Christian | 1 Nov 2008 01:23
Favicon

Re: Asterisk installation

Hi,
Many thanks for the info. Just a question, will I need to isntall anything else now?
Will i only need to install libpri and Asterisk from now on?
Best regards and thanks,
Christian

On 2008-11-01 at 09:06 David Klaverstyn wrote:

>Nothing changes except for the files.
>
>/etc/zaptel.conf becomes /etc/dahdi/system.conf
>/etc/asterisk/zapata.conf becomes /etc/asterisk/chan_dahdi.conf
>
>-----Original Message-----
>From: asterisk-users-bounces <at> lists.digium.com
>[mailto:asterisk-users-bounces <at> lists.digium.com] On Behalf Of Christian
>Sent: Saturday, 1 November 2008 8:49 AM
>To: dhartman <at> djhsolutions.com; asterisk-users <at> lists.digium.com;
>asterisk-users <at> lists.digium.com
>Subject: Re: [asterisk-users] Asterisk installation
>
>Hello,
>Many thanks for the info.
>OK, I didn't know that. I just installed it. Usually I read the included
>read me files and so on but not at this time.
>But I will be able to use my old zaptel hardware that i used with v1.4?
>Many thanks,
>Christian
>
>
(Continue reading)

Robert Lister | 1 Nov 2008 01:22

Re: Blank Voicemail.Conf after Password Change

On Wed, Oct 29, 2008 at 01:13:56PM -0400, Leah Newmark wrote:

> From time to time, voicemail.conf would go blank. We finally tracked it 
> down to happening when someone attempts to change their password.
> It seems the file is touched, but not written to, and we're left with a 
> blank voicemail file.
> 
> Permissions seem to be fine:
> -rw-rw-r-- 1 asterisk asterisk 12707 2008-10-29 12:14 
> /etc/asterisk/voicemail.conf

I believe what it does it create a new file called voicemail.conf.new in the 
same directory and then copies it into place, so worth checking the 
permissions on the directory as well, that asterisk can write to it.

> Asterisk is running as asterisk:
> 24560 ?        Ssl  409:34 /usr/sbin/asterisk -U asterisk

I see your asterisk is running "-U asterisk" but this ps output is 
ambiguous. What does ps xaguwww show?

if it really is running as UID asterisk, you should see 
something like:

asterisk  8506  0.0  0.6 443672 12912 ?  Ssl  Oct02  31:46 /usr/sbin/asterisk -U asterisk -G asterisk

> Nothing generated from voicemail is showing up in the asterisk logs, nor 
> does the console show any error after changing a password.

Otherwise, it could be some sort of odd file locking issue where multiple 
(Continue reading)

Jonn R Taylor | 1 Nov 2008 02:49
Favicon

Re: fax / t38 gateway

Thanks Kristian I will checkout the new script and see how it goes!

Jonn

-----Original Message-----
From: asterisk-users-bounces <at> lists.digium.com
[mailto:asterisk-users-bounces <at> lists.digium.com] On Behalf Of Kristian Kielhofner
Sent: Friday, October 31, 2008 1:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] fax / t38 gateway

On 10/31/08, Jonn R Taylor <jonnt <at> taylortelephone.com> wrote:
> Here is the QOS script that I use on my bridge.
>
>  http://www.taylortelephone.com/asterisk/astshape

  You should upgrade to the newer astshape script.  It classifies
traffic using iptables, which is much more flexible.  It also has beta
support for the HFSC qdisc:

http://astlinux.svn.sourceforge.net/viewvc/astlinux/trunk/package/iproute2/astshape

--

-- 
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com

_______________________________________________
(Continue reading)

Jeff LaCoursiere | 1 Nov 2008 02:52

Re: giving a user asterisk CLI access: how bad could it get


I think everyone is missing the point of the question.  He wants to know
if the user's shell is set to rasterisk, can they then use the CLI to get
a command shell.

The answer is "yes, they can", and in that case it may not be such a
good idea.  As someone else suggested, you can run a shell with "!".  I
imagine this could be compiled out of the CLI if you were so inclined.

j

On Sat, 1 Nov 2008, Tzafrir Cohen wrote:

> On Sat, Nov 01, 2008 at 12:38:52AM +0100, Dima wrote:
> > Setting the user's shell to /usr/sbin/rasterisk works. On login user
> > gets into asterisk CLI if asterisk is running (user just has to have
> > write permission to /var/lib/asterisk.*).
>
> How does that user "login"?
>
> --
>                Tzafrir Cohen
> icq#16849755              jabber:tzafrir.cohen <at> xorcom.com
> +972-50-7952406           mailto:tzafrir.cohen <at> xorcom.com
> http://www.xorcom.com  iax:guest <at> local.xorcom.com/tzafrir
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
(Continue reading)

Robert Lister | 1 Nov 2008 03:31

Re: twice normal beep before busy tone ??

On Fri, Oct 31, 2008 at 11:39:31PM +0000, Robert Lister wrote:
> On Fri, Oct 31, 2008 at 08:18:32AM +0100, Stefan Guenther wrote:
> > Hi,
> > 
> > I have a strange problem with our Asterisk installation. Outgoing calls 
> > are handled by the following lines:
> > 
> > exten => _0[2-9]X.,1,Set(CALLERID(num)=09999403${CALLERID(num)})
> > exten => _0[2-9]X.,2,SET(CALLERID(num)=${IF($[ ${CALLERID(num)} = 
> > 0999940321]?099994030:${CALLERID(num)})})
> > exten => _0[2-9]X.,3,DIAL(CAPI/g1/${CALLERID(num)}:${EXTEN},180,tr)
> > exten => _0[2-9]X.,4,GOTO(fehler,s-${DIALSTATUS},1)
> 
> What happens if you do Answer() before the Dial?

Also try without the "r" option to the dial command:

http://www.voip-info.org/wiki-Asterisk+cmd+dial

Rob

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

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

OCG Technical Support | 1 Nov 2008 05:36
Picon
Favicon

VoIP traffic shaping

This was so interesting I had to move it to its own thread!

 

Is anyone using this script?  How does it perform compared to the older WonderShaper script?

 

-M-

 

==================

 

Thanks Kristian I will checkout the new script and see how it goes!

 

Jonn

 

-----Original Message-----

From: asterisk-users-bounces <at> lists.digium.com [mailto:asterisk-users-bounces <at> lists.digium.com] On Behalf Of Kristian Kielhofner

Sent: Friday, October 31, 2008 1:32 AM

To: Asterisk Users Mailing List - Non-Commercial Discussion

Subject: Re: [asterisk-users] fax / t38 gateway

 

On 10/31/08, Jonn R Taylor <jonnt <at> taylortelephone.com> wrote:

> Here is the QOS script that I use on my bridge.

>

 

http://www.taylortelephone.com/asterisk/astshape

 

  You should upgrade to the newer astshape script.  It classifies

traffic using iptables, which is much more flexible.  It also has beta

support for the HFSC qdisc:

 

http://astlinux.svn.sourceforge.net/viewvc/astlinux/trunk/package/iproute2/astshape

 

--

Kristian Kielhofner

http://blog.krisk.org

http://www.submityoursip.com

http://www.astlinux.org

http://www.star2star.com

 

_______________________________________________

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

 

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 --

 

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 --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
Dan Austin | 1 Nov 2008 07:25
Favicon

Wierd queue question

I have just setup a small queue implementation for one
of my branch offices, replacing a 16 year old key system
that had a hacked together pseudo call queuing feature.

The 'agents' are not dedicated to the queues and want to
be able to logon and get one call only from the queue.
I know this is odd, but it is how my users want it to
work.

I have the login process setup using dynamic agents and
set a wrap-up time long enough for the agent to logout.
They have accepted this as a short term solution, but
they really want to be automatically logged out after
taking one and only one call.

Any tips or hints on how to accomplish this would be
greatly appreciated.

Thanks,
Dan

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

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

Julian Lyndon-Smith | 1 Nov 2008 08:24

Re: Wierd queue question

show application RemoveQueueMember
  -= Info about application 'RemoveQueueMember' =-

[Synopsis]
Dynamically removes queue members

[Description]
   RemoveQueueMember(queuename[|interface[|options]]):
Dynamically removes interface to an existing queue
If the interface is NOT in the queue and there exists an n+101 priority
then it will then jump to this priority.  Otherwise it will return an error
The option string may contain zero or more of the following characters:
       'j' -- jump to +101 priority when appropriate.
  This application sets the following channel variable upon completion:
     RQMSTATUS      The status of the attempt to remove a queue member as a
                     text string, one of
           REMOVED | NOTINQUEUE | NOSUCHQUEUE
Example: RemoveQueueMember(techsupport|SIP/3000)

Julian

Dan Austin wrote:
> I have just setup a small queue implementation for one
> of my branch offices, replacing a 16 year old key system
> that had a hacked together pseudo call queuing feature.
>
> The 'agents' are not dedicated to the queues and want to
> be able to logon and get one call only from the queue.
> I know this is odd, but it is how my users want it to
> work.
>
> I have the login process setup using dynamic agents and
> set a wrap-up time long enough for the agent to logout.
> They have accepted this as a short term solution, but
> they really want to be automatically logged out after
> taking one and only one call.
>
> Any tips or hints on how to accomplish this would be
> greatly appreciated.
>
> Thanks,
> Dan
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
>   

______________________________________________________________________
This email has been scanned by the MessageLabs Email Security System.
For more information please visit http://www.messagelabs.com/email 
______________________________________________________________________

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

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

Rodolfo Alcazar Portillo | 1 Nov 2008 16:15
Picon

SPA3102 interdigit timers bug?

Hi. I have a SPA3102 updated with with Software Version: 5.1.7(GW).

I have this settings on Voice/Regional:

Interdigit Long Timer:  10
Interdigit Short Timer: 3

Anyway, when hooking up (without dialing anything), the timeout starts
after 3 seconds. It's like the Long Timer is unused. After dialing, the
Short Timer is also used to timeout.

Is that normal? Am I missing something?

Thanks.
-- 
Rodolfo Alcazar
Responsable red y datos

Deutsche Gesellschaft für
Technische Zusammenarbeit (GTZ) GmbH

Programa de Apoyo a la Gestión Pública Descentralizada y
Lucha Contra La Pobreza - PADEP
Av. Sánchez Lima 2226
La Paz, Bolivia

Tel: +591 22417628 (121)
Fax: +591 22417628 (126)
Web: www.padep.org.bo
Email: rodolfo.alcazar <at> padep.org.bo

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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
Dan Austin | 1 Nov 2008 17:20
Favicon

Re: Wierd queue question

Julian wrote:
> show application RemoveQueueMember
>  -= Info about application 'RemoveQueueMember' =-

> [Synopsis]
> Dynamically removes queue members

> [Description]
>   RemoveQueueMember(queuename[|interface[|options]]):
> Dynamically removes interface to an existing queue
> If the interface is NOT in the queue and there exists an n+101 priority
> then it will then jump to this priority.  Otherwise it will return an error
> The option string may contain zero or more of the following characters:
>        'j' -- jump to +101 priority when appropriate.
>   This application sets the following channel variable upon completion:
>     RQMSTATUS      The status of the attempt to remove a queue member as a
>                     text string, one of
>           REMOVED | NOTINQUEUE | NOSUCHQUEUE
> Example: RemoveQueueMember(techsupport|SIP/3000)

> Julian

I should have mentioned that I already added a method for the agents to
logout using RemoveQueueMember.  What I am looking for is a way to trigger
it automatically, after the agent logs in and gets one call.

I admit I have not tried the simple and crude method:
exten => 123,1,Answer
exten => 123,n,AddQueueMember($member)
exten => 123,n,Wait($sometime); long enough for a call to be delivered
exten => 133,n,RemoveQueueMember($member)

I was hoping that someone might have a more elegant solution.

> Dan Austin wrote:
>> I have just setup a small queue implementation for one
>> of my branch offices, replacing a 16 year old key system
>> that had a hacked together pseudo call queuing feature.
>>
>> The 'agents' are not dedicated to the queues and want to
>> be able to logon and get one call only from the queue.
>> I know this is odd, but it is how my users want it to
>> work.
>>
>> I have the login process setup using dynamic agents and
>> set a wrap-up time long enough for the agent to logout.
>> They have accepted this as a short term solution, but
>> they really want to be automatically logged out after
>> taking one and only one call.
>>
>> Any tips or hints on how to accomplish this would be
>> greatly appreciated.
>>
>> Thanks,
>> Dan

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

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


Gmane