);
   transmit_callstate(s, l->instance, SKINNY_RINGOU
T, sub->callid);
   sub->ringing = 1;
+   ast_verbose(VERBOSE_PREFIX_3 "Started Ringingn"
);
   break;
   }
-   }
+   // }
/

___
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

[Asterisk-Users] IPSwitchBoard version 0.111 released

2005-04-30 Thread Thorben Jensen

Version 0.111 - 30. April 2005.

* Security added, you can now specify what the user of IPS is allowed to do
such as start different programs, hang-up calls etc. 
* Many bug fixes 

Download: http://ipswitchboard.thorben.dk


___
IPSwitchBoard is a FREE Windows.Net application that will allow you to: 

Unattended/attended transfers. 
Park calls and retrieve/forward them again. 
Organize all your Zap, SIP and IAX extensions (automatically retrieved from
Asterisk). 
Hotel/Call Shop Billing module 
Monitor all extensions. 
Monitor all queues. 
Monitor Agents. 
Monitor Parked Calls. 
Dynamically log extensions in and out of queues. 
Integration with CRM software on the web. 
Record conversations. 
Browse Call Records 
Drop any active call. 
Set Do Not Disturb on Extensions and give a reason. 
Speed Dialling. 
User selectable ring tones for IPSwitchBoard. 
User selectable button colors.

___
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: RE: [Asterisk-Users] Problems with TDM400P card

2005-04-30 Thread Rich Adamson

Look at the "ms =" statements in the code. I'm trying to rewrite
the code right now to provide something more useful.


> Im using RH9 and celerom 1.7 with 256 Mb RAM
> 
> Can you give me the detailed math on your calculations? 
> 
> |-Original Message-
> |From: [EMAIL PROTECTED] 
> |[mailto:[EMAIL PROTECTED] On Behalf Of 
> |Rich Adamson
> |Sent: Sábado, 30 de Abril de 2005 11:07 a.m.
> |To: Asterisk Users Mailing List - Non-Commercial Discussion
> |Subject: RE: RE: [Asterisk-Users] Problems with TDM400P card
> |
> |The way that zttest is written makes it a little difficult to 
> |interpret, but it essentially means that zttest tried to read
> |8192 bytes from the TDM card, and it took more then 1 second 
> |to do it (the objective is exactly 1.0 seconds, or 100%).
> |The 99.987 numbers says it took something like 1.02 
> |seconds to read the 8192 bytes instead. Because it took about 
> |21, microseconds too long, frame slips are going to be 
> |happening approximately every 10 seconds. (That's why spandsp 
> |doesn't work
> |right.)
> |I'm not sure (as yet) what the source of the delays are, but 
> |that's what some of us are trying to figure out.
> |
> |What OS distro are you using?
> |
> |
> |> Hows does this look?
> |> 
> |> Opened pseudo zap interface, measuring accuracy...
> |> 
> |> 8192 samples in 8192 sample intervals 100.00%
> |> 8192 samples in 8193 sample intervals 99.987793%
> |> 8192 samples in 8193 sample intervals 99.987793%
> |> 8192 samples in 8193 sample intervals 99.987793%
> |> 8192 samples in 8193 sample intervals 99.987793%
> |> 8192 samples in 8193 sample intervals 99.987793%
> |> 8192 samples in 8193 sample intervals 99.987793%
> |> 8192 samples in 8193 sample intervals 99.987793%
> |> 8192 samples in 8193 sample intervals 99.987793%
> |> 8192 samples in 8194 sample intervals 99.975586%
> |> 8192 samples in 8193 sample intervals 99.987793%
> |> 8192 samples in 8193 sample intervals 99.987793%
> |> 8192 samples in 8193 sample intervals 99.987793%
> |> --- Results after 13 passes ---
> |> Best: 100.00 -- Worst: 99.975586 -- Average: 99.987793
> |> 
> |> Good enough and what do I need to check in order to make 100%? What 
> |> does the test actually measure?
> |> 
> |>  
> |> 
> |> |-Original Message-
> |> |From: [EMAIL PROTECTED]
> |> |[mailto:[EMAIL PROTECTED] On Behalf Of Kim 
> |> |Culhan
> |> |Sent: Sábado, 30 de Abril de 2005 08:45 a.m.
> |> |To: asterisk-users@lists.digium.com
> |> |Subject: Re: RE: [Asterisk-Users] Problems with TDM400P card
> |> |
> |> |On Fri, April 29, 2005 4:58 pm, GEOFFREY SACHS said:
> |> |> I would also be interested in alternatives to the Tdm400p. I
> |> |have had
> |> |> endless problems with a tdm400p card not being able to get
> |> |the zttest
> |> |> numbers above
> |> |> 99.975 and as a result not being able eliminate an
> |> |intermitent but consistent echo.
> |> |> I have tried to date 4 different motherboard and hardware
> |> |combinations
> |> |> as well as different linux versions to no avial.I would
> |> |welcome some feedback on this.
> |> |
> |> |Since there appear to be several combinations of hardware and 
> |> |operating system which don't work well, here is a combination which 
> |> |appears to work fairly well:
> |> |
> |> |Intel 925XCV mb
> |> |
> |> |P-4 560 (3.6 gHz)
> |> |
> |> |wcfxs0: 
> |> |
> |> |FreeBSD 5.4-STABLE
> |> |
> |> |zttest -v
> |> |Opened pseudo zap interface, measuring accuracy...
> |> |
> |> |8192 samples in 8192 sample intervals 100.00%
> |> |8192 samples in 8192 sample intervals 100.00%
> |> |8192 samples in 8192 sample intervals 100.00%
> |> |8192 samples in 8192 sample intervals 100.00%
> |> |8192 samples in 8192 sample intervals 100.00%
> |> |8192 samples in 8192 sample intervals 100.00%
> |> |8192 samples in 8192 sample intervals 100.00%
> |> |8192 samples in 8192 sample intervals 100.00%
> |> |8192 samples in 8192 sample intervals 100.00%
> |> |8192 samples in 8192 sample intervals 100.00% ^C
> |> |--- Results after 10 passes ---
> |> |Best: 100.00 -- Worst: 100.00 -- Average: 100.00
> |> |
> |> |hope this helps
> |> |
> |> |-kim
> |> |
> |> |--
> |> |[EMAIL PROTECTED]
> |> |___
> |> |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
> |> |
> |> |
> |> 
> |> ___
> |> 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
> |> 
> |
> |---End of Original Message-
> |
> |
> |

RE: RE: [Asterisk-Users] Problems with TDM400P card

Im using RH9 and celerom 1.7 with 256 Mb RAM

Can you give me the detailed math on your calculations? 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Rich Adamson
|Sent: Sábado, 30 de Abril de 2005 11:07 a.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: RE: RE: [Asterisk-Users] Problems with TDM400P card
|
|The way that zttest is written makes it a little difficult to 
|interpret, but it essentially means that zttest tried to read
|8192 bytes from the TDM card, and it took more then 1 second 
|to do it (the objective is exactly 1.0 seconds, or 100%).
|The 99.987 numbers says it took something like 1.02 
|seconds to read the 8192 bytes instead. Because it took about 
|21, microseconds too long, frame slips are going to be 
|happening approximately every 10 seconds. (That's why spandsp 
|doesn't work
|right.)
|I'm not sure (as yet) what the source of the delays are, but 
|that's what some of us are trying to figure out.
|
|What OS distro are you using?
|
|
|> Hows does this look?
|> 
|> Opened pseudo zap interface, measuring accuracy...
|> 
|> 8192 samples in 8192 sample intervals 100.00%
|> 8192 samples in 8193 sample intervals 99.987793%
|> 8192 samples in 8193 sample intervals 99.987793%
|> 8192 samples in 8193 sample intervals 99.987793%
|> 8192 samples in 8193 sample intervals 99.987793%
|> 8192 samples in 8193 sample intervals 99.987793%
|> 8192 samples in 8193 sample intervals 99.987793%
|> 8192 samples in 8193 sample intervals 99.987793%
|> 8192 samples in 8193 sample intervals 99.987793%
|> 8192 samples in 8194 sample intervals 99.975586%
|> 8192 samples in 8193 sample intervals 99.987793%
|> 8192 samples in 8193 sample intervals 99.987793%
|> 8192 samples in 8193 sample intervals 99.987793%
|> --- Results after 13 passes ---
|> Best: 100.00 -- Worst: 99.975586 -- Average: 99.987793
|> 
|> Good enough and what do I need to check in order to make 100%? What 
|> does the test actually measure?
|> 
|>  
|> 
|> |-Original Message-
|> |From: [EMAIL PROTECTED]
|> |[mailto:[EMAIL PROTECTED] On Behalf Of Kim 
|> |Culhan
|> |Sent: Sábado, 30 de Abril de 2005 08:45 a.m.
|> |To: asterisk-users@lists.digium.com
|> |Subject: Re: RE: [Asterisk-Users] Problems with TDM400P card
|> |
|> |On Fri, April 29, 2005 4:58 pm, GEOFFREY SACHS said:
|> |> I would also be interested in alternatives to the Tdm400p. I
|> |have had
|> |> endless problems with a tdm400p card not being able to get
|> |the zttest
|> |> numbers above
|> |> 99.975 and as a result not being able eliminate an
|> |intermitent but consistent echo.
|> |> I have tried to date 4 different motherboard and hardware
|> |combinations
|> |> as well as different linux versions to no avial.I would
|> |welcome some feedback on this.
|> |
|> |Since there appear to be several combinations of hardware and 
|> |operating system which don't work well, here is a combination which 
|> |appears to work fairly well:
|> |
|> |Intel 925XCV mb
|> |
|> |P-4 560 (3.6 gHz)
|> |
|> |wcfxs0: 
|> |
|> |FreeBSD 5.4-STABLE
|> |
|> |zttest -v
|> |Opened pseudo zap interface, measuring accuracy...
|> |
|> |8192 samples in 8192 sample intervals 100.00%
|> |8192 samples in 8192 sample intervals 100.00%
|> |8192 samples in 8192 sample intervals 100.00%
|> |8192 samples in 8192 sample intervals 100.00%
|> |8192 samples in 8192 sample intervals 100.00%
|> |8192 samples in 8192 sample intervals 100.00%
|> |8192 samples in 8192 sample intervals 100.00%
|> |8192 samples in 8192 sample intervals 100.00%
|> |8192 samples in 8192 sample intervals 100.00%
|> |8192 samples in 8192 sample intervals 100.00% ^C
|> |--- Results after 10 passes ---
|> |Best: 100.00 -- Worst: 100.00 -- Average: 100.00
|> |
|> |hope this helps
|> |
|> |-kim
|> |
|> |--
|> |[EMAIL PROTECTED]
|> |___
|> |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
|> |
|> |
|> 
|> ___
|> 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
|> 
|
|---End of Original Message-
|
|
|___
|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
|
|

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/aste

Re: [Asterisk-Users] Dynamic phone groups.

2005-04-30 Thread Joris Vandalon

Citeren "Robert P. McKenzie" <[EMAIL PROTECTED]>:

> It may not be exactly what you are after but I do something like this:
>
> extensions.conf
>
>
HOUSEPHONES=SIP/somepc&SIP/anotherpc&IAX2/desktop&IAX2/someotherdesktop&SIP/sipuraline1&SIP/sipuraline2
>
> ; London Number - SIP Inbound provider
> exten => 1438645,1,Answer
> exten => 1438645,2,Dial(${HOUSEPHONES}|60|t)
> exten => 1438645,3,Voicemail(u50)
>
>
> Each phone listed above also has it's own extention, but the voicemail all
> goes to 50.  That way I can call any
> extension from anyother inside the house.  But calling 50 directly will make
> every phone ring.  Any phone not logged in
> will just be ignored and skipped.  The first phone to pick up gets it.  Call
> parking is on so if there is a need to
> transfer calls from one phone to another it can be done using parking.
The thing is that i want to add and remove phones dynamicly from the group with
astdb or so.

Cheers,
Joris
___
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] Queues configuration