Re: [asterisk-users] chanspy and DISA

[Alexander Lopez]

=> -Original Message-
=> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
=> boun...@lists.digium.com] On Behalf Of Steve Edwards
=> Sent: Tuesday, September 29, 2009 7:32 PM
=> To: Asterisk Users Mailing List - Non-Commercial Discussion
=> Subject: Re: [asterisk-users] chanspy and DISA
=> 
=> > Steve Edwards wrote:
=> >>
=> >> Is the manager or are the agents using disa()?
=> >>
=> >> How about:
=> >>
=> >>  exten = *,n,
set(SPYGROUP=ALLOW-SPYING)
=> >>
=> >> for the agents and:
=> >>
=> >>  exten = *,n,chanspy(,g(ALLOW-SPYING))
=> >>
=> >> the manager?
=> 
=> On Tue, 29 Sep 2009, John Millican wrote:
=> >
=> > The manager wants to be able to spy on agents who dial through the
=> PBX
=> > from their homes.  Currently the agents dial the main number, use
=> the
=> > "secret" code to get to authenticate and DISA, and then dial back
=> out
=> > for their sales calls. I have chanspy working great on all internal
=> > phones/extensions use group to limit who can spy and who can not.
It
=> not
=> > so much to allow spying it is finding the correct channel to spy on
=> for
=> > the remote users.
=> 
=> How about something like these snippets:
=> 
=> [i](!)
=>  exten = i,1,goto(${CONTEXT},s,1)
=> [s](!)
=>  exten = s,1,
=> verbose(1,[${CONTEXT}:${EXTEN}])
=> 
=> [home-agent-login](i,s)
=>  exten = s,n,read(AGENT-ID,enter-agent-
=> number)
=>  exten = s,n,set(SPYGROUP=${AGENT-ID})
=>  .
=>  .
=>  .
=> 
=> [supervisor-login](i,s)
=>  exten = s,n,read(AGENT-ID,enter-agent-
=> number)
=>  exten = s,n,chanspy(,g(${AGENT-ID}))
=>  exten = s,n,goto(s,1)
=>  .
=>  .
=>  .
=> 
=> --
=> Thanks in advance,
=>
--
=> ---
=> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867
=> PST
=> Newline  Fax: +1-760-731-
=> 3000

[Alexander Lopez] 
It adds another layer but have the sales agents login to DISA via a
local Channel. You can call these channels whatever you like. Once you
have a one to one relationship with the sales agent and their respective
Local Channel you can Chan spy on the Local Channels..

Alex



___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] digium fax: can't indicate condition 19?

[Trevor Peirce]

Kevin P. Fleming wrote:
> Where would you suggest this note be placed? We've tried to make our
> documentation as clear as possible that the download selector is the
> canonical place to get the proper FFA modules for any given version of
> Asterisk, and the fact that the newer versions of Asterisk are not
> listed there should have raised a red flag in people's minds that they
> might not be supported.
>
> I'm open to suggestions on how to make this better, but it should
> (hopefully) be a one time occurrence caused by the major changes we had
> to make in T.38 support to fix a number of issues.
>
>   
Yes, I was concerned when I noticed my version of asterisk was not 
listed. However, I took a leap of faith that if it supports some 1.6.1.x 
releases, it would support them all. My assumption was that they were 
simply not released when the most recent Fax app was released. Usually 
minor version releases don't break APIs :)

Adding an option to the selector for the more recent asterisk versions 
and then displaying a note indicating that the G.711 functionality is 
working but the T.38 support is temporarily broken would have been an 
excellent place to clear up the issue.

Thanks for caring :)

-- 
Trevor Peirce
Digital Conceptions Canada

http://www.digitalcon.ca
1-888-606-3030 / 250-391-7822




___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] dialing 0 in directory()

[Paul Dugas]

On Tue, Sep 29, 2009 at 3:11 PM, Doug Lytle  wrote:
> [directory]
>
> exten => s,1,Wait(1)
> exten => s,n,Directory(sip|sip|eb)
> exten => s,n,Playback(goodbye)
> exten => s,n,Hangup()
> exten => o,1,Goto(incoming,s,1)

I thought the second arg to Directory() was where it would look for
the "o" extension.  Briefly looking through app_meetme.c frim 1.6.1.6,
it seems to fit.  But you have "sip" for both the vm-context and the
dial-context.  Weird.

P
-- 
Paul Dugas -- Computer Engineer -- Dugas Enterprises, LLC
522 Black Canyon Park, Canton GA 30114 USA
p...@dugasenterprises.com -- +1.404.932.1355

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] digium fax: failed to queue document

[sean darcy]

On Tue, Sep 29, 2009 at 2:48 PM, David Backeberg  wrote:
> On Mon, Sep 28, 2009 at 10:08 PM, sean darcy  wrote:
>> On Mon, Sep 28, 2009 at 1:09 PM, David Backeberg  
>> wrote:
>>> Have you tried using ps2tiff?
>> I looked up ps2tiff. That seems to be a windows program. There is a
>> pstotiff linux program, but it seems to be unmaintained, and isn't
>> available on Fedora.
>
> My bad. I was thinking that because libtiff provides a tiff2ps they
> would also provide a ps2tiff. I was mistaken. I use tiff2pdf as part
> of my inbound fax receipt setup to make things play nice with the
> Windows desktops.
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2009 - October 13 - 15 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>

Actually tiff2ps could work. Use convert from imagemagick to get to
tiff, tiff2ps, and then gs to g3 fax. . test2ps is also old and
unmaintained. :-(

sean

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] chanspy and DISA

[Steve Edwards]

> Steve Edwards wrote:
>>
>> Is the manager or are the agents using disa()?
>>
>> How about:
>>
>>  exten = *,n,set(SPYGROUP=ALLOW-SPYING)
>>
>> for the agents and:
>>
>>  exten = *,n,chanspy(,g(ALLOW-SPYING))
>>
>> the manager?

On Tue, 29 Sep 2009, John Millican wrote:
>
> The manager wants to be able to spy on agents who dial through the PBX 
> from their homes.  Currently the agents dial the main number, use the 
> "secret" code to get to authenticate and DISA, and then dial back out 
> for their sales calls. I have chanspy working great on all internal 
> phones/extensions use group to limit who can spy and who can not. It not 
> so much to allow spying it is finding the correct channel to spy on for 
> the remote users.

How about something like these snippets:

[i](!)
 exten = i,1,goto(${CONTEXT},s,1)
[s](!)
 exten = s,1,verbose(1,[${CONTEXT}:${EXTEN}])

[home-agent-login](i,s)
 exten = s,n,read(AGENT-ID,enter-agent-number)
 exten = s,n,set(SPYGROUP=${AGENT-ID})
 .
 .
 .

[supervisor-login](i,s)
 exten = s,n,read(AGENT-ID,enter-agent-number)
 exten = s,n,chanspy(,g(${AGENT-ID}))
 exten = s,n,goto(s,1)
 .
 .
 .

-- 
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] kill sip user

[Paul Hales]

Death to all sip users!

Paulh


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] UpdateConfig

[Jared Smith]

- "Danny Nicholas"  wrote:
> Two questions: 1. do you need an ActionID line?

Danny,

It's *always* considered best practice to have an ActionID line in AMI 
commands, so that you can easily differentiate the responses, especially to 
asynchronous commands.

--
Jared Smith
Training Manager
Digium, Inc.

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] UpdateConfig

[Tzafrir Cohen]

On Tue, Sep 29, 2009 at 08:07:40PM +, Anahi Ludueña wrote:
> 
> Hi people, I need to update the voicemail.conf from the UpdateConfig Action 
> (AMI).
> The problem is that I executed:
> 
> Action: UpdateConfig
> srcFileName: voicemail.conf
> dstFileName: voicemail.conf
> Action-00:append
> Cat-00:test
> Var-00:exten
> Value-00:>999,test

Let me get this straight.

You want to add to section [test] the line

  exten = 999,test

That looks more like a line from extensions.conf than a line from
voicemail.conf (though 'test' is not really a valid dialplan app).

> 
> But I don't see the changes in the file. 

Is it writable to the user running Asterisk?

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Asterisk on DD-WRT : modules.conf not found

[Tzafrir Cohen]

On Tue, Sep 29, 2009 at 08:19:16PM +0200, jonas kellens wrote:
> Through the optware-package I have installed Asterisk on an external
> USB. Further I have a Linksys WRT610N with DD-WRT v24 mega.
> 
> I start asterisk with the following command : /opt/sbin/asterisk -c
> I get the following WARNING :
> 
> r...@dd-wrt:/opt/etc/asterisk# /opt/sbin/asterisk -c
> Asterisk 1.4.22.1, Copyright (C) 1999 - 2008 Digium, Inc. and others.
> Created by Mark Spencer 
> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty'
> for details.
> This is free software, with components licensed under the GNU General
> Public
> License version 2 and other licenses; you are welcome to redistribute it
> under
> certain conditions. Type 'core show license' for details.
> =
> [ Booting...
> [ Reading Master Configuration ]
> Errors detected in logger.conf: see above; default settings will be
> used.
> [Jan  1 00:56:22] Asterisk Event Logger
> Started /var/log/asterisk/event_log
> [Jan  1 00:56:22] Asterisk Dynamic Loader Starting:
> [Jan  1 00:56:22] WARNING[4942]: loader.c:786 load_modules: No
> 'modules.conf' found, no modules will be loaded.
> ...
> 
> Although I have a modules.conf file in /opt/etc/asterisk/-directory :
> 
> r...@dd-wrt:/opt/etc/asterisk# ls -l /opt/etc/asterisk/
> -rw-r--r--1 root root 2845 Jan 13  2009 agents.conf
> -rw-r--r--1 root root 2675 Jan 13  2009 alsa.conf
> -rw-r--r--1 root root  767 Jan 13  2009 amd.conf
> -rw-r--r--1 root root 1981 Jan  1 00:39 asterisk.conf
> -rw-r--r--1 root root 7324 Jan 13  2009 cdr.conf
> -rw-r--r--1 root root  190 Jan 13  2009 dnsmgr.conf
> -rw-r--r--1 root root 1506 Jan 13  2009 extconfig.conf
> -rw-r--r--1 root root 1976 Jan  1 00:42 extensions.conf
> -rw-r--r--1 root root 5301 Jan 13  2009 features.conf
> -rw-r--r--1 root root 3755 Jan 13  2009 followme.conf
> -rw-r--r--1 root root  997 Jan 13  2009 http.conf
> -rw-r--r--1 root root 1548 Jan  1 00:39 iax.conf
> -rw-r--r--1 root root25247 Jan  1 00:39 indications.conf
> -rw-r--r--1 root root 2174 Jan  1 00:49 logger.conf
> -rw-r--r--1 root root  926 Jan 13  2009 meetme.conf
> -rw-r--r--1 root root 1393 Jan  1 00:54 modules.conf
> -rw-r--r--1 root root 1943 Jan 13  2009 musiconhold.conf
> -rw-r--r--1 root root 3594 Jan 13  2009 oss.conf
> -rw-r--r--1 root root11732 Jan 13  2009 queues.conf
> -rw-r--r--1 root root  496 Jan 13  2009 rtp.conf
> -rw-r--r--1 root root38583 Jan 13  2009 sip.conf
> -rw-r--r--1 root root  317 Jan 13  2009 sip_notify.conf
> -rw-r--r--1 root root 6695 Jan 13  2009 sla.conf

Is /opt/etc/asterisk the compiled config directory (astetcdir)?

What is the contents of asterisk.conf?

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

[asterisk-users] Dumb Question - Dialing internal and external

[Danny Nicholas]

Hello listers,

  I'm running Asterisk 1.4.26.1 and 1.4-R201993 (SVN) using
Polycom 501's and POTS.  The problem I'm experiencing is that when I dial a
call, it takes 1-4 seconds before I hear a ring.  I understand that there is
a delay on POTS connectivity, but what's the deal on an internal call?  I
dial from 104 to 105, press send, 105 rings and about a second later I get
the ring indication on 104.  Everything is in-house and behind a firewall.
TIA.

 

Danny Nicholas

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] LDAP integration

[magicrhesus]

Hi,

I never try it on 1.6 but any information on further compability with 
1.6 could be interesting.

This version was developped for the 1.4 version.
If you need informations about installing or configuring this module, 
don't hesitate to contact me.


On 09/29/2009 09:21 PM, Rafael Seste wrote:

tks for all answers!!!

Antoine,
I will try to do it tomorrow.
just one question. Do you know if it works with asterisk1.6? I'm using
this version and looks like that your friend is using 1.4


On Tue, Sep 29, 2009 at 12:28 PM, Antoine Patte  wrote:
   

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Hi,

A realtime ldap driver exist.

He can put the user/peer sip/iax in a ldap directory and configuration
files.

A friend has updated as part of his final study of.

You can find-it there : http://wiki.ouranos.be/doku.php/stage:ldap

Or contact us at : magicrhe...@ouranos.be

For other questions, I'm a here ...

- --
Antoine Patte


Rafael Seste wrote:
 

Hi all,

I looked on the Internet but I didn't find any good how-to.
I would like to integrate a ldap server ( with all users data) with
asterisk to authenticate SIP users. With this solution I will only
need to add a user on ldap, it will not be necessary to add any
special configuration on sip.conf

Is that possible???If so, How can I configure this setup???

Thanks in advance

   

-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.10 (GNU/Linux)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iEYEARECAAYFAkrCJ6EACgkQBnIOcv+j7+wXtgCcDaoIAGZJfva39XFUtqRoMkih
XqYAoOnaMYa//DwR9F0doxtd3otPTeeF
=3D09
-END PGP SIGNATURE-

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

 



   
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] chanspy and DISA

[John Millican]

Steve Edwards wrote:
>>> On Tue, 29 Sep 2009, John Millican wrote:
>>>
 I have a request for remote users to be able to dial through the system
 so that the sales managers can barge/chanspy on the sales force. I have
 the DISA part working with authentication(rather straight forward) but
 what I can not figure out is how to enable the supervisors to be able to
 barge on these calls.  Is there a way to find the channel to barge on
 that would be usable by NON tech people?
> 
>> Steve Edwards wrote:
> 
>>> How do you see this working? I'm guessing the manager would like to either
>>> key in an agent ID number or be able to step through agents?
>>>
>>> The chanspy() "g" option may be part of your solution.
>>>
>>>  g(grp) - Match only channels where their ${SPYGROUP} variable is set
>>> to 'grp'.
>>>
> 
> On Tue, 29 Sep 2009, John Millican wrote:
>> Exactly, the problem is I can not determine the channel that DISA 
>> receives or places the call on.  Is there a way to set this in the dial 
>> plan? Or am I just missing something simple?  It was suggested to use 
>> the AMI and present the info as a web page but this will require 
>> retraining the manager, as we all know this is a notoriously difficult 
>> process.
> 
> Is the manager or are the agents using disa()?
> 
> How about:
> 
>  exten = *,n,set(SPYGROUP=ALLOW-SPYING)
> 
> for the agents and:
> 
>  exten = *,n,chanspy(,g(ALLOW-SPYING))
> 
> the manager?
> 

The manager wants to be able to spy on agents who dial through the PBX
from their homes.  Currently the agents dial the main number, use the
"secret" code to get to authenticate and DISA, and then dial back out
for their sales calls.
I have chanspy working great on all internal phones/extensions use group
to limit who can spy and who can not. It not so much to allow spying it
is finding the correct channel to spy on for the remote users.

JohnM


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] UpdateConfig

[Anahi Ludueña]

Thanks, the result was:


Response: Success







Anahi Ludueña
 



From: da...@debsinc.com
To: asterisk-users@lists.digium.com
Date: Tue, 29 Sep 2009 15:16:52 -0500
Subject: Re: [asterisk-users] UpdateConfig



















Two questions: 1. do you need an ActionID
line? 2. did you try this in a telnet session so you could see the feedback?

 









From:
asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña

Sent: Tuesday, September 29, 2009
3:08 PM

To: asterisk-users@lists.digium.com

Subject: [asterisk-users]
UpdateConfig



 

Hi people, I need to update the
voicemail.conf from the UpdateConfig Action (AMI).

The problem is that I executed:



Action: UpdateConfig

srcFileName: voicemail.conf

dstFileName: voicemail.conf

Action-00:append

Cat-00:test

Var-00:exten

Value-00:>999,test



But I don't see the changes in the file. 

Can anybody tell me if there is something wrong in that code?



Thanks,

















Anahi
Ludueña

 















Diferentes formas de estar en contacto con amigos y
familiares. Descúbrelas. Descúbrelas.

  
_
Descubre todas las formas en que puedes estar en contacto con amigos y 
familiares.
http://www.microsoft.com/windows/windowslive/default.aspx___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] UpdateConfig

[Danny Nicholas]

Two questions: 1. do you need an ActionID line? 2. did you try this in a
telnet session so you could see the feedback?

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña
Sent: Tuesday, September 29, 2009 3:08 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] UpdateConfig

 

Hi people, I need to update the voicemail.conf from the UpdateConfig Action
(AMI).
The problem is that I executed:

Action: UpdateConfig
srcFileName: voicemail.conf
dstFileName: voicemail.conf
Action-00:append
Cat-00:test
Var-00:exten
Value-00:>999,test

But I don't see the changes in the file. 
Can anybody tell me if there is something wrong in that code?

Thanks,





  _  

Anahi Ludueña

 





  _  

Diferentes formas de estar en contacto con amigos y familiares. Descúbrelas.
Descúbrelas.  

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] LDAP integration

[Gavin Henry]

Which version of the LDAP schema? I look after the one in 1.6.

Thanks.

On 29/09/2009, John A. Sullivan III  wrote:
> On Tue, 2009-09-29 at 11:01 -0300, Rafael Seste wrote:
>> Hi all,
>>
>> I looked on the Internet but I didn't find any good how-to.
>> I would like to integrate a ldap server ( with all users data) with
>> asterisk to authenticate SIP users. With this solution I will only
>> need to add a user on ldap, it will not be necessary to add any
>> special configuration on sip.conf
>>
>> Is that possible???If so, How can I configure this setup???
>>
>> Thanks in advance
>>
> I considered doing this using LDAP as a real-time database.  I decided
> not to for two reasons which I'll share below. However, I am very new to
> Asterisk so I would be very curious to know from more experienced folks
> if my assumptions were false.
>
> First, there were some good how-tos about using LDAP as a real-time
> database but, if I recall, the schema is extended in such a way that the
> regular user password is not the password used by Asterisk.
>
> Second, I believe we saw a way we could map the Asterisk password to the
> regular user password (it's been a while so I'm not sure about that) but
> were concerned about the problems of entering secure passwords from a
> phone keypad.  We enforce fairly secure passwords - at least nine
> characters with some variety of characters and encourage much longer
> passwords.  Having to enter lots of characters in both cases as well as
> symbols seemed difficult from a phone keypad.  Thus, we decided
> (reluctantly) to use separate simple passwords for phone access instead
> of the very secure passwords we use to data access.
>
> Hope this helps and looking forward to more informed comments than mine!
> - John
> --
> John A. Sullivan III
> Open Source Development Corporation
> +1 207-985-7880
> jsulli...@opensourcedevel.com
>
> http://www.spiritualoutreach.com
> Making Christianity intelligible to secular society
>
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2009 - October 13 - 15 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>

-- 
Sent from my mobile device

http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

[asterisk-users] UpdateConfig

[Anahi Ludueña]

Hi people, I need to update the voicemail.conf from the UpdateConfig Action 
(AMI).
The problem is that I executed:

Action: UpdateConfig
srcFileName: voicemail.conf
dstFileName: voicemail.conf
Action-00:append
Cat-00:test
Var-00:exten
Value-00:>999,test

But I don't see the changes in the file. 
Can anybody tell me if there is something wrong in that code?

Thanks,







Anahi Ludueña
 

  
_
Descubre todas las formas en que puedes estar en contacto con amigos y 
familiares.
http://www.microsoft.com/windows/windowslive/default.aspx___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

[asterisk-users] kill sip user

[Bayardo Sanchez]

I have a user but I need to give that user only kill and disable all
connection cut calls what is the command in the CLIC

-- 
Bayardo Sánchez García
Web Developer - Internet Portals - Asterisk Support - Windows Server Support
- Proxy Support - Linux Server
E-mail: bayardo.sanc...@gmail.com
Linux User: #418392
America Central - Managua, NI (505) 2249-2853 -  84886876
IM msn messenger: bjsanch...@hotmail.com
Skype: bayardo.sanchez
This email is intended solely for the person or organization to which it is
addressed. It may contain privileged and confidential information. If you
are not the intended recipient, you are prohibited from copying, disclosing
or distributing this email or its contents (as it may be unlawful for you to
do so) or taking any action in reliance on it. If you have received this
email by mistake, please delete it. All e-mail sent to this address will be
received by B.S. Solution e-mail system and is subject to archiving and
review by someone other than the recipient.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] chanspy and DISA

[Steve Edwards]

>> On Tue, 29 Sep 2009, John Millican wrote:
>>
>>> I have a request for remote users to be able to dial through the system
>>> so that the sales managers can barge/chanspy on the sales force. I have
>>> the DISA part working with authentication(rather straight forward) but
>>> what I can not figure out is how to enable the supervisors to be able to
>>> barge on these calls.  Is there a way to find the channel to barge on
>>> that would be usable by NON tech people?

> Steve Edwards wrote:

>> How do you see this working? I'm guessing the manager would like to either
>> key in an agent ID number or be able to step through agents?
>>
>> The chanspy() "g" option may be part of your solution.
>>
>>  g(grp) - Match only channels where their ${SPYGROUP} variable is set
>> to 'grp'.
>>

On Tue, 29 Sep 2009, John Millican wrote:
>
> Exactly, the problem is I can not determine the channel that DISA 
> receives or places the call on.  Is there a way to set this in the dial 
> plan? Or am I just missing something simple?  It was suggested to use 
> the AMI and present the info as a web page but this will require 
> retraining the manager, as we all know this is a notoriously difficult 
> process.

Is the manager or are the agents using disa()?

How about:

 exten = *,n,set(SPYGROUP=ALLOW-SPYING)

for the agents and:

 exten = *,n,chanspy(,g(ALLOW-SPYING))

the manager?

-- 
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] chanspy and DISA

[John Millican]

Steve Edwards wrote:
> On Tue, 29 Sep 2009, John Millican wrote:
> 
>> I have a request for remote users to be able to dial through the system 
>> so that the sales managers can barge/chanspy on the sales force. I have 
>> the DISA part working with authentication(rather straight forward) but 
>> what I can not figure out is how to enable the supervisors to be able to 
>> barge on these calls.  Is there a way to find the channel to barge on 
>> that would be usable by NON tech people?
> 
> How do you see this working? I'm guessing the manager would like to either 
> key in an agent ID number or be able to step through agents?
> 
> The chanspy() "g" option may be part of your solution.
> 
>  g(grp) - Match only channels where their ${SPYGROUP} variable is set 
> to 'grp'.
> 

Exactly, the problem is I can not determine the channel that DISA
receives or places the call on.  Is there a way to set this in the dial
plan? Or am I just missing something simple?  It was suggested to use
the AMI and present the info as a web page but this will require
retraining the manager, as we all know this is a notoriously difficult
process.
JohnM


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] LDAP integration

[Rafael Seste]

tks for all answers!!!

Antoine,
I will try to do it tomorrow.
just one question. Do you know if it works with asterisk1.6? I'm using
this version and looks like that your friend is using 1.4


On Tue, Sep 29, 2009 at 12:28 PM, Antoine Patte  wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Hi,
>
> A realtime ldap driver exist.
>
> He can put the user/peer sip/iax in a ldap directory and configuration
> files.
>
> A friend has updated as part of his final study of.
>
> You can find-it there : http://wiki.ouranos.be/doku.php/stage:ldap
>
> Or contact us at : magicrhe...@ouranos.be
>
> For other questions, I'm a here ...
>
> - --
> Antoine Patte
>
>
> Rafael Seste wrote:
>> Hi all,
>>
>> I looked on the Internet but I didn't find any good how-to.
>> I would like to integrate a ldap server ( with all users data) with
>> asterisk to authenticate SIP users. With this solution I will only
>> need to add a user on ldap, it will not be necessary to add any
>> special configuration on sip.conf
>>
>> Is that possible???If so, How can I configure this setup???
>>
>> Thanks in advance
>>
>
> -BEGIN PGP SIGNATURE-
> Version: GnuPG v1.4.10 (GNU/Linux)
> Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org
>
> iEYEARECAAYFAkrCJ6EACgkQBnIOcv+j7+wXtgCcDaoIAGZJfva39XFUtqRoMkih
> XqYAoOnaMYa//DwR9F0doxtd3otPTeeF
> =3D09
> -END PGP SIGNATURE-
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2009 - October 13 - 15 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
Rafael S. Seste

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] dialing 0 in directory()

[Doug Lytle]

Paul Dugas wrote:
> [attendant]
> ; 
> exten => *,1,NoOp(Attendant: Directory)
> exten => *,n,Directory(default,attendant,eb)
> exten => *,n,Goto(s,1)
>
> exten => o,1,NoOp(Zero)
> exten => o,n,Goto(0,1)
>
> exten => a,1,NoOp(Star)
> exten => a,n,Goto(0,1)
>   

Works fine for me, I've got:

exten => 31*,1,Set(CONNECTEDLINE(all)="Directory" <31*>)
exten => 31*,n,Goto(directory,s,1)


[directory]

exten => s,1,Wait(1)
exten => s,n,Directory(sip|sip|eb)
exten => s,n,Playback(goodbye)
exten => s,n,Hangup()
exten => o,1,Goto(incoming,s,1)

Doug


-- 
 
Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety."


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] digium fax: failed to queue document

[David Backeberg]

On Mon, Sep 28, 2009 at 10:08 PM, sean darcy  wrote:
> On Mon, Sep 28, 2009 at 1:09 PM, David Backeberg  wrote:
>> Have you tried using ps2tiff?
> I looked up ps2tiff. That seems to be a windows program. There is a
> pstotiff linux program, but it seems to be unmaintained, and isn't
> available on Fedora.

My bad. I was thinking that because libtiff provides a tiff2ps they
would also provide a ps2tiff. I was mistaken. I use tiff2pdf as part
of my inbound fax receipt setup to make things play nice with the
Windows desktops.

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] digium fax: can't indicate condition 19?

[Kevin P. Fleming]

Trevor Peirce wrote:

> A note somewhere would have been nice explaining this. I recently tried 
> the Digium Fax to determine if a client should buy some licenses, but 
> after seeing it reject invites for T.38 on incoming calls and never try 
> to switch for outgoing, I figured it was just broken in that regard.

Where would you suggest this note be placed? We've tried to make our
documentation as clear as possible that the download selector is the
canonical place to get the proper FFA modules for any given version of
Asterisk, and the fact that the newer versions of Asterisk are not
listed there should have raised a red flag in people's minds that they
might not be supported.

I'm open to suggestions on how to make this better, but it should
(hopefully) be a one time occurrence caused by the major changes we had
to make in T.38 support to fix a number of issues.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check us out at www.digium.com & www.asterisk.org

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

[asterisk-users] Asterisk on DD-WRT : modules.conf not found

[jonas kellens]

Through the optware-package I have installed Asterisk on an external
USB. Further I have a Linksys WRT610N with DD-WRT v24 mega.

I start asterisk with the following command : /opt/sbin/asterisk -c
I get the following WARNING :

r...@dd-wrt:/opt/etc/asterisk# /opt/sbin/asterisk -c
Asterisk 1.4.22.1, Copyright (C) 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer 
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty'
for details.
This is free software, with components licensed under the GNU General
Public
License version 2 and other licenses; you are welcome to redistribute it
under
certain conditions. Type 'core show license' for details.
=
[ Booting...
[ Reading Master Configuration ]
Errors detected in logger.conf: see above; default settings will be
used.
[Jan  1 00:56:22] Asterisk Event Logger
Started /var/log/asterisk/event_log
[Jan  1 00:56:22] Asterisk Dynamic Loader Starting:
[Jan  1 00:56:22] WARNING[4942]: loader.c:786 load_modules: No
'modules.conf' found, no modules will be loaded.
...

Although I have a modules.conf file in /opt/etc/asterisk/-directory :

r...@dd-wrt:/opt/etc/asterisk# ls -l /opt/etc/asterisk/
-rw-r--r--1 root root 2845 Jan 13  2009 agents.conf
-rw-r--r--1 root root 2675 Jan 13  2009 alsa.conf
-rw-r--r--1 root root  767 Jan 13  2009 amd.conf
-rw-r--r--1 root root 1981 Jan  1 00:39 asterisk.conf
-rw-r--r--1 root root 7324 Jan 13  2009 cdr.conf
-rw-r--r--1 root root  190 Jan 13  2009 dnsmgr.conf
-rw-r--r--1 root root 1506 Jan 13  2009 extconfig.conf
-rw-r--r--1 root root 1976 Jan  1 00:42 extensions.conf
-rw-r--r--1 root root 5301 Jan 13  2009 features.conf
-rw-r--r--1 root root 3755 Jan 13  2009 followme.conf
-rw-r--r--1 root root  997 Jan 13  2009 http.conf
-rw-r--r--1 root root 1548 Jan  1 00:39 iax.conf
-rw-r--r--1 root root25247 Jan  1 00:39 indications.conf
-rw-r--r--1 root root 2174 Jan  1 00:49 logger.conf
-rw-r--r--1 root root  926 Jan 13  2009 meetme.conf
-rw-r--r--1 root root 1393 Jan  1 00:54 modules.conf
-rw-r--r--1 root root 1943 Jan 13  2009 musiconhold.conf
-rw-r--r--1 root root 3594 Jan 13  2009 oss.conf
-rw-r--r--1 root root11732 Jan 13  2009 queues.conf
-rw-r--r--1 root root  496 Jan 13  2009 rtp.conf
-rw-r--r--1 root root38583 Jan 13  2009 sip.conf
-rw-r--r--1 root root  317 Jan 13  2009 sip_notify.conf
-rw-r--r--1 root root 6695 Jan 13  2009 sla.conf

Could I change something to point Asterisk to the correct
modules.conf-file ??

Also there are no sip and/or iax-users :

*CLI> sip show users
No such command 'sip show users' (type 'help sip show' for other
possible commands)
*CLI> iax2 show users
No such command 'iax2 show users' (type 'help iax2 show' for other
possible commands)

The command sip show and iax2 show do not seem to be present...

Can someone advise ??

Greetingz,
Jonas.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

[asterisk-users] dialing 0 in directory()

[Paul Dugas]

I've got a context in my dialplan like so but pressing 0 doesn't seem
to be working.  Instead of dropping out to the "o" extension, it's
just returning to the start of the direcotry app.   Same with star.
Anyone see where I've gone awry?

[attendant]
; 
exten => *,1,NoOp(Attendant: Directory)
exten => *,n,Directory(default,attendant,eb)
exten => *,n,Goto(s,1)

exten => o,1,NoOp(Zero)
exten => o,n,Goto(0,1)

exten => a,1,NoOp(Star)
exten => a,n,Goto(0,1)

Paul
--
Paul Dugas -- Computer Engineer -- Dugas Enterprises, LLC
522 Black Canyon Park, Canton GA 30114 USA
p...@dugasenterprises.com -- +1.404.932.1355

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] digium fax: can't indicate condition 19?

[Trevor Peirce]

Kevin P. Fleming wrote:
> I am working on getting this situation resolved and should have new
> releases of FFA out at the end of this week, but in the meantime if you
> want to use FFA with T.38 support you'll have to use one of the versions
> of Asterisk listed on the download selector page.
>   

A note somewhere would have been nice explaining this. I recently tried 
the Digium Fax to determine if a client should buy some licenses, but 
after seeing it reject invites for T.38 on incoming calls and never try 
to switch for outgoing, I figured it was just broken in that regard.

Regards,

-- 
Trevor Peirce
Digital Conceptions Canada

http://www.digitalcon.ca
1-888-606-3030 / 250-391-7822




___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] chanspy and DISA

[Steve Edwards]

On Tue, 29 Sep 2009, John Millican wrote:

> I have a request for remote users to be able to dial through the system 
> so that the sales managers can barge/chanspy on the sales force. I have 
> the DISA part working with authentication(rather straight forward) but 
> what I can not figure out is how to enable the supervisors to be able to 
> barge on these calls.  Is there a way to find the channel to barge on 
> that would be usable by NON tech people?

How do you see this working? I'm guessing the manager would like to either 
key in an agent ID number or be able to step through agents?

The chanspy() "g" option may be part of your solution.

 g(grp) - Match only channels where their ${SPYGROUP} variable is set 
to 'grp'.

-- 
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Secure passwords, was LDAP integration

[John A. Sullivan III]

On Tue, 2009-09-29 at 11:23 -0500, Tilghman Lesher wrote:
> On Tuesday 29 September 2009 10:30:37 John A. Sullivan III wrote:
> > Second, I believe we saw a way we could map the Asterisk password to the
> > regular user password (it's been a while so I'm not sure about that) but
> > were concerned about the problems of entering secure passwords from a
> > phone keypad.  We enforce fairly secure passwords - at least nine
> > characters with some variety of characters and encourage much longer
> > passwords.  Having to enter lots of characters in both cases as well as
> > symbols seemed difficult from a phone keypad.  Thus, we decided
> > (reluctantly) to use separate simple passwords for phone access instead
> > of the very secure passwords we use to data access.
> 
> I would hope that you're at least restricting your peers to be limited to a
> set of IPs distinctive to your phones.  Otherwise, this is a recipe for
> disaster, especially if a) your registration server is accessible externally,
> and b) your phones are permitted to make toll calls, especially international
> numbers.
> 
> Most good IP phones permit a method of configuration which does not require
> typing a password into a keypad.  You should probably learn to use that method
> or switch to a phone with that ability, then use secure passwords.  Phones are
> just as important as data and should be supplied with complex passwords.
> 
Thanks for the feedback.  Indeed, we do restrict the SIP domains and do
not allow registration from outside the internal network and we do use
passwords - just not as sophisticated.

Perhaps I am being overly conscious of client simplicity.  I was
thinking of the case where internal users might temporarily move to
another phone.  Rather than pulling up the web interface to the phone,
we wanted them to be able to register through the phone keypad.  I
suppose they would need to enter their IDs anyway and those are
alpha-numeric.  Thus, the entering passwords would be similar to
entering the IDs.  On the other hand, we do tend to use the same
registration password for voicemail and meetme and those are regularly
entered from the key pad.  Thanks - John
-- 
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com

http://www.spiritualoutreach.com
Making Christianity intelligible to secular society


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] digium fax: can't indicate condition 19?

[Kevin P. Fleming]

Scott L. Lykens wrote:

> As we are eliminating our PRI soon I am trying to get faxing via T.38 working 
> properly. I'm not interested in running Callweaver next to Asterisk just to 
> support fax. :/ 
> 
> Any insight into this error would be greatly appreciated.

This is occurring because the current versions of Asterisk have an
updated T.38 API that the current res_fax/res_fax_digium modules do not
yet support (this is why the current versions of Asterisk are not listed
on the Fax For Asterisk download selector page yet).

I am working on getting this situation resolved and should have new
releases of FFA out at the end of this week, but in the meantime if you
want to use FFA with T.38 support you'll have to use one of the versions
of Asterisk listed on the download selector page.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check us out at www.digium.com & www.asterisk.org

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Secure passwords, was LDAP integration

[Tilghman Lesher]

On Tuesday 29 September 2009 10:30:37 John A. Sullivan III wrote:
> Second, I believe we saw a way we could map the Asterisk password to the
> regular user password (it's been a while so I'm not sure about that) but
> were concerned about the problems of entering secure passwords from a
> phone keypad.  We enforce fairly secure passwords - at least nine
> characters with some variety of characters and encourage much longer
> passwords.  Having to enter lots of characters in both cases as well as
> symbols seemed difficult from a phone keypad.  Thus, we decided
> (reluctantly) to use separate simple passwords for phone access instead
> of the very secure passwords we use to data access.

I would hope that you're at least restricting your peers to be limited to a
set of IPs distinctive to your phones.  Otherwise, this is a recipe for
disaster, especially if a) your registration server is accessible externally,
and b) your phones are permitted to make toll calls, especially international
numbers.

Most good IP phones permit a method of configuration which does not require
typing a password into a keypad.  You should probably learn to use that method
or switch to a phone with that ability, then use secure passwords.  Phones are
just as important as data and should be supplied with complex passwords.

-- 
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com & www.asterisk.org

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Retrieve Call setup - QoS

[Danny Nicholas]

I believe that this information is at least indirectly in the CDR.


 

104

106

DLPN_DialPlan1

"Danny Nicholas" <104>

SIP/104-b790d5f8

SIP/106-084585d0

Dial

SIP/106|20|iKkTtwW







97

92

ANSWERED

DOCUMENTATION

1.25E+09

If you subtract the 92 from the 97, you get the 5 second number you're
looking for.  These fields have actual names, but they aren't relevant to me
since I'm using the flat-text CDR Master.csv.

 

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlo Dimaggio
Sent: Tuesday, September 29, 2009 10:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Retrieve Call setup - QoS

 

Hi all,

 

I would like to know if there is a function/setting for extracting the  

call setup time in asterisk (1.4 or 1.6).

I need this value for every call processed by asterisk as specified in  

(ETSI TR 101 329-1 v3.1.2):

Call set-up time is the time elapsed from the end of the user  

interface command by the caller (keypad dialling, E-mail alias typing,  

etc.) to the receipt by the caller of a meaningful progress information

 

In other words:

  Call setup time = time from call start (caller press the # key -  

start the invite) to session progress (183) or ringing (180)

 

 

Thanks and best regards,

Carlo Dimaggio

 

___

-- Bandwidth and Colocation Provided by http://www.api-digital.com --

 

AstriCon 2009 - October 13 - 15 Phoenix, Arizona

Register Now: http://www.astricon.net

 

asterisk-users mailing list

To UNSUBSCRIBE or update options visit:

   http://lists.digium.com/mailman/listinfo/asterisk-users

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] digium fax: can't indicate condition 19?

[Scott L. Lykens]

> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of David Backeberg
> Sent: Monday, September 21, 2009 10:42 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] digium fax: can't indicate condition 19?
> 
> On Mon, Sep 21, 2009 at 9:26 PM, sean darcy 
> wrote:
> > In my attempts to set up digium fax I get an odd warning:
> >
> >     -- Executing [...@capture-fax:2] ReceiveFAX("SIP/173-b53023e8",
> > "/var/spool/asterisk/fax/20090921_1806.tif") in new stack
> >     -- Channel 'SIP/173-b53023e8' receiving fax
> > '/var/spool/asterisk/fax/20090921_1806.tif'
> > [Sep 21 18:06:37] WARNING[5149]: chan_sip.c:5561 sip_indicate: Don't
> > know how to indicate condition 19
> >     -- Channel 'SIP/173-b53023e8' fax session '7' started
> >
> > Is there some list of these conditions? Should I care?
> 
> I get a lot of fax messages that seem extraneous so I usually ignore
> them. There is already an open ticket want an SVN fix for one that I
> found particularly annoying. I'm talking about one that repeatedly
> talks about a missing segment or something like that, and it happens a
> gazillion times per fax. A gazillion being "way more than I want it
> to".
> 
> Did the fax come through okay? If yes I recommend ignoring the error
> unless you want to be particularly diligent and dig through the bug
> tracker to see if it's already been fixed.

I am getting the same error as the OP when trying to receive a fax with a T.38 
capable provider. The fax never comes through and usually fails with No Carrier 
from the sender.

There does not appear to be a relevant bug in the tracker.

As we are eliminating our PRI soon I am trying to get faxing via T.38 working 
properly. I'm not interested in running Callweaver next to Asterisk just to 
support fax. :/ 

Any insight into this error would be greatly appreciated.

Thanks!

sl

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

[asterisk-users] Static on the line randomly

[Julian Lyndon-Smith]

We've been having a strange problem all day where when making outbound
calls, all we get is static on the far end (i.e we can hear, they
can't).

We've restarted asterisk a couple of times to no avail. It now
transpires that it is only mobile numbers that are affected (not all
mobile networks, not all of the time)

Is this a supplier problem (BT ISDN/32) ?

Thanks

Julian

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] LDAP integration

[John A. Sullivan III]

On Tue, 2009-09-29 at 11:01 -0300, Rafael Seste wrote:
> Hi all,
> 
> I looked on the Internet but I didn't find any good how-to.
> I would like to integrate a ldap server ( with all users data) with
> asterisk to authenticate SIP users. With this solution I will only
> need to add a user on ldap, it will not be necessary to add any
> special configuration on sip.conf
> 
> Is that possible???If so, How can I configure this setup???
> 
> Thanks in advance
> 
I considered doing this using LDAP as a real-time database.  I decided
not to for two reasons which I'll share below. However, I am very new to
Asterisk so I would be very curious to know from more experienced folks
if my assumptions were false.

First, there were some good how-tos about using LDAP as a real-time
database but, if I recall, the schema is extended in such a way that the
regular user password is not the password used by Asterisk.

Second, I believe we saw a way we could map the Asterisk password to the
regular user password (it's been a while so I'm not sure about that) but
were concerned about the problems of entering secure passwords from a
phone keypad.  We enforce fairly secure passwords - at least nine
characters with some variety of characters and encourage much longer
passwords.  Having to enter lots of characters in both cases as well as
symbols seemed difficult from a phone keypad.  Thus, we decided
(reluctantly) to use separate simple passwords for phone access instead
of the very secure passwords we use to data access.

Hope this helps and looking forward to more informed comments than mine!
- John
-- 
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com

http://www.spiritualoutreach.com
Making Christianity intelligible to secular society


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

[asterisk-users] Retrieve Call setup - QoS

[Carlo Dimaggio]

Hi all,

I would like to know if there is a function/setting for extracting the  
call setup time in asterisk (1.4 or 1.6).
I need this value for every call processed by asterisk as specified in  
(ETSI TR 101 329-1 v3.1.2):
Call set-up time is the time elapsed from the end of the user  
interface command by the caller (keypad dialling, E-mail alias typing,  
etc.) to the receipt by the caller of a meaningful progress information

In other words:
Call setup time = time from call start (caller press the # key -  
start the invite) to session progress (183) or ringing (180)


Thanks and best regards,
Carlo Dimaggio

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] LDAP integration

[Antoine Patte]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Hi,

A realtime ldap driver exist.

He can put the user/peer sip/iax in a ldap directory and configuration
files.

A friend has updated as part of his final study of.

You can find-it there : http://wiki.ouranos.be/doku.php/stage:ldap

Or contact us at : magicrhe...@ouranos.be

For other questions, I'm a here ...

- --
Antoine Patte


Rafael Seste wrote:
> Hi all,
> 
> I looked on the Internet but I didn't find any good how-to.
> I would like to integrate a ldap server ( with all users data) with
> asterisk to authenticate SIP users. With this solution I will only
> need to add a user on ldap, it will not be necessary to add any
> special configuration on sip.conf
> 
> Is that possible???If so, How can I configure this setup???
> 
> Thanks in advance
> 

-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.10 (GNU/Linux)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iEYEARECAAYFAkrCJ6EACgkQBnIOcv+j7+wXtgCcDaoIAGZJfva39XFUtqRoMkih
XqYAoOnaMYa//DwR9F0doxtd3otPTeeF
=3D09
-END PGP SIGNATURE-

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] DAHDI channel congested busy

[Jerry Geis]

Shaun Ruffell wrote:
> On 09/29/2009 06:52 AM, Jerry Geis wrote:
>> A user report that this issue:
>>
>> https://issues.asterisk.org/view.php?id=15429
>>
>>
>> Has resolved their problem with a TDM card.
>>
>> My card is a T1/PRI card. Different module to load.
>> I have the same issue.
>>
>> Does this same problem exist in the PRI code and needs fixed their also?
>> Has it been fixed? and does this issue warrant a new release?
>>
>
> Unfortunately, if you're seeing this with a PRI code, it would be 
> completely unrelated to issue 15429.  Do you see anything interesting 
> when you enable pri intense debug and try to make an outbound call?

I will enable that now - keep the trace and when I see it again
re-enable it and compare. unfortunately that may take a couple weeks.

Thanks,

Jerry

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] DAHDI channel congested busy

[Shaun Ruffell]

On 09/29/2009 06:52 AM, Jerry Geis wrote:
> A user report that this issue:
>
> https://issues.asterisk.org/view.php?id=15429
>
>
> Has resolved their problem with a TDM card.
>
> My card is a T1/PRI card. Different module to load.
> I have the same issue.
>
> Does this same problem exist in the PRI code and needs fixed their also?
> Has it been fixed? and does this issue warrant a new release?
>

Unfortunately, if you're seeing this with a PRI code, it would be 
completely unrelated to issue 15429.  Do you see anything interesting 
when you enable pri intense debug and try to make an outbound call?

-- 
Shaun Ruffell
Digium, Inc. | Linux Kernel Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com & www.asterisk.org

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] chanspy and DISA

[Danny Nicholas]

What you could use would be an AMI interface that does a "core show channels
verbose" to get the active call information, then display that as an HTML
table.  When the supervisor clicks on the call he/she wants, the AMI
originates a chanspy/barge command as appropriate.  75% of the responders on
this can write one of these on their coffee break.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Millican
Sent: Tuesday, September 29, 2009 10:05 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] chanspy and DISA

Hello all,
OS OpenSuSE 10.3
* ver 1.4.26.2
zaptel ver. 1.12
Digium TE122

I have a request for remote users to be able to dial through the system
  so that the sales managers can barge/chanspy on the sales force.
I have the DISA part working with authentication(rather straight
forward) but what I can not figure out is how to enable the supervisors
to be able to barge on these calls.  Is there a way to find the channel
to barge on that would be usable by NON tech people?
Any thoughts?

TIA,
JohnM


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

[asterisk-users] chanspy and DISA

[John Millican]

Hello all,
OS OpenSuSE 10.3
* ver 1.4.26.2
zaptel ver. 1.12
Digium TE122

I have a request for remote users to be able to dial through the system
  so that the sales managers can barge/chanspy on the sales force.
I have the DISA part working with authentication(rather straight
forward) but what I can not figure out is how to enable the supervisors
to be able to barge on these calls.  Is there a way to find the channel
to barge on that would be usable by NON tech people?
Any thoughts?

TIA,
JohnM


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

[asterisk-users] Wrong hint, ringing when idle. after hangup.

[Leif Neland]

I have 3 phones, SIP/3, SIP/6 and SIP/9
SIP/3 subscribes on hint on SIP/9

Phone 6 calls phone 9, blf on phone 3 flashes until 9 picks up, then it 
is steady red. That's correct.
But when 9 hangs up the hint goes to "InUse&Ringing", the light on 3 is 
still flashing.

It keeps flashing until somebody calls 9 and hangs up again.

-- Executing [...@local:1] Dial("SIP/6-35236014", "SIP/9,120") in new 
stack
-- Called 9
  == Extension Changed 9[hintcontext] new state Ringing for Notify User 3
-- SIP/9-36a07014 is ringing
  == Extension Changed 9[hintcontext] new state InUse for Notify User 3
-- SIP/9-36a07014 answered SIP/6-35236014
-- Packet2Packet bridging SIP/6-35236014 and SIP/9-36a07014
-- Executing [...@local:1] Hangup("SIP/6-35236014", "") in new stack
  == Spawn extension (local, h, 1) exited non-zero on 'SIP/6-35236014'
  == Spawn extension (local, 9, 1) exited non-zero on 'SIP/6-35236014'
-- Executing [...@local:1] Hangup("SIP/6-35236014", "") in new stack
  == Spawn extension (local, h, 1) exited non-zero on 'SIP/6-35236014'
  == Extension Changed 9[hintcontext] new state Idle for Notify User 3
  == Extension Changed 9[hintcontext] new state InUse&Ringing for Notify 
User 3 (queued)
  == Extension Changed 9[hintcontext] new state InUse&Ringing for Notify 
User 3

Asterisk 1.6.0.15 built by root @ arnold.neland.dk on a i386 running 
FreeBSD on 2009-09-29 07:49:45 UTC

Leif




___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] LDAP integration

[Danny Nicholas]

You could get the Free PERL module Asterisk::Ldap and use it to periodically
update your users from the LDAP server.  You could make it a daily cron job
run at midnight so any new LDAP users would be Asterisk users the new
business day and you could also run the module on-demand.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rafael Seste
Sent: Tuesday, September 29, 2009 9:02 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] LDAP integration

Hi all,

I looked on the Internet but I didn't find any good how-to.
I would like to integrate a ldap server ( with all users data) with
asterisk to authenticate SIP users. With this solution I will only
need to add a user on ldap, it will not be necessary to add any
special configuration on sip.conf

Is that possible???If so, How can I configure this setup???

Thanks in advance

-- 
Rafael S. Seste

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

[asterisk-users] LDAP integration

[Rafael Seste]

Hi all,

I looked on the Internet but I didn't find any good how-to.
I would like to integrate a ldap server ( with all users data) with
asterisk to authenticate SIP users. With this solution I will only
need to add a user on ldap, it will not be necessary to add any
special configuration on sip.conf

Is that possible???If so, How can I configure this setup???

Thanks in advance

-- 
Rafael S. Seste

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Who am xxx talking to.agi

[Danny Nicholas]

IMO the easiest way to accomplish this would be to do an AMI call to "core
show channels verbose" and pick out the line containing the extension.  You
could also pick out the customer number so a record could be made if another
agent was talking to the customer.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leif Neland
Sent: Tuesday, September 29, 2009 4:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Who am xxx talking to.agi

In relation to our CRM-system I'd like to send a query to asterisk who 
is extension xxx talking to.

When the operator enters the page with customer data, the crm should 
send a query to asterisk, to get the cli of the call the operator is having.
If the number is matching the customers number in crm, a record will be 
made, if it is not, a popup "Are you talking with this customer now?", 
if confirmed, the number will be recorded in the crm.

Can asterisk answer this question?

I've tried using sip show channels and sip show peer, but the cli is not 
in an obvious place.

Is it better done by parsing logfile or storing numbers in the internal 
database from the dialplan?

Leif


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Native bridging analog phones trouble DAHDI channels.

[Kevin P. Fleming]

Maurizio Faccio adinet wrote:
> I own a TDM2400 board, with three FXO modules and one FXS.
> I'am having trouble with analog sip phones, from two different 
> equipment. (Grandstream GXW-4024 192.168.0.105, and Audiocodes MP202), 
> sometimes when I am calling someone, then I press flash, and then call 
> someone else, both calls stay connected after I hang up.

That's because you have just completed a flash-hook based transfer of
the first call to the second call. If you don't want this feature, set
'transfer=no' for the relevant channels in chan_dahdi.conf.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check us out at www.digium.com & www.asterisk.org

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

[asterisk-users] Native bridging analog phones trouble DAHDI channels.

[Maurizio Faccio adinet]

I own a TDM2400 board, with three FXO modules and one FXS.
I'am having trouble with analog sip phones, from two different 
equipment. (Grandstream GXW-4024 192.168.0.105, and Audiocodes MP202), 
sometimes when I am calling someone, then I press flash, and then call 
someone else, both calls stay connected after I hang up.

[Sep 29 07:18:06] VERBOSE[3218] logger.c: -- Called g2/16
[Sep 29 07:18:06] DEBUG[3218] chan_dahdi.c: Sent deferred digit string: T16w
[Sep 29 07:18:07] VERBOSE[3218] logger.c: -- DAHDI/9-1 answered 
SIP/130-085c6958
[Sep 29 07:18:17] VERBOSE[3218] logger.c: -- Started music on hold, 
class 'default', on DAHDI/9-1
[Sep 29 07:18:17] NOTICE[3218] rtp.c: Unknown RTP codec 126 received 
from '192.168.0.105'
[Sep 29 07:18:17] VERBOSE[3056] logger.c: -- Stopped music on hold 
on DAHDI/8-1
[Sep 29 07:18:17] VERBOSE[3056] logger.c: -- Stopped music on hold 
on DAHDI/9-1
[Sep 29 07:18:17] DEBUG[3056] chan_sip.c: SIP transfer: Succeeded to 
masquerade channels.
[Sep 29 07:18:17] DEBUG[3218] chan_dahdi.c: New owner for channel 8 is 
DAHDI/8-1
[Sep 29 07:18:17] DEBUG[3218] chan_dahdi.c: master: 8, slave: 9, 
nothingok: 0
[Sep 29 07:18:17] DEBUG[3218] chan_dahdi.c: Stopping tones on 8/0 
talking to 9/0
[Sep 29 07:18:17] DEBUG[3218] chan_dahdi.c: Stopping tones on 9/0 
talking to 8/0
[Sep 29 07:18:17] DEBUG[3218] chan_dahdi.c: Making 9 slave to master 8 at 0
[Sep 29 07:18:17] DEBUG[3218] chan_dahdi.c: Added 20 to conference 9/8
[Sep 29 07:18:17] DEBUG[3218] chan_dahdi.c: Added 19 to conference 9/9
[Sep 29 07:18:17] VERBOSE[3218] logger.c: -- Native bridging 
DAHDI/8-1 and DAHDI/9-1

I am using Elastix, 1.5.2-2.3. I do not know if the trouble is caused by 
elastix or some trouble with the digium board configuration. I am using 
DAHDI modules.


Dahdi show status.
Description  Alarms IRQ
bpviol CRC4 
Wildcard TDM2400P Board 1OK 2  
0  0

Core show version

Asterisk 1.4.25.1 built by root @ rpmbuild32.elastix.palosanto.com on a i686 
running Linux on 2009-06-14 11:49:25 UTC



I do not understand what it is happening.

Thank you in advance

Maurizio Faccio

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] DAHDI channel congested busy

[Jerry Geis]

A user report that this issue:

https://issues.asterisk.org/view.php?id=15429


Has resolved their problem with a TDM card.

My card is a T1/PRI card. Different module to load.
I have the same issue.

Does this same problem exist in the PRI code and needs fixed their also?
Has it been fixed? and does this issue warrant a new release?

Thanks,

Jerry

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Fax and dial-up connection issues

[Vinícius Fontes]

And of course I forgot the most important stuff:

Asterisk version: 1.4.22
DAHDI Linux: 2.2.0.2
DAHDI Tools: 2.2.0

- "Vinícius Fontes"  escreveu:

> I have a pretty large setup on one of my customers. Digium TE420B
> (with echo cancelling module), 3 Xorcom Astribanks with 32 FXS each
> and 1 Xorcom Astribank with 16 FXO. These FXO ports are NOT used for
> fax/data transmission, as they are connected to cell phones. Not
> really related to the issue, but there are also 250 SIP phones.
> 
> The problem is that fax and dial-up connections are really unreliable.
> Faxes send and received rarely get more than 2 pages without giving a
> transmission error and sending garbage to the other end. Not
> connecting at all (and stating COMMUNICATION ERROR) is not uncommon.
> As of dial-up connections, of course I'm not expecting to get 33600 on
> that, but this customer has some dedicated systems that must dial the
> "mothership" and use low speeds like 9600. I also have
> echocancelwhenbridged=no set up on all PRI and FXS channels.
> 
> I really don't know what else I could do to solve the problem.
> xpp_sync is set to DAHDI so the same sync received from telco is used
> on the FXS ports. For the TE420, only two spans are being used today.
> One of them is set as the sync master (1) and all others to slaves
> (0). Spans 1 and 3 are connected to the telco, the others are not
> being used.
> 
> Follows my config files (snipped to the relevant portions, because
> most are really big) and some other related info. Any hints on how to
> solve this will be really really appreciated.
> 
> 
> /etc/dahdi/system.conf:
> loadzone=br
> defaultzone=br
> 
> span=1,1,0,ccs,hdb3,crc4
> bchan=1-15,17-31
> hardhdlc=16
> 
> span=2,0,0,ccs,hdb3,crc4
> bchan=32-46,48-62
> hardhdlc=47
> 
> span=3,0,0,ccs,hdb3,crc4
> bchan=63-77,79-93
> hardhdlc=78
> 
> span=4,0,0,ccs,hdb3,crc4
> bchan=94-108,110-124
> hardhdlc=109
> 
> echocanceller=oslec,125-162
> echocanceller=oslec,163-200
> echocanceller=oslec,201-238
> echocanceller=oslec,239-254
> 
> fxoks=125-162
> fxoks=163-200
> fxoks=201-238
> fxsls=239-254
> 
> 
> 
> 
> /etc/asterisk/chan_dahdi.conf:
> [trunkgroups]
> 
> [channels]
> language=pt_BR
> group=1
> switchtype=euroisdn
> pridialplan=unknown
> prilocaldialplan=unknown
> priindication=outofband
> signalling=pri_cpe
> echocancel=yes
> echocancelwhenbridged=no
> relaxdtmf=no
> 
> switchtype=euroisdn
> context=e1-embratel
> group=1
> signalling=pri_cpe
> channel =>1-15,17-31
> 
> switchtype=euroisdn
> context=e1-bp250
> group=2
> signalling=pri_net
> overlapdial=yes
> channel =>32-46,48-62
> 
> switchtype=euroisdn
> context=e1-embratel
> group=1
> signalling=pri_cpe
> channel => 63-77,79-91
> 
> switchtype=euroisdn
> echocancel=yes
> context=e1-embratel
> group=3
> signalling=pri_cpe
> channel =>94-108,110-124
> 
> context=ddd
> echocancel=256
> echocancelwhenbridged=no
> usecallerid=yes
> cidsignalling=bell
> cidstart=ring
> threewaycalling=yes
> callwaiting=no
> transfer=yes
> relaxdtmf=no
> txgain=0.0
> rxgain=0.0
> language=pt_BR
> cancallforward=yes
> signalling=fxo_ks
> callgroup=1
> pickupgroup=1
> ;flash=500
> ;rxflash=500
> 
> ;Cadencias padrao
> cadence=125,125,2000,-4000
> cadence=250,250,500,1000,250,250,500,-4000
> cadence=125,125,125,125,125,-4000
> cadence=1000,500,2500,-5000
> 
> ;Chamada interna
> cadence=500,300,500,3500,500,300,500,-4000
> 
> ;Ring continuo
> cadence=1,1,6,1
> 
> callerid="" <7875>
> context=fax
> callwaiting=no
> callgroup=3
> pickupgroup=3
> mailbox=7875
> channel => 125
> 
> 
> /etc/asterisk/extensions.conf:
> [fax]
> ignorepat => 0
> include => local
> 
> ;Ligacoes locais
> exten => _0,1,SetTransferCapability(3K1AUDIO)
> exten => _0,n,Dial(DAHDI/g1/${EXTEN:1},60)
> 
> ;Ligacoes DDD - telefones fixos
> exten => _00XX[2-6]XXX,1,SetTransferCapability(3K1AUDIO)
> exten => _00XX[2-6]XXX,n,Dial(DAHDI/g1/021${EXTEN:2},60)
> 
> 
> # dahdi_test:
> svoip01:~# dahdi_test -vv
> Opened pseudo dahdi interface, measuring accuracy...
> 
> 8192 samples in 8199.664 system clock sample intervals (100.094%)
> 8192 samples in 8198.728 system clock sample intervals (100.082%)
> 8192 samples in 8191.720 system clock sample intervals (99.997%)
> 8192 samples in 8190.992 system clock sample intervals (99.988%)
> 8192 samples in 8191.456 system clock sample intervals (99.993%)
> 8192 samples in 8191.664 system clock sample intervals (99.996%)
> 8192 samples in 8191.880 system clock sample intervals (99.999%)^C
> --- Results after 7 passes ---
> Best: 99.999 -- Worst: 99.906 -- Average: 99.970915, Difference:
> 100.021108
> 
> 
> # xpp_sync:
> svoip01:~# xpp_sync 
> Current sync: DAHDI
> Best Available Syncers:
>XBUS-01 (@usb-:00:1d.7-6) [usb:142]  [ FXO*2 ]
>XBUS-00 (@usb-:00:1d.7-5) [usb:1254] [ FXS*4 ]
>XBUS-02 (@usb-:00:1d.7-3.1)   [usb:X1036520] [ FXS*4 ]
>XBUS-03 (@usb-:00:1d.7-3.2)   [usb:X1036521] [ FXS*4 ]
> ===

[asterisk-users] Music On Hold

[Cyprus VoIP]

Hello,

We need help in debugging Music On Hold on our Asterisk 1.6.1.6

 From the SIP debug, I see that an extension sends an INVITE of the call 
to the Asterisk, whenever the HOLD or Transfer buttons are pressed, but 
I don't see in the console any reference to the call being placed on hold.

When I typed "moh show files", I see the wav files of the 
/var/lib/asterisk/moh folder.

How can I debug this?

Thanks.

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

[asterisk-users] Fax and dial-up connection issues

[Vinícius Fontes]

I have a pretty large setup on one of my customers. Digium TE420B (with echo 
cancelling module), 3 Xorcom Astribanks with 32 FXS each and 1 Xorcom Astribank 
with 16 FXO. These FXO ports are NOT used for fax/data transmission, as they 
are connected to cell phones. Not really related to the issue, but there are 
also 250 SIP phones.

The problem is that fax and dial-up connections are really unreliable. Faxes 
send and received rarely get more than 2 pages without giving a transmission 
error and sending garbage to the other end. Not connecting at all (and stating 
COMMUNICATION ERROR) is not uncommon. As of dial-up connections, of course I'm 
not expecting to get 33600 on that, but this customer has some dedicated 
systems that must dial the "mothership" and use low speeds like 9600. I also 
have echocancelwhenbridged=no set up on all PRI and FXS channels.

I really don't know what else I could do to solve the problem. xpp_sync is set 
to DAHDI so the same sync received from telco is used on the FXS ports. For the 
TE420, only two spans are being used today. One of them is set as the sync 
master (1) and all others to slaves (0). Spans 1 and 3 are connected to the 
telco, the others are not being used.

Follows my config files (snipped to the relevant portions, because most are 
really big) and some other related info. Any hints on how to solve this will be 
really really appreciated.


/etc/dahdi/system.conf:
loadzone=br
defaultzone=br

span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
hardhdlc=16

span=2,0,0,ccs,hdb3,crc4
bchan=32-46,48-62
hardhdlc=47

span=3,0,0,ccs,hdb3,crc4
bchan=63-77,79-93
hardhdlc=78

span=4,0,0,ccs,hdb3,crc4
bchan=94-108,110-124
hardhdlc=109

echocanceller=oslec,125-162
echocanceller=oslec,163-200
echocanceller=oslec,201-238
echocanceller=oslec,239-254

fxoks=125-162
fxoks=163-200
fxoks=201-238
fxsls=239-254




/etc/asterisk/chan_dahdi.conf:
[trunkgroups]

[channels]
language=pt_BR
group=1
switchtype=euroisdn
pridialplan=unknown
prilocaldialplan=unknown
priindication=outofband
signalling=pri_cpe
echocancel=yes
echocancelwhenbridged=no
relaxdtmf=no

switchtype=euroisdn
context=e1-embratel
group=1
signalling=pri_cpe
channel =>1-15,17-31

switchtype=euroisdn
context=e1-bp250
group=2
signalling=pri_net
overlapdial=yes
channel =>32-46,48-62

switchtype=euroisdn
context=e1-embratel
group=1
signalling=pri_cpe
channel => 63-77,79-91

switchtype=euroisdn
echocancel=yes
context=e1-embratel
group=3
signalling=pri_cpe
channel =>94-108,110-124

context=ddd
echocancel=256
echocancelwhenbridged=no
usecallerid=yes
cidsignalling=bell
cidstart=ring
threewaycalling=yes
callwaiting=no
transfer=yes
relaxdtmf=no
txgain=0.0
rxgain=0.0
language=pt_BR
cancallforward=yes
signalling=fxo_ks
callgroup=1
pickupgroup=1
;flash=500
;rxflash=500

;Cadencias padrao
cadence=125,125,2000,-4000
cadence=250,250,500,1000,250,250,500,-4000
cadence=125,125,125,125,125,-4000
cadence=1000,500,2500,-5000

;Chamada interna
cadence=500,300,500,3500,500,300,500,-4000

;Ring continuo
cadence=1,1,6,1

callerid="" <7875>
context=fax
callwaiting=no
callgroup=3
pickupgroup=3
mailbox=7875
channel => 125


/etc/asterisk/extensions.conf:
[fax]
ignorepat => 0
include => local

;Ligacoes locais
exten => _0,1,SetTransferCapability(3K1AUDIO)
exten => _0,n,Dial(DAHDI/g1/${EXTEN:1},60)

;Ligacoes DDD - telefones fixos
exten => _00XX[2-6]XXX,1,SetTransferCapability(3K1AUDIO)
exten => _00XX[2-6]XXX,n,Dial(DAHDI/g1/021${EXTEN:2},60)


# dahdi_test:
svoip01:~# dahdi_test -vv
Opened pseudo dahdi interface, measuring accuracy...

8192 samples in 8199.664 system clock sample intervals (100.094%)
8192 samples in 8198.728 system clock sample intervals (100.082%)
8192 samples in 8191.720 system clock sample intervals (99.997%)
8192 samples in 8190.992 system clock sample intervals (99.988%)
8192 samples in 8191.456 system clock sample intervals (99.993%)
8192 samples in 8191.664 system clock sample intervals (99.996%)
8192 samples in 8191.880 system clock sample intervals (99.999%)^C
--- Results after 7 passes ---
Best: 99.999 -- Worst: 99.906 -- Average: 99.970915, Difference: 100.021108


# xpp_sync:
svoip01:~# xpp_sync 
Current sync: DAHDI
Best Available Syncers:
   XBUS-01 (@usb-:00:1d.7-6) [usb:142]  [ FXO*2 ]
   XBUS-00 (@usb-:00:1d.7-5) [usb:1254] [ FXS*4 ]
   XBUS-02 (@usb-:00:1d.7-3.1)   [usb:X1036520] [ FXS*4 ]
   XBUS-03 (@usb-:00:1d.7-3.2)   [usb:X1036521] [ FXS*4 ]
==
WARNING: FXO which is not the syncer cause bad PCM
 Affected Astribanks are:
--
XBUS-01
==



# cat /proc/interrupts 
   CPU0   CPU1   
  0:  13949  0   IO-APIC-edge  timer
  1:   4056  0   IO-APIC-edge  i8042
  8:  1  0   IO-APIC-edge  rtc0
  9:  0  0   IO-APIC-fasteoi   acpi

Re: [asterisk-users] OT - In which countries are ISDN subaddressesused ?

[Alec Davis]

When I say no reliable internet, some days it's good, others it's not, so to
try to push a call over an IAX trunk is going to fail.

For the choice of providers, when it comes to a business with branches
around the country (NZ is small enough) we'd choose one provider.

Subaddress is not limited to PRI, it's also available on BRI lines.

We have 2 PRI connected sites and 1 BRI ISDN site, all 3 with asterisk, and
non overlapping extension dial plans.
SITE A: 8500-8599
SITE B: 8800-8899
SITE C: 8300-8399

The flow goes something like this:
Site A:
8501 dials 8801
dialplan sets CallingSubaddr to 8501
dialplan calls SITE B's main number (perhaps 09 1234567) with
CalledSubaddr set to 8801.

Site B:
the main number (09 1234567) is processed by asterisk
dialplan sees that CallingSubaddr is set to 8501, so sets CallerId
to 8501
dialplan sees that CalledSubaddr was set to 8801 so dials 8801.

Regarding POTS it will not work, but some ISDN PABX's have the dialling
concept of dialling a '*' as the separator between Number and SubAddress and
'#' as end of dialling.

Alec Davis
 

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F
Sent: Tuesday, 29 September 2009 7:20 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] OT - In which countries are ISDN
subaddressesused ?

On Mon, Sep 28, 2009 at 11:51 PM, Alec Davis 
wrote:
> I'm interested, and I expect others will be on how you might use it.
>
> Our use on the mantis bug, is to allow 3 ISDN connected sites (no 
> reliable
> internet) each running asterisk, to dial other staff members in the 
> other


You say no reliable internet, if you can get ISDN wouldn't the providers
offer Internet thru the same ISDN Physical links? why are those not
considered reliable? What I mean is what we have here in the states called
Internet over T1, I know I am assuming that your ISDN links are T/E1 and if
that's not the case I know the answer is price.
But if it is T/E1 then I could see a fractional T/E1 where some channels are
used for data as a way better/easier/cheaper way than subaddressing.

> branches.
> The key to this working is the subaddress being populated with the 
> dailled extension, but the main number is called.

This is interesting, wouldn't the provider on both ends have to support
this?
While it could work with ISDN lines, I don't think it could ever work with
POTS, since the switches servicing these POTS lines have dialplans
programmed that after (US example) 7 digits - where the first digit is
something other than 0 or 1 and the next 2 digits are not 11 - it starts
calling not listening/taking any additional digits.
In the case where a 1 is dialed it's expecting 10 more digits with the first
3 having the same rules as the first 3 of the 7 digits, in an area where 10
digit dialing is required there is usually one more rule that where the
first 3 digits is the same as the area code of the phone making the call,
then it waits for 7 more digits with the first
3 of the 7 having the same rule as the 7 digit dialing above.
In any event you can't really dial more digits than the national dialing
plan when calling national from POTS here in the US.

>
> The dialplan see's the incoming call, checks the calledsubaddress and 
> then directs the call to the appropriate extension, all in a split 
> second. No IVR, no DDI.
>
> The natural answer is why not use DDI, that's one for each staff 
> member, could get a bit expensive.
>
> Alec
> 
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
> Sent: Tuesday, 29 September 2009 3:07 p.m.
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] OT - In which countries are ISDN 
> subaddressesused ?
>
>
>
> 2009/9/26 Alec Davis 
>>
>> When did that happen? Added to libpri, someone beat me to it.
>
> Hello Alec,
>
> In this question, I was refering to your ongoing work I'm sorry if 
> this question let anyone believe this ISDN subaddresswas already done 
> and pushed into libpri trunk.
>
> Regards
>
>>
>>
>> Alec Davis
>>
>> 
>> From: asterisk-users-boun...@lists.digium.com
>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
>> Sent: Saturday, 26 September 2009 9:06 a.m.
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: [asterisk-users] OT - In which countries are ISDN 
>> subaddresses used ?
>>
>> Hi,
>>
>> I've seen this ISDN subaddress feature added to libpri.
>> Which countries are using it ?
>> How is this billed ? Do you have to pay an extra to your telco to 
>> benefit from this subaddresses ?
>>
>> Cheers
>>
>> ___
>> -- Bandwidth and Colocation Provided by http://www.api-digital.co

[asterisk-users] Who am xxx talking to.agi

[Leif Neland]

In relation to our CRM-system I'd like to send a query to asterisk who 
is extension xxx talking to.

When the operator enters the page with customer data, the crm should 
send a query to asterisk, to get the cli of the call the operator is having.
If the number is matching the customers number in crm, a record will be 
made, if it is not, a popup "Are you talking with this customer now?", 
if confirmed, the number will be recorded in the crm.

Can asterisk answer this question?

I've tried using sip show channels and sip show peer, but the cli is not 
in an obvious place.

Is it better done by parsing logfile or storing numbers in the internal 
database from the dialplan?

Leif


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] play audio file within an active call

[Matthew Edmondson]

I didn't think of Benny's solution. That would be the way to go as it is a
core asterisk command.

On Tue, Sep 29, 2009 at 6:47 PM, Vieri  wrote:

>
> --- On Tue, 9/29/09, Matthew Edmondson  wrote:
>
> > If you redirect the channel, the
> > person they're talking to is likely
> > to be dropped.
>
> Thanks for pointing that out. So it sounds like RedirectChannel() is
> similar to Transfer().
>
> > The only way I know of doing this is with a conference
> > bridge like
> > meetme. You would have to have both parties in the
> > conference and then
> > call the 3rd party (your msg) into it.
>
> This may be trivial but how can I "force" both parties to enter a
> conference (eg. meetme)?
>
> Also, once they're in the conference and I've "called" a third party
> playing a sound file, how can I "force" them to exit the conference and
> revert to their bridged call as before the conference? Or, if I have to keep
> them within the conference/meetme, then I'd have to make sure that the "3rd
> party" can :
> 1- play a msg such as "on the phone too long; consider hanging up"
> 2- wait N minutes
> 3- play "you have Z minutes of conversation left. Call will be hung up
> automatically"
> 4- hang up both parties in conference
>
> Does this make sense?
>
> Thanks,
>
> Vieri
>
> > On Tue, Sep 29, 2009 at 6:05 PM, Vieri 
> > wrote:
> > > Hi,
> > >
> > > I'm wondering if someone can share their thoughts on
> > how to implement a system that periodically checks active
> > channels which have been up for more than X minutes and
> > plays/injects a sound file. The idea is to simply warn users
> > that they've been on the phone for quite a while and maybe
> > they should consider hanging up. If the call stays up for
> > more than Y minutes, it is dropped automatically
> > (softhangup).
> > >
> > > What's the simplest approach to playing a sound file
> > within an active channel?
> > >
> > > I thought of writing a cron agi script that scans
> > active channels, retrieves their duration and if it's > X
> > minutes then "RedirectChannel" to a context which executes a
> > Playback(file); if it's > Y minutes then
> > "RedirectChannel" to a context which executes both a
> > Playback("forcing hang up now...") and HangUp.
> > >
> > > Any thoughts?
> > >
>
>
>
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2009 - October 13 - 15 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] play audio file within an active call

[Vieri]

--- On Tue, 9/29/09, Benny Amorsen  wrote:

> > I'm wondering if someone can share their thoughts on
> how to implement a system that periodically checks active
> channels which have been up for more than X minutes and
> plays/injects a sound file. The idea is to simply warn users
> that they've been on the phone for quite a while and maybe
> they should consider hanging up. If the call stays up for
> more than Y minutes, it is dropped automatically
> (softhangup).
> >
> > What's the simplest approach to playing a sound file
> within an active channel?
> 
> I think you should be able to do this with ChanSpy and the
> whisper
> option. However, Asterisk already has a facility for this.
> This is from
> core show application Dial
> 
>     L(x[:y][:z]) - Limit the call to 'x' ms. Play
> a warning when 'y' ms are
>            left. Repeat
> the warning every 'z' ms. The following special
>            variables can
> be used with this option:
>            *
> LIMIT_PLAYAUDIO_CALLER   yes|no (default
> yes)
>                
>                
>       Play sounds to the caller.
>            *
> LIMIT_PLAYAUDIO_CALLEE   yes|no
>                
>                
>       Play sounds to the callee.
>            *
> LIMIT_TIMEOUT_FILE       File to
> play when time is up.
>            *
> LIMIT_CONNECT_FILE       File to
> play when call begins.
>            *
> LIMIT_WARNING_FILE       File to
> play as warning if 'y' is defined.
>                
>                
>       The default is to say the time
> remaining.

Benny,
thanks a lot. Will try that out.

Vieri



  

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] play audio file within an active call

[Vieri]

--- On Tue, 9/29/09, Matthew Edmondson  wrote:

> If you redirect the channel, the
> person they're talking to is likely
> to be dropped.

Thanks for pointing that out. So it sounds like RedirectChannel() is similar to 
Transfer().

> The only way I know of doing this is with a conference
> bridge like
> meetme. You would have to have both parties in the
> conference and then
> call the 3rd party (your msg) into it.

This may be trivial but how can I "force" both parties to enter a conference 
(eg. meetme)?

Also, once they're in the conference and I've "called" a third party playing a 
sound file, how can I "force" them to exit the conference and revert to their 
bridged call as before the conference? Or, if I have to keep them within the 
conference/meetme, then I'd have to make sure that the "3rd party" can :
1- play a msg such as "on the phone too long; consider hanging up"
2- wait N minutes
3- play "you have Z minutes of conversation left. Call will be hung up 
automatically"
4- hang up both parties in conference

Does this make sense?

Thanks,

Vieri

> On Tue, Sep 29, 2009 at 6:05 PM, Vieri 
> wrote:
> > Hi,
> >
> > I'm wondering if someone can share their thoughts on
> how to implement a system that periodically checks active
> channels which have been up for more than X minutes and
> plays/injects a sound file. The idea is to simply warn users
> that they've been on the phone for quite a while and maybe
> they should consider hanging up. If the call stays up for
> more than Y minutes, it is dropped automatically
> (softhangup).
> >
> > What's the simplest approach to playing a sound file
> within an active channel?
> >
> > I thought of writing a cron agi script that scans
> active channels, retrieves their duration and if it's > X
> minutes then "RedirectChannel" to a context which executes a
> Playback(file); if it's > Y minutes then
> "RedirectChannel" to a context which executes both a
> Playback("forcing hang up now...") and HangUp.
> >
> > Any thoughts?
> >


  

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] play audio file within an active call

[Benny Amorsen]

Vieri  writes:

> Hi,
>
> I'm wondering if someone can share their thoughts on how to implement a 
> system that periodically checks active channels which have been up for more 
> than X minutes and plays/injects a sound file. The idea is to simply warn 
> users that they've been on the phone for quite a while and maybe they should 
> consider hanging up. If the call stays up for more than Y minutes, it is 
> dropped automatically (softhangup).
>
> What's the simplest approach to playing a sound file within an active channel?

I think you should be able to do this with ChanSpy and the whisper
option. However, Asterisk already has a facility for this. This is from
core show application Dial

L(x[:y][:z]) - Limit the call to 'x' ms. Play a warning when 'y' ms are
   left. Repeat the warning every 'z' ms. The following special
   variables can be used with this option:
   * LIMIT_PLAYAUDIO_CALLER   yes|no (default yes)
  Play sounds to the caller.
   * LIMIT_PLAYAUDIO_CALLEE   yes|no
  Play sounds to the callee.
   * LIMIT_TIMEOUT_FILE   File to play when time is up.
   * LIMIT_CONNECT_FILE   File to play when call begins.
   * LIMIT_WARNING_FILE   File to play as warning if 'y' is defined.
  The default is to say the time remaining.


/Benny


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] play audio file within an active call

[Matthew Edmondson]

If you redirect the channel, the person they're talking to is likely
to be dropped.

The only way I know of doing this is with a conference bridge like
meetme. You would have to have both parties in the conference and then
call the 3rd party (your msg) into it.

On Tue, Sep 29, 2009 at 6:05 PM, Vieri  wrote:
> Hi,
>
> I'm wondering if someone can share their thoughts on how to implement a 
> system that periodically checks active channels which have been up for more 
> than X minutes and plays/injects a sound file. The idea is to simply warn 
> users that they've been on the phone for quite a while and maybe they should 
> consider hanging up. If the call stays up for more than Y minutes, it is 
> dropped automatically (softhangup).
>
> What's the simplest approach to playing a sound file within an active channel?
>
> I thought of writing a cron agi script that scans active channels, retrieves 
> their duration and if it's > X minutes then "RedirectChannel" to a context 
> which executes a Playback(file); if it's > Y minutes then "RedirectChannel" 
> to a context which executes both a Playback("forcing hang up now...") and 
> HangUp.
>
> Any thoughts?
>
>
>
>
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2009 - October 13 - 15 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

[asterisk-users] play audio file within an active call

[Vieri]

Hi,

I'm wondering if someone can share their thoughts on how to implement a system 
that periodically checks active channels which have been up for more than X 
minutes and plays/injects a sound file. The idea is to simply warn users that 
they've been on the phone for quite a while and maybe they should consider 
hanging up. If the call stays up for more than Y minutes, it is dropped 
automatically (softhangup).

What's the simplest approach to playing a sound file within an active channel?

I thought of writing a cron agi script that scans active channels, retrieves 
their duration and if it's > X minutes then "RedirectChannel" to a context 
which executes a Playback(file); if it's > Y minutes then "RedirectChannel" to 
a context which executes both a Playback("forcing hang up now...") and HangUp.

Any thoughts?



  

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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