[asterisk-users] Asterisk on NGINX Server?

[Dominic Son]

Hi.

Was the AGI Server to write dialplans in any programming language in
Asterisk assumed to be configured for the apache web server?

Or should it not matter what web server you have (in my case NGINX)?


- Dominic

The ability to simplify means to eliminate the unnecessary so that the
necessary may speak.
-Hofstadter's Law

___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

Re: [asterisk-users] Connecting Legacy Pbx With Asterisk With FXS.

[Olivier]

Hi,

So, if you dedicate PBX ports to serve as a trunk, you're likely to loose
the abilty to forward DID calls : when a call for an Asterisk user comes
into Panasonic PBX, it will be forwarded to Panasonic FXS trunk ports.
Then, Asterisk should have no mean to decode to which extension, the call
has to be forwarded, has it comes from an FXO port which won't carry any
data such as CallerID.

I'm not 100% sure of that but that's the way analog ports works here, on
some legacy PBX : analog port means no service.

regards
___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

Re: [asterisk-users] online active call watching

[Dinesh Nair]

On Mon, 10 Sep 2007 13:43:46 -0800, Mojo with Horan & Company, LLC wrote:

> Though still in the proof-of-concept stage, my project "AstSee" from 
> http://www.astsee.com/ might be fun to play with if you're using 
> linux/XWindows.  There are screenshots there.

that may be so, but without source, there's no way we can test it on
freebsd. i'll stick with fop for the timebeing, thank you. 

-- 
Regards,   /\_/\   "All dogs go to heaven."
[EMAIL PROTECTED](0 0)   http://www.openmalaysiablog.com/
+==oOO--(_)--OOo==+
| for a in past present future; do|
|   for b in clients employers associates relatives neighbours pets; do   |
|   echo "The opinions here in no way reflect the opinions of my $a $b."  |
| done; done  |
+=+

___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

Re: [asterisk-users] Connecting Legacy Pbx With Asterisk With FXS.

[Sanspareils Greenlans]

Sir,

I want to dedicate two or three Panasonic port to communicate with Asterisk 
and vise-versa. I am having Panasonic pbx 1232.

Rajeev.

> Hello,
>
> 2007/9/10, C F <[EMAIL PROTECTED]>:
> > Which Panasonic PBX?
> >
> > On 9/10/07, Sanspareils Greenlans <[EMAIL PROTECTED]> wrote:
> > > Sir,
> > >
> > > I am having Asterisk pbx which is running without any problem now i
> > > want
> >
> > to
> >
> > > connect this with Panasonic pbx with FXS port so, if any body want to
> >
> > call
> >
> > > panasonic users than he will call or vise-versa.
>
> How ?
> Do you plan to dedicate Panasonic PBX FXS ports to act as a trunk or would
> dedicate one port for each Asterisk user ?
> ------ next part --
> An HTML attachment was scrubbed...
> URL:
> http://lists.digium.com/pipermail/asterisk-users/attachments/20070910/1b2b2
>6a8/attachment-0001.htm
>
> --
>

___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

Re: [asterisk-users] Asterisk Manager API - Originate command

[Wai Wu]

Just checked. I do have Async set to yes.  

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wai Wu
Sent: Monday, September 10, 2007 7:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk Manager API - Originate command

Thanks for sharing your experience. I will play around with the Asteirsk
server tomorrow again. I took a look at it just before I left the
office. It has loads of crap. It's got all those non-essential things
and X windows running. Also, I can probably be able to get away with
starting a call every 30-50ms. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Riddell
Sent: Monday, September 10, 2007 5:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk Manager API - Originate command

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Wai Wu wrote:
> Hi all,
>  
> Just ran into some issue with the originate AMI command. It seems that

> there is a limit of around 120 calls I can place with the originate 
> command simutanously. By that I mean sending Asterisk a lot of 
> originate command very fast. Anyone know if there is a limitation?
Thnx.

First off, you should be using Async: true

Secondly, you shouldn't really be doing 120 simultaneous calls.

If your server can take say 300 concurrent calls, you will probably need
to start those up with about 30ms between them.

If you really need to start 120 calls all at the identical time, you
probably want to be looking at clustering Asterisk servers.

In SmoothTorque we set a minimum value for delay between calls, then
have a funnel which accepts calls from predictive campaigns.

The funnel knows about the connections to the Asterisk servers, and
distributes the calls in a round robin fashion (assuming all servers are
up).

Each server has a queue which allows calls sent to that server to back
up, and if a queue gets too full calls won't be sent to that server.

If, after stopping the sending of calls to a server, the queue does not
empty out, the server is placed into an inactive state, and a server
marked as standby is moved into the active state.

Any calls which remain in the queue are redistributed to other servers.

Hope that helps!

- --
Kind Regards,

Matt Riddell
Director
___

http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss)
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.7 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iD8DBQFG5bbeDQNt8rg0Kp4RArWFAKCoMPxaDmVLwPD+hupU9T8n+NuFYQCguq8c
T3+G284pc4LV/JMlj13v8gU=
=oaJj
-END PGP SIGNATURE-

___

Sign up now for AstriCon 2007!  September 25-28th.
http://www.astricon.net/ 

--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

___

Sign up now for AstriCon 2007!  September 25-28th.
http://www.astricon.net/ 

--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

___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

Re: [asterisk-users] Asterisk Manager API - Originate command

[Wai Wu]

Thanks for sharing your experience. I will play around with the Asteirsk
server tomorrow again. I took a look at it just before I left the
office. It has loads of crap. It's got all those non-essential things
and X windows running. Also, I can probably be able to get away with
starting a call every 30-50ms. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Riddell
Sent: Monday, September 10, 2007 5:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk Manager API - Originate command

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Wai Wu wrote:
> Hi all,
>  
> Just ran into some issue with the originate AMI command. It seems that

> there is a limit of around 120 calls I can place with the originate 
> command simutanously. By that I mean sending Asterisk a lot of 
> originate command very fast. Anyone know if there is a limitation?
Thnx.

First off, you should be using Async: true

Secondly, you shouldn't really be doing 120 simultaneous calls.

If your server can take say 300 concurrent calls, you will probably need
to start those up with about 30ms between them.

If you really need to start 120 calls all at the identical time, you
probably want to be looking at clustering Asterisk servers.

In SmoothTorque we set a minimum value for delay between calls, then
have a funnel which accepts calls from predictive campaigns.

The funnel knows about the connections to the Asterisk servers, and
distributes the calls in a round robin fashion (assuming all servers are
up).

Each server has a queue which allows calls sent to that server to back
up, and if a queue gets too full calls won't be sent to that server.

If, after stopping the sending of calls to a server, the queue does not
empty out, the server is placed into an inactive state, and a server
marked as standby is moved into the active state.

Any calls which remain in the queue are redistributed to other servers.

Hope that helps!

- --
Kind Regards,

Matt Riddell
Director
___

http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss)
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.7 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iD8DBQFG5bbeDQNt8rg0Kp4RArWFAKCoMPxaDmVLwPD+hupU9T8n+NuFYQCguq8c
T3+G284pc4LV/JMlj13v8gU=
=oaJj
-END PGP SIGNATURE-

___

Sign up now for AstriCon 2007!  September 25-28th.
http://www.astricon.net/ 

--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

___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

[asterisk-users] rtptimeout on Asterisk 1.4.x

[Rodrigo P. Telles]

Hi Folks,

Since I upgraded my asterisk box from 1.2.x to 1.4.x (1.4.10.1 now) I noticed 
some dead calls "apparently" running for
more than 8 hours.
I'm using rtptimeout=60 and rtpholdtimeout=120 and found some log messages like 
this:

chan_sip.c: 'SIP/XXX-085a9308' will not be disconnected in 61 seconds because 
it is directly bridged to another RTP stream

I can kill that calls using 'soft hangup ' but I'd like to know if its 
a new BUG introduced in 1.4.x releases
and if possible, how to fix this?

Thanks in advance.
Rodrigo P. Telles

___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

Re: [asterisk-users] Asterisk Manager API - Originate command

[Wai Wu]

Just to clear things up. It was one TCP connection to the manager
interface and the originate commands are send in a batch. I was able to
get away with 80 calls in a batch. Anything more than that is not good. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Atis
Sent: Monday, September 10, 2007 5:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk Manager API - Originate command

On 9/11/07, Wai Wu <[EMAIL PROTECTED]> wrote:
> Just ran into some issue with the originate AMI command. It seems that

> there is a limit of around 120 calls I can place with the originate 
> command simutanously. By that I mean sending Asterisk a lot of 
> originate command very fast. Anyone know if there is a limitation?
Thnx.

What did you mean by "simultaneously"? Opening 120 manager connections,
and originating call at exactly the same time? I doubt..
So, probably there is some interval - within second/minute, etc.. And
how many manager connections do you use? Maybe asterisk have some limit
of them. Also - i think, there is some limit of asterisk accepting
commands sequentially from one connection.

Btw, what is your CPU load, when creating those 120 calls "instantly"?

Regards,
Atis


--
Atis Lezdins,
IT Responsible of BEST Riga,
[EMAIL PROTECTED]
ICQ: 142239285
Skype: atis.lezdins
Cell Phone: +371 28806004 [Tele2, Latvia] Work phone: +1 800 7502835
[Toll free, USA] ?BEST? -> www.BEST.eu.org

___

Sign up now for AstriCon 2007!  September 25-28th.
http://www.astricon.net/ 

--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

___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

Re: [asterisk-users] online active call watching

[Mojo with Horan & Company, LLC]

Though still in the proof-of-concept stage, my project "AstSee" from 
http://www.astsee.com/ might be fun to play with if you're using 
linux/XWindows.  There are screenshots there.

Mojo


satish patel wrote:
> Dear all
>
>I have asterisk 1.4.11 i am new in asterisk i want 
> to see online call list how it is possible to see how man call 
> currently active is there any command or tool to see online call ?? 
> from --- to
>
>
> Regards
>
>
>
>
> 
> Looking for a deal? Find great prices on flights and hotels 
> 
>  
> with Yahoo! FareChase.
> 
>
> ___
>
> Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 
>
> --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


___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

Re: [asterisk-users] HA - How to detect software failure?

[Matt Riddell]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Yann JOUANIN wrote:
> Hi all,
> 
>  
> 
> I would like to have your opinion about the best way to detect a asterisk
> failure, I mean when asterisk stop working but the process keep existing.

There's a few ways you could do it.

Something like:

asterisk -rx 'iax2 show peers' | wc -l

Would count the number of iax peers (assuming the command didn't return
if asterisk wasn't working).

Or you could connect to the manager interface on port 5038 and issue a
few commands:

http://www.voip-info.org/wiki-Asterisk+manager+API

- --
Kind Regards,

Matt Riddell
Director
___

http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss)
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.7 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iD8DBQFG5bkxDQNt8rg0Kp4RAuTAAJ9fImZF2H1jfXz2uWvWD4xH9Or0vgCgohfc
5U2Sx92QzzACUOvawsE4eJw=
=sGKd
-END PGP SIGNATURE-

___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

[asterisk-users] DTMF

[Ira]

Hi

Ever since I upgraded to the most recent V1.2 * and Zaptel DTMF 
stopped working. If I call my cell and press a key, I can hear that 
it's trying to send a tone, but there's not enough to trigger the 
menus at the places I call.  I can't see that this is user adjustable 
and it use to work just fine.  Any suggestions on how to fix or 
troubleshoot this.  I did recently install * and Zaptel 1.4 and then 
go back to 1.2 if that matters.

Thanks ever so much,  Ira


___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

Re: [asterisk-users] Which cause less CPU usage: GSM or wav??

[Barton Fisher]

Thanks, again. That did the trick!

Bart

Matt Riddell wrote:

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Barton Fisher wrote:
  

Thanks, OK, a bit confused  The cards are TE410P.  I really don't
see how the set a codec for this, other than it might default to
something in code like ulaw.  Any clue on how to verify codec in use
during a call?



Basically its going to be g711.ulaw for T1 (USA) and g711.alaw for E1
(rest of world) 99.9% of the time.

Unless you have something strange or different, I'd record in ulaw for T1.

- --
Kind Regards,

Matt Riddell
Director
___

http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss)
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.7 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iD8DBQFG5MI9DQNt8rg0Kp4RAgRWAKCL2l8egvLV2Xu3T754KJMzGXrKnQCfboCx
aFwrtGNKZ0EbZr176MDZUkY=
=HvDo
-END PGP SIGNATURE-



__ NOD32 2517 (20070910) Information __

This message was checked by NOD32 antivirus system.
http://www.eset.com



  


--

Barton Fisher
Innovative Communications
714-228-5400 Ext 5410
http://www.icpage.com

begin:vcard
fn:Barton Fisher
n:Fisher;Barton
org:Innovative Communications
adr:;;7439 La Palma Ave # 255;Buena Park;CA;90620;USA
email;internet:[EMAIL PROTECTED]
tel;work:714-228-5410
url:http://icpage.com
version:2.1
end:vcard

___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

Re: [asterisk-users] Asterisk Manager API - Originate command

[Matt Riddell]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Wai Wu wrote:
> Hi all, 
>  
> Just ran into some issue with the originate AMI command. It seems that
> there is a limit of around 120 calls I can place with the originate
> command simutanously. By that I mean sending Asterisk a lot of originate
> command very fast. Anyone know if there is a limitation? Thnx.

First off, you should be using Async: true

Secondly, you shouldn't really be doing 120 simultaneous calls.

If your server can take say 300 concurrent calls, you will probably need
to start those up with about 30ms between them.

If you really need to start 120 calls all at the identical time, you
probably want to be looking at clustering Asterisk servers.

In SmoothTorque we set a minimum value for delay between calls, then
have a funnel which accepts calls from predictive campaigns.

The funnel knows about the connections to the Asterisk servers, and
distributes the calls in a round robin fashion (assuming all servers are
up).

Each server has a queue which allows calls sent to that server to back
up, and if a queue gets too full calls won't be sent to that server.

If, after stopping the sending of calls to a server, the queue does not
empty out, the server is placed into an inactive state, and a server
marked as standby is moved into the active state.

Any calls which remain in the queue are redistributed to other servers.

Hope that helps!

- --
Kind Regards,

Matt Riddell
Director
___

http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss)
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.7 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iD8DBQFG5bbeDQNt8rg0Kp4RArWFAKCoMPxaDmVLwPD+hupU9T8n+NuFYQCguq8c
T3+G284pc4LV/JMlj13v8gU=
=oaJj
-END PGP SIGNATURE-

___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

Re: [asterisk-users] Asterisk Manager API - Originate command

[Atis]

On 9/11/07, Wai Wu <[EMAIL PROTECTED]> wrote:
> Just ran into some issue with the originate AMI command. It seems that there
> is a limit of around 120 calls I can place with the originate command
> simutanously. By that I mean sending Asterisk a lot of originate command
> very fast. Anyone know if there is a limitation? Thnx.

What did you mean by "simultaneously"? Opening 120 manager
connections, and originating call at exactly the same time? I doubt..
So, probably there is some interval - within second/minute, etc.. And
how many manager connections do you use? Maybe asterisk have some
limit of them. Also - i think, there is some limit of asterisk
accepting commands sequentially from one connection.

Btw, what is your CPU load, when creating those 120 calls "instantly"?

Regards,
Atis


-- 
Atis Lezdins,
IT Responsible of BEST Riga,
[EMAIL PROTECTED]
ICQ: 142239285
Skype: atis.lezdins
Cell Phone: +371 28806004 [Tele2, Latvia]
Work phone: +1 800 7502835 [Toll free, USA]
?BEST? -> www.BEST.eu.org

___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

Re: [asterisk-users] Partitioning DSL input

[Ira]

At 02:11 PM 9/10/2007, you wrote:

>Can people on this list share their experiences on how they 
>partition a DSL for small business internet service with a router so 
>that a portion is dedicated to VOIP and another portion to 
>computers.  Of course, the idea is to do this with a low cost router 
>(under $100).


dd-wrt or Sveasoft on a Linksys router though I understand there are 
better choices in routers today.

Ira 


___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

[asterisk-users] Asterisk Manager API - Originate command

[Wai Wu]

Hi all, 
 
Just ran into some issue with the originate AMI command. It seems that
there is a limit of around 120 calls I can place with the originate
command simutanously. By that I mean sending Asterisk a lot of originate
command very fast. Anyone know if there is a limitation? Thnx.
___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

Re: [asterisk-users] E1 Line Tapping

[Andrew Latham]

I think they mean the Rhino Dax...  http://rhinoequipment.com/minidax.html

On 9/10/07, Steve Totaro <[EMAIL PROTECTED]> wrote:
> http://www.voipsupply.com/manufacturers/RedFone_Communications.html?gclid=CKmd5OrbuY4CFVB1OAodfC7PxQ
>
>
> Ricardo Gemignani wrote:
> > Thanks Steve,
> >
> >   If somebody knows about this hardware, or already used it. Please give
> > me some help.
> >
> > TIA,
> > Ricardo
> >
> > On 9/10/07, *Steve Totaro* <[EMAIL PROTECTED]
> > > wrote:
> >
> > You could buy two identical servers and use the device (name escapes me)
> > that will detect one server going down and flip the ISDN traffic to the
> > spare.
> >
> > Or you could just buy a really good server with redundant power
> > supplies, raid 5, and hope for the best.
> >
> > Thanks,
> > Steve
> >
> > Ricardo Gemignani wrote:
> >  > Thanks for answering guys!
> >  >
> >  >   Ok, let me see if i understood.
> >  >
> >  >   If I use the line tapping strategy I wont be able to use
> > asterisk to
> >  > do the recordings. Correct?
> >  >
> >  >   So, i need to use the asterisk as the Man in the Middle ( I think
> >  > that's the same as the "back to back" suggestion from Tzafrir,
> > Isn't it?
> >  > ). Ok, so every call will pass through Asterisk and I can do
> > anything i
> >  > want with it. Thats cool, but since all the calls pass through my
> >  > recording box I've just created another fail point. And if
> > someday my
> >  > recording box stop responding? Is there someway to minimize that?
> >  >
> >  > TIA,
> >  > Ricardo
> >  >
> >  > On 9/5/07, *Andrew Latham* < [EMAIL PROTECTED]
> > 
> >  > >> wrote:
> >  >
> >  > or a man in the middle...
> >  >
> >  > http://www.tuxtone.com/index.php/VoIP:T1_man_in_the_middle
> >  > 
> >  >
> >  >
> >  >
> >  > On 9/5/07, Steve Totaro < [EMAIL PROTECTED]
> > 
> >  >  > >> wrote:
> >  >  > Ricardo Gemignani wrote:
> >  >  > > Hi all,
> >  >  > >
> >  >  > >   My name is Ricardo and unfortunately I'm just crawling
> > in this
> >  >  > > telecomm/asterisk world. So, after reading all day long
> > i still
> >  > don't
> >  >  > > understand a few things. :D
> >  >  > >
> >  >  > >   I'm trying to "develop" a call recorder for a
> > costumer. He has a
> >  >  > > small call center ( 10 agents ) and want to record all
> > calls.
> >  > Since he
> >  >  > > already has everything (ACD only) working perfectly in
> > the PBX and
> >  >  > > don't want me to "touch" it, I need do develop a  less
> > intrusive as
> >  >  > > possible system.
> >  >  > >
> >  >  > >   I was thinking to do a line tapping in his E1 branch
> > before it
> >  >  > > reaches the PBX and record it using Asterisk, then develop a
> >  > small web
> >  >  > > interface to recover the recordings.
> >  >  > >
> >  >  > >   In my research about E1 line tapping I found this
> > product from
> >  >  > > Sangoma ( http://www.sangoma.com/datasheets/tapping )
> > but could not
> >  >  > > understand exactly how it really works.
> >  >  > >
> >  >  > >   Does anybody already used it?
> >  >  > >
> >  >  > >   Is it possible to use it with Asterisk?
> >  >  > >
> >  >  > > tia,
> >  >  > > Ricardo Gemignani
> >  >  > >
> >  >  >
> >  >  > Check out OrecX but you should be able to record that
> > volume of
> >  > calls
> >  >  > natively on the box (that is assuming you are using
> > Asterisk as your
> >  >  > call center system.
> >  >  >
> >  >  > Thanks,
> >  >  > Steve
> >  >  >
> >
>
> ___
>
> Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/
>
> --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
>


-- 
/*
 Andrew Latham
 LATHAMA (lay-th-ham-eh)
 [EMAIL PROTECTED]
 [EMAIL PROTECTED]
*/

___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

Re: [asterisk-users] E1 Line Tapping

[Steve Totaro]

http://www.voipsupply.com/manufacturers/RedFone_Communications.html?gclid=CKmd5OrbuY4CFVB1OAodfC7PxQ


Ricardo Gemignani wrote:
> Thanks Steve,
> 
>   If somebody knows about this hardware, or already used it. Please give 
> me some help.
> 
> TIA,
> Ricardo
> 
> On 9/10/07, *Steve Totaro* <[EMAIL PROTECTED] 
> > wrote:
> 
> You could buy two identical servers and use the device (name escapes me)
> that will detect one server going down and flip the ISDN traffic to the
> spare.
> 
> Or you could just buy a really good server with redundant power
> supplies, raid 5, and hope for the best.
> 
> Thanks,
> Steve
> 
> Ricardo Gemignani wrote:
>  > Thanks for answering guys!
>  >
>  >   Ok, let me see if i understood.
>  >
>  >   If I use the line tapping strategy I wont be able to use
> asterisk to
>  > do the recordings. Correct?
>  >
>  >   So, i need to use the asterisk as the Man in the Middle ( I think
>  > that's the same as the "back to back" suggestion from Tzafrir,
> Isn't it?
>  > ). Ok, so every call will pass through Asterisk and I can do
> anything i
>  > want with it. Thats cool, but since all the calls pass through my
>  > recording box I've just created another fail point. And if
> someday my
>  > recording box stop responding? Is there someway to minimize that?
>  >
>  > TIA,
>  > Ricardo
>  >
>  > On 9/5/07, *Andrew Latham* < [EMAIL PROTECTED]
> 
>  > >> wrote:
>  >
>  > or a man in the middle...
>  >
>  > http://www.tuxtone.com/index.php/VoIP:T1_man_in_the_middle
>  > 
>  >
>  >
>  >
>  > On 9/5/07, Steve Totaro < [EMAIL PROTECTED]
> 
>  >  >> wrote:
>  >  > Ricardo Gemignani wrote:
>  >  > > Hi all,
>  >  > >
>  >  > >   My name is Ricardo and unfortunately I'm just crawling
> in this
>  >  > > telecomm/asterisk world. So, after reading all day long
> i still
>  > don't
>  >  > > understand a few things. :D
>  >  > >
>  >  > >   I'm trying to "develop" a call recorder for a
> costumer. He has a
>  >  > > small call center ( 10 agents ) and want to record all
> calls.
>  > Since he
>  >  > > already has everything (ACD only) working perfectly in
> the PBX and
>  >  > > don't want me to "touch" it, I need do develop a  less
> intrusive as
>  >  > > possible system.
>  >  > >
>  >  > >   I was thinking to do a line tapping in his E1 branch
> before it
>  >  > > reaches the PBX and record it using Asterisk, then develop a
>  > small web
>  >  > > interface to recover the recordings.
>  >  > >
>  >  > >   In my research about E1 line tapping I found this
> product from
>  >  > > Sangoma ( http://www.sangoma.com/datasheets/tapping )
> but could not
>  >  > > understand exactly how it really works.
>  >  > >
>  >  > >   Does anybody already used it?
>  >  > >
>  >  > >   Is it possible to use it with Asterisk?
>  >  > >
>  >  > > tia,
>  >  > > Ricardo Gemignani
>  >  > >
>  >  >
>  >  > Check out OrecX but you should be able to record that
> volume of
>  > calls
>  >  > natively on the box (that is assuming you are using
> Asterisk as your
>  >  > call center system.
>  >  >
>  >  > Thanks,
>  >  > Steve
>  >  >
> 

___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

Re: [asterisk-users] Cisco UC 500

[Drew Gibson]

Jeremy Mann wrote:


Is the Cisco UC 500 able to integrate with Asterisk?  Specifically 
does it work via SIP?  Just curious, as the Cold Call Cisco sales rep 
had no clue what SIP even was, and this device looks interesting.



Google "cisco UC500", hit #2 = 
http://www.cisco.com/en/US/products/ps7293/products_data_sheet0900aecd8061fb06.html


Quotes: 

"Core components of the Cisco Unified Communications 500 Series 
include:Cisco Unified IP phones, including wireless handsets and 
Session Initiation Protocol (SIP) phones"


"PSTN interfaces and features:  SIP trunks and RFC 2833 support"

Does that help?

I'll bet Asterisk is cheaper though. :-)

regards,

Drew

--
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com

___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

Re: [asterisk-users] E1 Line Tapping

[Ricardo Gemignani]

Thanks Steve,

  If somebody knows about this hardware, or already used it. Please give me
some help.

TIA,
Ricardo

On 9/10/07, Steve Totaro <[EMAIL PROTECTED]> wrote:
>
> You could buy two identical servers and use the device (name escapes me)
> that will detect one server going down and flip the ISDN traffic to the
> spare.
>
> Or you could just buy a really good server with redundant power
> supplies, raid 5, and hope for the best.
>
> Thanks,
> Steve
>
> Ricardo Gemignani wrote:
> > Thanks for answering guys!
> >
> >   Ok, let me see if i understood.
> >
> >   If I use the line tapping strategy I wont be able to use asterisk to
> > do the recordings. Correct?
> >
> >   So, i need to use the asterisk as the Man in the Middle ( I think
> > that's the same as the "back to back" suggestion from Tzafrir, Isn't it?
> > ). Ok, so every call will pass through Asterisk and I can do anything i
> > want with it. Thats cool, but since all the calls pass through my
> > recording box I've just created another fail point. And if someday my
> > recording box stop responding? Is there someway to minimize that?
> >
> > TIA,
> > Ricardo
> >
> > On 9/5/07, *Andrew Latham* < [EMAIL PROTECTED]
> > > wrote:
> >
> > or a man in the middle...
> >
> > http://www.tuxtone.com/index.php/VoIP:T1_man_in_the_middle
> > 
> >
> >
> >
> > On 9/5/07, Steve Totaro < [EMAIL PROTECTED]
> > > wrote:
> >  > Ricardo Gemignani wrote:
> >  > > Hi all,
> >  > >
> >  > >   My name is Ricardo and unfortunately I'm just crawling in
> this
> >  > > telecomm/asterisk world. So, after reading all day long i still
> > don't
> >  > > understand a few things. :D
> >  > >
> >  > >   I'm trying to "develop" a call recorder for a costumer. He
> has a
> >  > > small call center ( 10 agents ) and want to record all calls.
> > Since he
> >  > > already has everything (ACD only) working perfectly in the PBX
> and
> >  > > don't want me to "touch" it, I need do develop a  less
> intrusive as
> >  > > possible system.
> >  > >
> >  > >   I was thinking to do a line tapping in his E1 branch before
> it
> >  > > reaches the PBX and record it using Asterisk, then develop a
> > small web
> >  > > interface to recover the recordings.
> >  > >
> >  > >   In my research about E1 line tapping I found this product
> from
> >  > > Sangoma ( http://www.sangoma.com/datasheets/tapping ) but could
> not
> >  > > understand exactly how it really works.
> >  > >
> >  > >   Does anybody already used it?
> >  > >
> >  > >   Is it possible to use it with Asterisk?
> >  > >
> >  > > tia,
> >  > > Ricardo Gemignani
> >  > >
> >  >
> >  > Check out OrecX but you should be able to record that volume of
> > calls
> >  > natively on the box (that is assuming you are using Asterisk as
> your
> >  > call center system.
> >  >
> >  > Thanks,
> >  > Steve
> >  >
> >  > ___
> >  > --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
> >  >
> >
> >
> > --
> > /*
> > Andrew Latham
> > LATHAMA (lay-th-ham-eh)
> > [EMAIL PROTECTED] 
> > [EMAIL PROTECTED] 
> > */
> >
> > ___
> >
> > Sign up now for AstriCon 2007!  September 25-28th.
> > http://www.astricon.net/
> >
> > --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
> >
> >
> >
> >
> > --
> > haoole
> > "alea jacta est"
> >
> >
> > 
> >
> > ___
> >
> > Sign up now for AstriCon 2007!  September 25-28th.
> http://www.astricon.net/
> >
> > --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
>
>
> ___
>
> Sign up now for AstriCon 2007!  September 25-28th.
> http://www.astricon.net/
>
> --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
>



-- 
haoole
"alea jacta est"
___

Sign up now for AstriCon 2007!  Se

Re: [asterisk-users] Partitioning DSL input

[David Gomillion]

On 9/10/07, Ira <[EMAIL PROTECTED]> wrote:
>
> At 02:11 PM 9/10/2007, you wrote:
>
> >Can people on this list share their experiences on how they
> >partition a DSL for small business internet service with a router so
> >that a portion is dedicated to VOIP and another portion to
> >computers.  Of course, the idea is to do this with a low cost router
> >(under $100).
>
>
> dd-wrt or Sveasoft on a Linksys router though I understand there are
> better choices in routers today.


Don't expect too much out of traffic shaping. While it should work nearly
perfectly upstream, there's only so much you can do to control the
downstream (from your ISP to you).
___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

[asterisk-users] Cisco UC 500

[Jeremy Mann]

Is the Cisco UC 500 able to integrate with Asterisk?  Specifically does it work 
via SIP?  Just curious, as the Cold Call Cisco sales rep had no clue what SIP 
even was, and this device looks interesting.


This e-mail, facsimile, or letter and any files or attachments transmitted with 
it contains information that is confidential and privileged. This information 
is intended only for the use of the individual(s) and entity(ies) to whom it is 
addressed. If you are the intended recipient, further disclosures are 
prohibited without proper authorization. If you are not the intended recipient, 
any disclosure, copying, printing, or use of this information is strictly 
prohibited and possibly a violation of federal or state law and regulations. If 
you have received this information in error, please notify Texas Health 
Management Group immediately at 1-817-310-4999. Texas Health Management Group, 
its subsidiaries, and affiliates hereby claim all applicable privileges related 
to this information.

-- 
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.

___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

Re: [asterisk-users] E1 Line Tapping

[Steve Totaro]

You could buy two identical servers and use the device (name escapes me) 
that will detect one server going down and flip the ISDN traffic to the 
spare.

Or you could just buy a really good server with redundant power 
supplies, raid 5, and hope for the best.

Thanks,
Steve

Ricardo Gemignani wrote:
> Thanks for answering guys!
> 
>   Ok, let me see if i understood.
> 
>   If I use the line tapping strategy I wont be able to use asterisk to 
> do the recordings. Correct?
> 
>   So, i need to use the asterisk as the Man in the Middle ( I think 
> that's the same as the "back to back" suggestion from Tzafrir, Isn't it? 
> ). Ok, so every call will pass through Asterisk and I can do anything i 
> want with it. Thats cool, but since all the calls pass through my 
> recording box I've just created another fail point. And if someday my 
> recording box stop responding? Is there someway to minimize that?
> 
> TIA,
> Ricardo
> 
> On 9/5/07, *Andrew Latham* < [EMAIL PROTECTED] 
> > wrote:
> 
> or a man in the middle...
> 
> http://www.tuxtone.com/index.php/VoIP:T1_man_in_the_middle
> 
> 
> 
> 
> On 9/5/07, Steve Totaro < [EMAIL PROTECTED]
> > wrote:
>  > Ricardo Gemignani wrote:
>  > > Hi all,
>  > >
>  > >   My name is Ricardo and unfortunately I'm just crawling in this
>  > > telecomm/asterisk world. So, after reading all day long i still
> don't
>  > > understand a few things. :D
>  > >
>  > >   I'm trying to "develop" a call recorder for a costumer. He has a
>  > > small call center ( 10 agents ) and want to record all calls.
> Since he
>  > > already has everything (ACD only) working perfectly in the PBX and
>  > > don't want me to "touch" it, I need do develop a  less intrusive as
>  > > possible system.
>  > >
>  > >   I was thinking to do a line tapping in his E1 branch before it
>  > > reaches the PBX and record it using Asterisk, then develop a
> small web
>  > > interface to recover the recordings.
>  > >
>  > >   In my research about E1 line tapping I found this product from
>  > > Sangoma ( http://www.sangoma.com/datasheets/tapping ) but could not
>  > > understand exactly how it really works.
>  > >
>  > >   Does anybody already used it?
>  > >
>  > >   Is it possible to use it with Asterisk?
>  > >
>  > > tia,
>  > > Ricardo Gemignani
>  > >
>  >
>  > Check out OrecX but you should be able to record that volume of
> calls
>  > natively on the box (that is assuming you are using Asterisk as your
>  > call center system.
>  >
>  > Thanks,
>  > Steve
>  >
>  > ___
>  > --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
>  >
> 
> 
> --
> /*
> Andrew Latham
> LATHAMA (lay-th-ham-eh)
> [EMAIL PROTECTED] 
> [EMAIL PROTECTED] 
> */
> 
> ___
> 
> Sign up now for AstriCon 2007!  September 25-28th.  
> http://www.astricon.net/
> 
> --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
> 
> 
> 
> 
> -- 
> haoole
> "alea jacta est"
> 
> 
> 
> 
> ___
> 
> Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 
> 
> --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


___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

Re: [asterisk-users] Siemans SIP/PSTN phone S450

[Gordon Henderson]

On Mon, 10 Sep 2007, Adrian Marsh wrote:

> Hi All,
>
> Just added a Siemens DECT SIP/PSTN S450 phone to login to my A*k server,
> and I see "Got SIP response 405 "Method Not Allowed" back from
> 192.168.3.64" but the phone seems to work ok.
>
> Any ideas where it falls over in the SIP protocol?  I've included this
> in the debug below.

I have several Siemens C460IP's on various servers... They all do the same 
thing too. Doesn't seem to have any adverse effect though.

Gordon

___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

Re: [asterisk-users] E1 Line Tapping

[Ricardo Gemignani]

Thanks for answering guys!

  Ok, let me see if i understood.

  If I use the line tapping strategy I wont be able to use asterisk to do
the recordings. Correct?

  So, i need to use the asterisk as the Man in the Middle ( I think that's
the same as the "back to back" suggestion from Tzafrir, Isn't it? ). Ok, so
every call will pass through Asterisk and I can do anything i want with it.
Thats cool, but since all the calls pass through my recording box I've just
created another fail point. And if someday my recording box stop responding?
Is there someway to minimize that?

TIA,
Ricardo

On 9/5/07, Andrew Latham <[EMAIL PROTECTED]> wrote:
>
> or a man in the middle...
>
> http://www.tuxtone.com/index.php/VoIP:T1_man_in_the_middle
>
>
>
> On 9/5/07, Steve Totaro < [EMAIL PROTECTED]> wrote:
> > Ricardo Gemignani wrote:
> > > Hi all,
> > >
> > >   My name is Ricardo and unfortunately I'm just crawling in this
> > > telecomm/asterisk world. So, after reading all day long i still don't
> > > understand a few things. :D
> > >
> > >   I'm trying to "develop" a call recorder for a costumer. He has a
> > > small call center ( 10 agents ) and want to record all calls. Since he
>
> > > already has everything (ACD only) working perfectly in the PBX and
> > > don't want me to "touch" it, I need do develop a  less intrusive as
> > > possible system.
> > >
> > >   I was thinking to do a line tapping in his E1 branch before it
> > > reaches the PBX and record it using Asterisk, then develop a small web
> > > interface to recover the recordings.
> > >
> > >   In my research about E1 line tapping I found this product from
> > > Sangoma ( http://www.sangoma.com/datasheets/tapping ) but could not
> > > understand exactly how it really works.
> > >
> > >   Does anybody already used it?
> > >
> > >   Is it possible to use it with Asterisk?
> > >
> > > tia,
> > > Ricardo Gemignani
> > >
> >
> > Check out OrecX but you should be able to record that volume of calls
> > natively on the box (that is assuming you are using Asterisk as your
> > call center system.
> >
> > Thanks,
> > Steve
> >
> > ___
> > --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
> >
>
>
> --
> /*
> Andrew Latham
> LATHAMA (lay-th-ham-eh)
> [EMAIL PROTECTED]
>  [EMAIL PROTECTED]
> */
>
> ___
>
> Sign up now for AstriCon 2007!  September 25-28th.
> http://www.astricon.net/
>
> --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
>



-- 
haoole
"alea jacta est"
___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

Re: [asterisk-users] Partitioning DSL input

[C. Savinovich]

 Looks good. a lot of initial work, but looks worth the effort.  Do you find
that it improves the quality of your VOIP calls?

 

C. Savinovich

 

 

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex Robar
Sent: Monday, September 10, 2007 11:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [PHISH] Re: [asterisk-users] Partitioning DSL input

 

pfSense works very well for this. You can use it to setup VLANs (one for
your PCs, the other for your VoIP equipment), and it has a traffic
shaping/queuing mechanism for prioritizing VoIP.

AR

On 9/10/07, C. Savinovich <[EMAIL PROTECTED]> wrote:

Can people on this list share their experiences on how they partition a DSL
for small business internet service with a router so that a portion is
dedicated to VOIP and another portion to computers.  Of course, the idea is
to do this with a low cost router (under $100).

 

Many Thanks

C. Savinovich

 


___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/
 

--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
 




-- 
Alex Robar
[EMAIL PROTECTED] 

___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

Re: [asterisk-users] Partitioning DSL input

[Steve Totaro]

C. Savinovich wrote:
> Can people on this list share their experiences on how they partition a 
> DSL for small business internet service with a router so that a portion 
> is dedicated to VOIP and another portion to computers.  Of course, the 
> idea is to do this with a low cost router (under $100).
> 
>  
> 
> Many Thanks
> 
> C. Savinovich
> 
>  
> 

Check the recent archives.  Someone announced that they had a Beta 
package for the WRT54G (and possibly other 3rd party compatible firmware 
routers) that would achieve exactly that.

Beyond that, checkout 3rd party firmwares that run on these routers, 
some have QoS and traffic shaping abilities.

Thanks,
Steve

___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

Re: [asterisk-users] Asterisk on Ubuntu Feisty

[Ron Wellsted]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Christian wrote:
> Hello,
> 
> 
> On 2007-09-09 at 22:36 Ron Wellsted wrote:
> 
> Christian wrote:
 Hi,
 What parameter should I use to that command?


 On 2007-09-09 at 13:45 Ron Wellsted wrote:

 Tzafrir Cohen wrote:
>>> On Sun, Sep 09, 2007 at 02:32:14AM +0200, Christian wrote:
 Hi all,
 Have just installed v1.4.11 of Asterisk, but I am trying to have it 
 start at boot but with no luck.
 I have used the make config command but it doesn't start. Any help 
 would be apreciated, many thanks!
>>> use the command update-rc.d
>>>
>>> Also, as always in the case of software that has already been
> packaged,
>>> it may help to look at the existing package.
>>>
 I used "update-rc.d asterisk 30" to ensure that it started after zaptel
 and mysql (which by default start at 20).


> Sorry, it should have read "sudo update-rc.d asterisk defaults 30"
>> Many thanks, will try that.
>> Is Zaptel already loaded or will I need to do another command for that?
>> Still learning.
>> Many thanks,
>> Christian
Zaptel will need to be loaded if needed for hardware and/or timing.

In the Zaptel source directory, there is zaptel.init, modify this for
your /etc/init.d/zaptel file


- --
Ron Wellsted
[EMAIL PROTECTED] http://www.wellsted.org.uk
N 52.567623, W 2.136111 Linux Counter No. 202120
Ekiga: 645022
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.6 (GNU/Linux)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iQEVAwUBRuWM3ktP/KMNOfRbAQJpSwf9HmC6lP/DI7/ych2CKKQz6Nk2/S8pJoy+
2btD75+tyhImepe03KOeQyWuWu0HLhJW7pakrIFov3Ey7gwX4rqj+z9sr1r/goA4
JGkYJZN4cRFeZZpgN0YTXAbQpWSaXKwxjXJI6i3va3vEk9h1csXzFlKPZQHDtBI8
3ByJveuAmifBC6+5DBhO2nSFBAZhhPZjfN02ggLoZMW6hCIJH52iL+tw/2NVNY8e
Wqw/Fe/FyAt6H10N4Zud6WtRdSQdJ0sF5ZMJ4qCgPG683oUgsYF+sdTSt00KMwAF
OX73EKPyHmrBwIOmn8w/cupLN7Hx7MDhXC2dTpqGvuVVArAWcL9swA==
=ZkRf
-END PGP SIGNATURE-


___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

Re: [asterisk-users] Partitioning DSL input

[Alex Robar]

pfSense works very well for this. You can use it to setup VLANs (one for
your PCs, the other for your VoIP equipment), and it has a traffic
shaping/queuing mechanism for prioritizing VoIP.

AR

On 9/10/07, C. Savinovich <[EMAIL PROTECTED]> wrote:
>
>  Can people on this list share their experiences on how they partition a
> DSL for small business internet service with a router so that a portion is
> dedicated to VOIP and another portion to computers.  Of course, the idea is
> to do this with a low cost router (under $100).
>
>
>
> Many Thanks
>
> C. Savinovich
>
>
>
> ___
>
> Sign up now for AstriCon 2007!  September 25-28th.
> http://www.astricon.net/
>
> --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
>



-- 
Alex Robar
[EMAIL PROTECTED]
___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

[asterisk-users] Partitioning DSL input

[C. Savinovich]

Can people on this list share their experiences on how they partition a DSL
for small business internet service with a router so that a portion is
dedicated to VOIP and another portion to computers.  Of course, the idea is
to do this with a low cost router (under $100).

 

Many Thanks

C. Savinovich

 

___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

[asterisk-users] Siemans SIP/PSTN phone S450

[Adrian Marsh]

Hi All,

Just added a Siemens DECT SIP/PSTN S450 phone to login to my A*k server,
and I see "Got SIP response 405 "Method Not Allowed" back from
192.168.3.64" but the phone seems to work ok.

Any ideas where it falls over in the SIP protocol?  I've included this
in the debug below.



ubiphone*CLI>
<-- SIP read from 192.168.3.64:5060:

--- (0 headers 0 lines) Nat keepalive ---
ubiphone*CLI>
<-- SIP read from 192.168.3.64:5060:

--- (0 headers 0 lines) Nat keepalive ---
-- Got SIP response 489 "Bad event" back from 192.168.3.10
ubiphone*CLI>
<-- SIP read from 192.168.3.64:5060:

--- (0 headers 0 lines) Nat keepalive ---
12 headers, 0 lines
Reliably Transmitting (NAT) to 192.168.3.64:5060:
OPTIONS sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.6:5060;branch=z9hG4bK4a6d0d34;rport
From: "asterisk" ;tag=as35c7a074
To: 
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 10 Sep 2007 17:23:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
ubiphone*CLI>
<-- SIP read from 192.168.3.64:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.3.6:5060;branch=z9hG4bK4a6d0d34;rport=5060
From: "asterisk" ;tag=as35c7a074
To: ;tag=1624959632
Call-ID: [EMAIL PROTECTED]
CSeq: 102 OPTIONS
Contact: "Adrian Marsh" 
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO
Accept: application/sdp,application/dtmf-relay
Accept-Encoding: identity
Accept-Language: en
Content-Length: 0


--- (12 headers 0 lines) ---
Destroying call '[EMAIL PROTECTED]'
ubiphone*CLI>
<-- SIP read from 192.168.3.64:5060:
REGISTER sip:some.server.com SIP/2.0
Via: SIP/2.0/UDP
192.168.3.64:5060;branch=z9hG4bK51fe39cf13e93bc714bfe8ea31b6b958;rport
From: Adrian Marsh ;tag=3054246604
To: Adrian Marsh 
Call-ID: [EMAIL PROTECTED]
CSeq: 291 REGISTER
Contact: "Adrian Marsh" 
Max-Forwards: 70
User-Agent: S450 IP0207
Expires: 180
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO
Content-Length: 0


--- (12 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.3.64 : 5060 (NAT)
Transmitting (NAT) to 192.168.3.64:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.3.64:5060;branch=z9hG4bK51fe39cf13e93bc714bfe8ea31b6b958;receive
d=192.168.3.64;rport=5060
From: Adrian Marsh ;tag=3054246604
To: Adrian Marsh 
Call-ID: [EMAIL PROTECTED]
CSeq: 291 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: 
Content-Length: 0


---
Transmitting (NAT) to 192.168.3.64:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
192.168.3.64:5060;branch=z9hG4bK51fe39cf13e93bc714bfe8ea31b6b958;receive
d=192.168.3.64;rport=5060
From: Adrian Marsh ;tag=3054246604
To: Adrian Marsh ;tag=as5908b79f
Call-ID: [EMAIL PROTECTED]
CSeq: 291 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="3960830f"
Content-Length: 0


---
Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms
ubiphone*CLI>
<-- SIP read from 192.168.3.64:5060:
REGISTER sip:some.server.com SIP/2.0
Via: SIP/2.0/UDP
192.168.3.64:5060;branch=z9hG4bKca6e25fd9fe65366c967bc15f17a7b1;rport
From: Adrian Marsh ;tag=3054246604
To: Adrian Marsh 
Call-ID: [EMAIL PROTECTED]
CSeq: 292 REGISTER
Contact: "Adrian Marsh" 
Authorization: Digest username="6627", realm="asterisk", algorithm=MD5,
uri="sip:some.server.com", nonce="3960830f",
response="7e032e9766f943e9f60f7d1f46114dee"
Max-Forwards: 70
User-Agent: S450 IP0207
Expires: 180
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO
Content-Length: 0


--- (13 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.3.64 : 5060 (NAT)
Transmitting (NAT) to 192.168.3.64:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.3.64:5060;branch=z9hG4bKca6e25fd9fe65366c967bc15f17a7b1;received
=192.168.3.64;rport=5060
From: Adrian Marsh ;tag=3054246604
To: Adrian Marsh 
Call-ID: [EMAIL PROTECTED]
CSeq: 292 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: 
Content-Length: 0


---
Transmitting (NAT) to 192.168.3.64:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.3.64:5060;branch=z9hG4bKca6e25fd9fe65366c967bc15f17a7b1;received
=192.168.3.64;rport=5060
From: Adrian Marsh ;tag=3054246604
To: Adrian Marsh ;tag=as5908b79f
Call-ID: [EMAIL PROTECTED]
CSeq: 292 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Expires: 180
Contact: ;expires=180
Date: Mon, 10 Sep 2007 17:23:10 GMT
Content-Length: 0


---
Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms
12 headers, 3 lines
Reliably Transmitting (NAT) to 192.168.3.64:5060:
NOTIFY sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.6:5060;branch=z9hG4bK2fa265b3;rport
From: "asterisk" ;tag=as539ed18c
To: 
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: me

Re: [asterisk-users] Failover SIP logic

[Jeremy Mann]

Asterisk 1.4.11

Sorry, meant to include that

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrea Spadaccini
Sent: Monday, September 10, 2007 10:59 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Failover SIP logic

Ciao Jeremy,

> I need some extensions logic assistance, I'm trying to dial out one of
> multiple SIP trunks, in sequence.  I need to detect a busy SIP trunk(I only
> allow 1 call per trunk) and roll over to a second or third depending on that
> busy status
>
> Here's what I've got for a macro thusfar, but it's not working(fails if the
> 1st trunk is busy) extensions.conf:
>
> [globals]
> trunk_1 => SIP/trunk1
> trunk_2 => SIP/trunk2
> trunk_3 => SIP/trunk3
>
> [macro-trunkdial]
> exten => s,1,Dial(${trunk_1}/${ARG1})
> exten => s,2,Hangup()
> exten => s,102,Dial(${trunk_2}/${ARG1})
> exten => s,103,Hangup()
> exten => s,203,Dial(${trunk_3}/${ARG1})
> exten => s,204,Hangup()


Which asterisk version are you using?
IIRC, priority jumping (ie. going to n+101) was disabled by default in some
1.2.x version. You should rely on DIALSTATUS. See Dial() page in voip-info.org.

HTH,


This e-mail, facsimile, or letter and any files or attachments transmitted with 
it contains information that is confidential and privileged. This information 
is intended only for the use of the individual(s) and entity(ies) to whom it is 
addressed. If you are the intended recipient, further disclosures are 
prohibited without proper authorization. If you are not the intended recipient, 
any disclosure, copying, printing, or use of this information is strictly 
prohibited and possibly a violation of federal or state law and regulations. If 
you have received this information in error, please notify Texas Health 
Management Group immediately at 1-817-310-4999. Texas Health Management Group, 
its subsidiaries, and affiliates hereby claim all applicable privileges related 
to this information.

-- 
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.


___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

Re: [asterisk-users] Which cause less CPU usage: GSM or wav??

[Michiel van Baak]

On 08:04, Mon 10 Sep 07, Barton Fisher wrote:
> Thanks Guys...  ulaw it is.  One more question if you don't mind.  If a 
> phase recorded as both .wav and .ulaw in the same folder, which will 
> asterisk pick using Playback(), Read() and Background() since you can't 
> specify the file extension in the command?
> I thought I change my script to begin recording new messages in ulaw 
> instead of converting them all to ulaw at once. So it's possible to have 
> two prompts with both file extension at a time

It will use the one for the channel codec.
So you can have a file in every format and asterisk will
pick the one that matches the channel codec.
-- 

Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD

"Why is it drug addicts and computer afficionados are both called users?"


___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

Re: [asterisk-users] Failover SIP logic

[Andrea Spadaccini]

Ciao Jeremy,

> I need some extensions logic assistance, I'm trying to dial out one of
> multiple SIP trunks, in sequence.  I need to detect a busy SIP trunk(I only
> allow 1 call per trunk) and roll over to a second or third depending on that
> busy status
> 
> Here's what I've got for a macro thusfar, but it's not working(fails if the
> 1st trunk is busy) extensions.conf:
> 
> [globals]
> trunk_1 => SIP/trunk1
> trunk_2 => SIP/trunk2
> trunk_3 => SIP/trunk3
> 
> [macro-trunkdial]
> exten => s,1,Dial(${trunk_1}/${ARG1})
> exten => s,2,Hangup()
> exten => s,102,Dial(${trunk_2}/${ARG1})
> exten => s,103,Hangup()
> exten => s,203,Dial(${trunk_3}/${ARG1})
> exten => s,204,Hangup()


Which asterisk version are you using?
IIRC, priority jumping (ie. going to n+101) was disabled by default in some
1.2.x version. You should rely on DIALSTATUS. See Dial() page in voip-info.org.

HTH,

-- 
Dr. Andrea Spadaccini
Multimedia Technologies Institute - MTI S.r.l.
Web: www.x-voice.it - Tel: +39 (0) 95 7224945

___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

Re: [asterisk-users] Which cause less CPU usage: GSM or wav??

[Thomas Kenyon]

Atis wrote:
> 
> A little caveat - "sox" doesn't understands file extensions used by
> asterisk (or it's just asterisk, trying to use file extensions that
> match codec name). So - some sox commandline hints:
> 
> ulaw: -t ul
> alaw: -t al
> slin: -t raw -s -w
> 
Or (since 1.4.0) in the asterisk cli type:

Convert /path/to/.wav /path/to/.ulaw


___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

Re: [asterisk-users] Connecting Legacy Pbx With Asterisk With FXS.

[Olivier]

Hello,

2007/9/10, C F <[EMAIL PROTECTED]>:
>
> Which Panasonic PBX?
>
> On 9/10/07, Sanspareils Greenlans <[EMAIL PROTECTED]> wrote:
> > Sir,
> >
> > I am having Asterisk pbx which is running without any problem now i want
> to
> > connect this with Panasonic pbx with FXS port so, if any body want to
> call
> > panasonic users than he will call or vise-versa.


How ?
Do you plan to dedicate Panasonic PBX FXS ports to act as a trunk or would
dedicate one port for each Asterisk user ?
___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

Re: [asterisk-users] Which cause less CPU usage: GSM or wav??

[Atis]

On 9/10/07, Barton Fisher <[EMAIL PROTECTED]> wrote:
> Thanks Guys...  ulaw it is.  One more question if you don't mind.  If a
> phase recorded as both .wav and .ulaw in the same folder, which will
> asterisk pick using Playback(), Read() and Background() since you can't
> specify the file extension in the command?
> I thought I change my script to begin recording new messages in ulaw
> instead of converting them all to ulaw at once. So it's possible to have
> two prompts with both file extension at a time

Asterisk will try to find file in codec currently in use, and if it
can't find, it will try to use file with less translation time (try
"show transcoding" in CLI). So - you can have files in all the codecs
used in your PBX, asterisk will choose most appropriate. The same goes
for MOH.

A little caveat - "sox" doesn't understands file extensions used by
asterisk (or it's just asterisk, trying to use file extensions that
match codec name). So - some sox commandline hints:

ulaw: -t ul
alaw: -t al
slin: -t raw -s -w

Regards,
Atis

-- 
Atis Lezdins,
IT Responsible of BEST Riga,
[EMAIL PROTECTED]
ICQ: 142239285
Skype: atis.lezdins
Cell Phone: +371 28806004 [Tele2, Latvia]
Work phone: +1 800 7502835 [Toll free, USA]
?BEST? -> www.BEST.eu.org

___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

Re: [asterisk-users] Failover SIP logic

[Yusuf]

X-ECN Telecoms-MailScanner-Information: Please contact ECN Telecoms for more 
information
X-ECN Telecoms-MailScanner: Found to be clean
X-ECN Telecoms-MailScanner-From: [EMAIL PROTECTED]
X-Spam-Status: No

Jeremy Mann wrote:
> I need some extensions logic assistance, I'm trying to dial out one of 
> multiple SIP trunks, in sequence.  I need to detect a busy SIP trunk(I only 
> allow 1 call per trunk) and roll over to a second or third depending on that 
> busy status
> 
> Here's what I've got for a macro thusfar, but it's not working(fails if the 
> 1st trunk is busy)
> extensions.conf:
> 
> [globals]
> trunk_1 => SIP/trunk1
> trunk_2 => SIP/trunk2
> trunk_3 => SIP/trunk3
> 
> [macro-trunkdial]
> exten => s,1,Dial(${trunk_1}/${ARG1})
> exten => s,2,Hangup()
> exten => s,102,Dial(${trunk_2}/${ARG1})
> exten => s,103,Hangup()
> exten => s,203,Dial(${trunk_3}/${ARG1})
> exten => s,204,Hangup()
> 
> [from-internal]
> exten => _NXXNXX,1,Macro(trunkdial,+1${EXTEN})
> exten => _1NXXNXX,1,Macro(trunkdial,+${EXTEN})
> 
> sip.conf:
> 
> [trunk1]
> host=xxx.xxx.xxx.xxx
> port=5060
> type=peer
> allow=ulaw
> dtmfmode=rfc2833
> canreinvite=no
> reinvite=no
> nat=no
> fromuser=+xxx
> call-limit=1
> 
> [trunk2]
> host=xxx.xxx.xxx.xxx
> port=5060
> type=peer
> allow=ulaw
> dtmfmode=rfc2833
> canreinvite=no
> reinvite=no
> nat=no
> fromuser=+xxx
> call-limit=1
> 
> [trunk3]
> host=xxx.xxx.xxx.xxx
> port=5060
> type=peer
> allow=ulaw
> dtmfmode=rfc2833
> canreinvite=no
> reinvite=no
> nat=no
> fromuser=+xxx
> call-limit=1
> 
> Here's asterisk output when someone dials out:
> Executing [EMAIL PROTECTED]:1] Macro("SIP/6001-007e2840", 
> "trunkdial|+1xx") in new stack
> -- Executing [EMAIL PROTECTED]:1] Dial("SIP/6001-007e2840", 
> "SIP/trunk1/+1xx") in new stack
> [Sep 10 09:06:52] ERROR[16253]: chan_sip.c:3192 update_call_counter: Call to 
> peer 'trunk1' rejected due to usage limit of 1
> -- Couldn't call trunk1/+1xx
>   == Everyone is busy/congested at this time (0:0/0/0)
> -- Executing [EMAIL PROTECTED]:2] Hangup("SIP/6001-007e2840", "") in new 
> stack
> 
> I don't want the dialplan to cascade like:
> 
> exten => 1,dial...
> exten => 2,dial...
> 
> Because if the remote end hangs up I don't want it going to priority 2 to 
> dial out again(in case my user doesn't hit hang-up on their end) so I need 
> logic to detect a busy channel and jump to the next section..

If you have this:

exten => _X.,1,Dial(SIP/trunk1)
exten => _X.,2,Dial(SIP/trunk2)
exten => _X.,3,Dial(SIP/trunk3)

then, only if trunk is busy, will it go to trunk2, if thats busy, it will go to 
trunk 3. 
Reason is, is that control wont return to the dial plan(except h) if the call 
was 
successfull.  SO if the call went through on trunk 1, then it will exit, not 
dial trunk2 
or trunk3.  So this dial plan will work.  But its very sequential, i.e. will 
try trunk1, 
then trunk2, then trunk3.  If you want to replicate round-robin, r, then do 
this:

[globals]
IPt=trunk1-trunk2-trunk3
COUNTt=0

NoOfChannels=3


[just-an-idea]
exten => _X.,1,Gotoif($["${COUNTt}" = "${NoOfChannels}"] ? 2:3)
exten => _X.,2,SetGlobalVar(COUNTt=0])
exten => _X.,3,SetGlobalVar(COUNTt=$[${COUNTt}+1])
exten => _X.,4,Set(tr=${CUT(IPt,-,${COUNTt})})
exten => _X.,5,Dial(SIP/tr/${EXTEN})


modify at your leisure.  So if you get a few more trunks, you just change 
"NoOfChannels"


-- 

thanks,
Yusuf

___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

Re: [asterisk-users] Which cause less CPU usage: GSM or wav??

[Jason Parker]

It will automatically pick the "best" recording for the current codec, so if
you are in ulaw, it will choose the ulaw prompt.

Barton Fisher wrote:
> Thanks Guys...  ulaw it is.  One more question if you don't mind.  If a
> phase recorded as both .wav and .ulaw in the same folder, which will
> asterisk pick using Playback(), Read() and Background() since you can't
> specify the file extension in the command?
> I thought I change my script to begin recording new messages in ulaw
> instead of converting them all to ulaw at once. So it's possible to have
> two prompts with both file extension at a time
> 
> Bart
> 

-- 
Jason Parker
Digium

___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

Re: [asterisk-users] Register Extension

[Tim Panton]

On 7 Sep 2007, at 17:56, phananhvu wrote:

> I means i want to use a software library to write a program that  
> register an extension to Asterisk system. After that, i can bind my  
> IP Phone to that extension.
> I wonder if Asterisk-Java can deal with this ??

Ah, you mean create an extension that a phone can register with ?
Last time I looked, the answer is no, Asterisk-java doesn't help you
create entries in extensions.conf or sip.conf .

The way I've done this in java is to map sip.conf (In my case iax.conf)
to a database table (see extconfig.conf). Then have your java write
to that database table using JDBC. After some trouble I even got it  
working with Oracle.

Tim.

___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

[asterisk-users] Failover SIP logic

[Jeremy Mann]

I need some extensions logic assistance, I'm trying to dial out one of multiple 
SIP trunks, in sequence.  I need to detect a busy SIP trunk(I only allow 1 call 
per trunk) and roll over to a second or third depending on that busy status

Here's what I've got for a macro thusfar, but it's not working(fails if the 1st 
trunk is busy)
extensions.conf:

[globals]
trunk_1 => SIP/trunk1
trunk_2 => SIP/trunk2
trunk_3 => SIP/trunk3

[macro-trunkdial]
exten => s,1,Dial(${trunk_1}/${ARG1})
exten => s,2,Hangup()
exten => s,102,Dial(${trunk_2}/${ARG1})
exten => s,103,Hangup()
exten => s,203,Dial(${trunk_3}/${ARG1})
exten => s,204,Hangup()

[from-internal]
exten => _NXXNXX,1,Macro(trunkdial,+1${EXTEN})
exten => _1NXXNXX,1,Macro(trunkdial,+${EXTEN})

sip.conf:

[trunk1]
host=xxx.xxx.xxx.xxx
port=5060
type=peer
allow=ulaw
dtmfmode=rfc2833
canreinvite=no
reinvite=no
nat=no
fromuser=+xxx
call-limit=1

[trunk2]
host=xxx.xxx.xxx.xxx
port=5060
type=peer
allow=ulaw
dtmfmode=rfc2833
canreinvite=no
reinvite=no
nat=no
fromuser=+xxx
call-limit=1

[trunk3]
host=xxx.xxx.xxx.xxx
port=5060
type=peer
allow=ulaw
dtmfmode=rfc2833
canreinvite=no
reinvite=no
nat=no
fromuser=+xxx
call-limit=1

Here's asterisk output when someone dials out:
Executing [EMAIL PROTECTED]:1] Macro("SIP/6001-007e2840", 
"trunkdial|+1xx") in new stack
-- Executing [EMAIL PROTECTED]:1] Dial("SIP/6001-007e2840", 
"SIP/trunk1/+1xx") in new stack
[Sep 10 09:06:52] ERROR[16253]: chan_sip.c:3192 update_call_counter: Call to 
peer 'trunk1' rejected due to usage limit of 1
-- Couldn't call trunk1/+1xx
  == Everyone is busy/congested at this time (0:0/0/0)
-- Executing [EMAIL PROTECTED]:2] Hangup("SIP/6001-007e2840", "") in new 
stack

I don't want the dialplan to cascade like:

exten => 1,dial...
exten => 2,dial...

Because if the remote end hangs up I don't want it going to priority 2 to dial 
out again(in case my user doesn't hit hang-up on their end) so I need logic to 
detect a busy channel and jump to the next section..


Thanks for any help.

This e-mail, facsimile, or letter and any files or attachments transmitted with 
it contains information that is confidential and privileged. This information 
is intended only for the use of the individual(s) and entity(ies) to whom it is 
addressed. If you are the intended recipient, further disclosures are 
prohibited without proper authorization. If you are not the intended recipient, 
any disclosure, copying, printing, or use of this information is strictly 
prohibited and possibly a violation of federal or state law and regulations. If 
you have received this information in error, please notify Texas Health 
Management Group immediately at 1-817-310-4999. Texas Health Management Group, 
its subsidiaries, and affiliates hereby claim all applicable privileges related 
to this information.

-- 
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.


___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

Re: [asterisk-users] Which cause less CPU usage: GSM or wav??

[Barton Fisher]

Thanks Guys...  ulaw it is.  One more question if you don't mind.  If a 
phase recorded as both .wav and .ulaw in the same folder, which will 
asterisk pick using Playback(), Read() and Background() since you can't 
specify the file extension in the command?
I thought I change my script to begin recording new messages in ulaw 
instead of converting them all to ulaw at once. So it's possible to have 
two prompts with both file extension at a time


Bart

Matt Riddell wrote:

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Barton Fisher wrote:
  

Thanks, OK, a bit confused  The cards are TE410P.  I really don't
see how the set a codec for this, other than it might default to
something in code like ulaw.  Any clue on how to verify codec in use
during a call?



Basically its going to be g711.ulaw for T1 (USA) and g711.alaw for E1
(rest of world) 99.9% of the time.

Unless you have something strange or different, I'd record in ulaw for T1.

- --
Kind Regards,

Matt Riddell
Director
___

http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss)
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.7 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iD8DBQFG5MI9DQNt8rg0Kp4RAgRWAKCL2l8egvLV2Xu3T754KJMzGXrKnQCfboCx
aFwrtGNKZ0EbZr176MDZUkY=
=HvDo
-END PGP SIGNATURE-



__ NOD32 2517 (20070910) Information __

This message was checked by NOD32 antivirus system.
http://www.eset.com



  


--

Barton Fisher
Innovative Communications
714-228-5400 Ext 5410
http://www.icpage.com

begin:vcard
fn:Barton Fisher
n:Fisher;Barton
org:Innovative Communications
adr:;;7439 La Palma Ave # 255;Buena Park;CA;90620;USA
email;internet:[EMAIL PROTECTED]
tel;work:714-228-5410
url:http://icpage.com
version:2.1
end:vcard

___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

Re: [asterisk-users] nat=yes

[Marco Bartholomew]

C F wrote:
> BTW, AFAIK, there is no such thing as host=static it's either dynamic
> or an IP/Name.
>   
>
Yeah, I learned that the hard way.  I had only set up dynamic devices 
for a couple of months, and the first time I had reason to set up a 
device with a static IP, I just assumed that 'host=static' would work in 
sip.conf.  Dur, it took me a couple of hours to figure out why my 
fax machine could fax, but not receive.

___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

Re: [asterisk-users] Broken UDP streams

[Al lists]

Maximum retries exceeded on transmission usually comes from NAT issues.
you can try this system without NAT and see if problem has resolved.


On 9/7/07, Adrian Marsh <[EMAIL PROTECTED]> wrote:
>
>  Hi All,
>
>
>
> I'm working from home today (DSL -> Internet -> 2MB leased line -> A*K
> server behind NAT), and trying to pickup voicemail using Zoiper..
>
> I can access the VM system, I hear all the prompts, and I can even hear
> part of the message playback.
>
> But then I get silence on the call (call stays up), and I get:
>
>
>
> Parsing '/var/spool/asterisk/voicemail/default/2027/Old/msg.txt':
> Found
>
> -- Playing '/var/spool/asterisk/voicemail/default/2027/Old/msg'
> (language 'en')
>
> Sep  7 13:51:30 WARNING[30737]: chan_sip.c:1228 retrans_pkt: Maximum
> retries exceeded on transmission
> NmM3YmNhNjk0NzhhMjFlYmU5Yzg1YTBmNThlZDNhYWQ. for seqno 2 (Critical Response)
>
> Sep  7 13:51:30 WARNING[30737]: chan_sip.c:1245 retrans_pkt: Hanging up
> call NmM3YmNhNjk0NzhhMjFlYmU5Yzg1YTBmNThlZDNhYWQ. - no reply to our critical
> packet.
>
> == Spawn extension (from-sip, voicemail, 4) exited non-zero on
> 'SIP/427-b780fa40'
>
>
>
> On the A8k log.
>
>
>
> I'm guessing packets are getting lost, but don't understand why it would
> only be in VM playback that it happens.
>
>
>
> Any ideas?
>
>
>
> Adrian
>
> ___
>
> Sign up now for AstriCon 2007!  September 25-28th.
> http://www.astricon.net/
>
> --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
>
___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

Re: [asterisk-users] DTMF Relay Problems

[Joseph Begumisa]

Actually this problem is with a telco in the US [the setup is in the US]. I
will get in touch with them to have them look into it.  There is another
similar setup with the same telco and there are no such problems.  The only
difference in the setups is that in this case, the T1 is terminated into a
Cisco 2430 Integrated Access Device and then a T1 from that device
terminates into the Asterisk PBX.  Probably I will have them bypass the
Cisco device and see whether I can replicate this again.

Joseph.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tony
Mountifield
Sent: Monday, September 10, 2007 7:39 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] DTMF Relay Problems

In article <[EMAIL PROTECTED]>,
Joseph Begumisa <[EMAIL PROTECTED]> wrote:
> Thanks.  My results after applying the patch and recompiling are that the
> problem can only be replicated with calls from mobile networks.  Digits
like
> 160 entered in the digital receptionist by a caller are received by the
> asterisk server as 16660 sometimes.  Other times it is received as 1660.
> Digits like 1234 are received as 1222334 etc...  From fixed lines, there
is
> no problem.  Digits are received as they have been sent.
> 
> Any other pointers?

Hmm, that sounds like a problem with the GSM-to-PSTN gateway that the calls
are passing through.

Unless things are different in Uganda, I believe when a user presses a DTMF
key on their mobile, it doesn't send a tone through the mobile network, but
rather a "start dtmf" control message followed by a "stop dtmf" control
message. When the call gets gatewayed from GSM to the PSTN network, it is
the job of the gateway to generate the tones as instructed by the control
protocol. (Someone please correct me if I'm wrong).

So you may need to take it up with your telco.

Cheers
Tony

> Thanks a lot.
> 
> Joseph
> 
> 
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Michiel van
> Baak
> Sent: Sunday, September 09, 2007 12:21 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] DTMF Relay Problems
> 
> On 12:09, Sun 09 Sep 07, Joseph Begumisa wrote:
> > I applied the patch, however, I'd like to know which particular files to
> > copy after running a make.  I do not wish to run "make install" as it
will
> > overwrite other configuration changes I have made.  
> 
> A make install will not overwrite any configfile.
> It will install the asterisk binary and the modules (thus
> overwriting the existing files) but configfiles will only be
> overwritten when you run: make samples
> 
> -- 
> 
> Michiel van Baak
> [EMAIL PROTECTED]
> http://michiel.vanbaak.eu
> GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD
> 
> "Why is it drug addicts and computer afficionados are both called users?"
> 
> 
> ___
> 
> Sign up now for AstriCon 2007!  September 25-28th.
http://www.astricon.net/
> 
> 
> --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
> 
> 
> 
> 
> ___
> 
> Sign up now for AstriCon 2007!  September 25-28th.
http://www.astricon.net/ 
> 
> --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
> 


-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org

___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/


--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




___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

Re: [asterisk-users] Which cause less CPU usage: GSM or wav??

[James FitzGibbon]

On 9/9/07, Barton Fisher <[EMAIL PROTECTED]> wrote:
>
> Thanks, OK, a bit confused  The cards are TE410P.  I really don't
> see how the set a codec for this, other than it might default to
> something in code like ulaw.  Any clue on how to verify codec in use
> during a call?


If you absolutely want to be sure, use 'pri intense debug span X' and watch
for SETUP messages:

< Protocol Discriminator: Q.931 (8)  len=62
< Call Ref: len= 2 (reference 542/0x21E) (Originator)
< Message type: SETUP (5)
< [04 03 80 90 a2]
< Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer
capability: Speech (0)
<  Ext: 1  Trans mode/rate: 64kbps, circuit-mode
(16)
<  Ext: 1  User information layer 1: u-Law (34)
< [18 03 a9 83 95]
< Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare: 0  Exclusive
Dchan: 0

Re: [asterisk-users] online active call watching

[Yehavi Bourvine +972-8-9489444]

try the astman command.

   __Yehavi:

___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

Re: [asterisk-users] nat=yes

[C F]

So I'll rephrase to some devices will not operate properly, since
after your message I am assuming that you tested this with most
devices.

On 9/10/07, Benjamin Jacob <[EMAIL PROTECTED]> wrote:
> C F, I have nat=yes set by default for all my extensions(with
> canreinvite=no). And things work fine.
>
> Bilal, about Asterisk sending packets to public/private :
> Asterisk will send packets to the public IP advertised by the msg/recv
> from address. It is the NAT's headache on the endpoints network
> periphery to send the response from Asterisk to the endpoint.
>
>
> C F wrote:
>
> >If you set yes then asterisk assumes that the address its coming from
> >is not the same as the UA thinks it is. most devices will not operate
> >properly if set to yes when they are in fact local.
> >
> >On 9/9/07, bilal ghayyad <[EMAIL PROTECTED]> wrote:
> >
> >
> >>Hi List;
> >>
> >>If I set nat=yes, then asterisk will send the packets
> >>to the public IP address or to the private IP address
> >>(which will be for the endpoint that is behind the
> >>nating)?
> >>
> >>And by setting the nat=yes, then what exactly will be
> >>ignored at asterisk side when reading the
> >>registeration messages from the endpoint?
> >>
> >>Any help.
> >>
> >>Regards
> >>Bilal
> >>
> >>
> >>
> >>
> >>Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated
> >>for today's economy) at Yahoo! Games.
> >>http://get.games.yahoo.com/proddesc?gamekey=monopolyherenow
> >>
> >>___
> >>
> >>Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/
> >>
> >>--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
> >>
> >>
> >>
> >
> >___
> >
> >Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/
> >
> >--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
> >
> >
> >
>
>
> EMAIL DISCLAIMER : This email and any files transmitted with it are 
> confidential and intended solely for the use of the individual or entity to 
> whom they are addressed. Any unauthorised distribution or copying is strictly 
> prohibited. If you receive this transmission in error, please notify the 
> sender by reply email and then destroy the message. Opinions, conclusions and 
> other information in this message that do not relate to official business of 
> Mascon shall be understood to be neither given nor endorsed by Mascon. Any 
> information contained in this email, when addressed to Mascon clients is 
> subject to the terms and conditions in governing client contract.
>
> Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, 
> we can not guarantee that any email or attachment is free from computer 
> viruses and you are strongly advised to undertake your own anti-virus 
> precautions. Mascon grants no warranties regarding performance, use or 
> quality of any e-mail or attachment and undertakes no liability for loss or 
> damage, howsoever caused.
>
>
>
> ___
>
> Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/
>
> --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
>

___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

Re: [asterisk-users] Maximum retries exceeded on transmission

[Apa Minerala]

Tom,
The device is voxbone from voxbone.com . I am using a DID as an access 
number...it worked with same config with asterisk 1.2.12 and a2billing 1.2.3, 
but doesn't work with asterisk 1.4.11 and a2billing 1.3 

Can you tell me what am I missing?

Apa

Tom Lynn <[EMAIL PROTECTED]> wrote: I suspect if you remove the callerid entry 
from this device's sip.conf definition things will work better.  

On 9/9/07, Apa Minerala < [EMAIL PROTECTED]> wrote:
 
 I have searched this list and others, and see other pepole having this 
 issue. However, I have not seen how to fix it.
 
 Sep 6 18:52:36 *WARNING*[4620]: *chan_sip.c*:*1835 retrans_pkt*: Maximum
 retries exceeded on transmission
 778f89593967725f0abe40eb1752504c for seqno 1620 (Critical 
 Response)
 
 Sep 6 18:52:36 *WARNING*[4620]: *chan_sip.c*:*1835 retrans_pkt*: Hanging up
 call 778f89593967725f0abe40eb1752504c no reply to our critical
 packet.
 
 What is the critical packet that is not being responded to? Please help. 
 
 

-
 Pinpoint customers who are looking for what you sell.   

___

Sign up now for AstriCon 2007!  September 25-28th.   http://www.astricon.net/

--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 


 ___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

   
-
Be a better Globetrotter. Get better travel answers from someone who knows.
Yahoo! Answers - Check it out.___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

Re: [asterisk-users] Connecting Legacy Pbx With Asterisk With FXS.

[C F]

Which Panasonic PBX?

On 9/10/07, Sanspareils Greenlans <[EMAIL PROTECTED]> wrote:
> Sir,
>
> I am having Asterisk pbx which is running without any problem now i want to
> connect this with Panasonic pbx with FXS port so, if any body want to call
> panasonic users than he will call or vise-versa. i want to connect only two
> extension with Asterisk so, all communication done only on these two line.
>
> what is the process and what is the setting in sip.conf and extensions.conf to
> communicate with Asterisk and Panasonic pbx.
>
> Rajeev.
>
> ___
>
> Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/
>
> --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
>

___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

Re: [asterisk-users] DTMF bug in dsp.c and 1.4.11

[Tony Mountifield]

In article <[EMAIL PROTECTED]>,
Jerry Geis <[EMAIL PROTECTED]> wrote:
> I was wondering if this bug: http://bugs.digium.com/view.php?id=10535
> would affect a PRI connection.
> 
> I seem to be dropping DTMF digits on the PRI.
> The company says they have test the line and they way the PRI is fine
> as far as they are concerned.
> 
> So will this bug and patch help me? I am running 1.4.11

Yes, that bug was submitted by me, and it was a PRI on which I was having
the problems.

If there is a slight bounce on the leading edge of a digit, then it can
easily be dropped altogether by Asterisk. The patch fixes that, and also
adds debouncing of the trailing edge (else a trailing bounce might give
a double-digit).

Give it a try - I expect it will help a lot.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org

___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

Re: [asterisk-users] Which cause less CPU usage: GSM or wav??

[Al lists]

Also your Disk subsystem speed.
having disk RAM , makes sense in your case.

On 9/10/07, Thomas Kenyon <[EMAIL PROTECTED]> wrote:
>
> Barton Fisher wrote:
> > Thanks, OK, a bit confused  The cards are TE410P.  I really don't
> > see how the set a codec for this, other than it might default to
> > something in code like ulaw.  Any clue on how to verify codec in use
> > during a call?
> >
> G.711ulaw and G.711alaw are the audio transmission methods used for
> ISDN. If you have a T1 line then the transmission method is G.711ulaw.
>
> I've been told that if you play a ulaw signal down an alaw line (T1
> signal down E1) then at the other end the voice sounds a bit like a
> dalek. (Iit's very hard to do this with asterisk since it automatically
> transcodes between endpoints).
>
> The lack of a performance hit is quite striking when you have a
> recording playing back as a native format rather than being transcoded.
> (well, it's quite striking when you have thousands of them running
> simultaneously).
>
> > Bart
> >
> > Steve Totaro wrote:
> >> Michiel van Baak wrote:
> >>
> >>> On 10:28, Sun 09 Sep 07, Barton Fisher wrote:
> >>>
>  I have 4 TDM T1's going in to a IVR system.  The IVR messages are
>  recorded .wav format - The system appears to crap out at about 40
>  calls - Would using GSM or some other format help save CPU cycles?
>  Using 1.2, Dual Xeon and 2GB ram
> 
> >>> depends on what codec the T1 is using.
> >>> Best to transcode the ivr sounds to the same codec to
> >>> prevent on-the-fly transcoding by asterisk.
> >>>
> >>>
> >> The answer is going to ulaw or alaw depending where you live.  T1
> >> should most likely be using ulaw so make everything ulaw, end to end.
> >>
> >> Thanks,
> >> Steve Totaro
> >>
> >> ___
> >>
> >> Sign up now for AstriCon 2007!  September 25-28th.
> >> http://www.astricon.net/
> >> --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
> >>
> >>
> >>
> >> __ NOD32 2516 (20070909) Information __
> >>
> >> This message was checked by NOD32 antivirus system.
> >> http://www.eset.com
> >>
> >>
> >>
> >>
> >
> > ___
> >
> > Sign up now for AstriCon 2007!  September 25-28th.
> http://www.astricon.net/
> >
> > --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
>
> ___
>
> Sign up now for AstriCon 2007!  September 25-28th.
> http://www.astricon.net/
>
> --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
>
___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

Re: [asterisk-users] What is the difference between increasingtheverbose level and the debug level?

[Steve Langstaff]

Except in the cases where what you observe in real life is buggy
behaviour, and not what the designer/implementor intended.

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of Dovid B
> Sent: 10 September 2007 12:34
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] What is the difference between 
> increasingtheverbose level and the debug level?
> 
> I just want to add that it is the best way to learn.  Till 
> today I thank those on the list that told me to stay away 
> from GUI's and learn "the real asterisk".
> 
> If you still can't figure out the difference I can help you 
> out but it is better if you learn on your own.
> 
> - Original Message -
> From: "C F" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
> 
> Sent: Monday, September 10, 2007 2:42 AM
> Subject: Re: [asterisk-users] What is the difference between 
> increasing theverbose level and the debug level?
> 
> 
> > In general keep in mind, asterisk is very user friendly and 
> wont bite 
> > :). Trial and error is a good friend to get to know 
> asterisk so that 
> > you know what all of these mean.
> >
> > On 9/9/07, bilal ghayyad <[EMAIL PROTECTED]> wrote:
> >> Hi List;
> >>
> >> What is the difference between increasing the verbose 
> level and the 
> >> debug level?
> 

___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

Re: [asterisk-users] DTMF Relay Problems

[Tony Mountifield]

In article <[EMAIL PROTECTED]>,
Joseph Begumisa <[EMAIL PROTECTED]> wrote:
> Thanks.  My results after applying the patch and recompiling are that the
> problem can only be replicated with calls from mobile networks.  Digits like
> 160 entered in the digital receptionist by a caller are received by the
> asterisk server as 16660 sometimes.  Other times it is received as 1660.
> Digits like 1234 are received as 1222334 etc...  From fixed lines, there is
> no problem.  Digits are received as they have been sent.
> 
> Any other pointers?

Hmm, that sounds like a problem with the GSM-to-PSTN gateway that the calls
are passing through.

Unless things are different in Uganda, I believe when a user presses a DTMF
key on their mobile, it doesn't send a tone through the mobile network, but
rather a "start dtmf" control message followed by a "stop dtmf" control
message. When the call gets gatewayed from GSM to the PSTN network, it is
the job of the gateway to generate the tones as instructed by the control
protocol. (Someone please correct me if I'm wrong).

So you may need to take it up with your telco.

Cheers
Tony

> Thanks a lot.
> 
> Joseph
> 
> 
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Michiel van
> Baak
> Sent: Sunday, September 09, 2007 12:21 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] DTMF Relay Problems
> 
> On 12:09, Sun 09 Sep 07, Joseph Begumisa wrote:
> > I applied the patch, however, I'd like to know which particular files to
> > copy after running a make.  I do not wish to run "make install" as it will
> > overwrite other configuration changes I have made.  
> 
> A make install will not overwrite any configfile.
> It will install the asterisk binary and the modules (thus
> overwriting the existing files) but configfiles will only be
> overwritten when you run: make samples
> 
> -- 
> 
> Michiel van Baak
> [EMAIL PROTECTED]
> http://michiel.vanbaak.eu
> GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD
> 
> "Why is it drug addicts and computer afficionados are both called users?"
> 
> 
> ___
> 
> Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/
> 
> 
> --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
> 
> 
> 
> 
> ___
> 
> Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 
> 
> --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
> 


-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org

___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

Re: [asterisk-users] USA Termination

[Dovid B]

There is a Biz list for a reason. Please look at the emails headers 
"Non-Commercial Discussion"

- Original Message - 
From: "Claude Cunningham" <[EMAIL PROTECTED]>
To: "Commercial and Business-Oriented Asterisk Discussion" 
<[EMAIL PROTECTED]>; "Asterisk Users Mailing List - 
Non-Commercial Discussion" 
Sent: Monday, September 10, 2007 12:09 PM
Subject: [asterisk-users] USA Termination


> Send us your traffic, we can terminate it in the USA  for you ---
>
> $.00475 US  TERMINATION. International Origination Traffic  sent with
> international CLI*  1/1 Billing  50,000/day $.006/minute 100,000/day
> $.00575/minute   250,000/day  $.00555/minute   500,000/day
> $.0050/minute  1,000,000/day $.00475/minute
>
> "off-net" traffic$.011/minute
>
> statsASR 87%   ACD 9+   G711/729SIP or H323.
> Therefore on-net % will increase.
>
> Unlimited Port Capacity. Our  footprint is largest.
>
> Unlimited Port Capacity. Our  footprint is largest.
>
> Send us your CDRs &  we will analyze for "on-net" and
> "off-net"traffic ratio. US CLI can replace  international CLI sent,
> Unlimited capacity, SIP or H323,  G711 or G729.
>
>
> Email me off list - [EMAIL PROTECTED]
>
> Claude +1 954 905 8612
>
> ___
>
> Sign up now for AstriCon 2007!  September 25-28th. 
> http://www.astricon.net/
>
> --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
> 



___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

Re: [asterisk-users] What is the difference between increasing theverbose level and the debug level?

[Dovid B]

I just want to add that it is the best way to learn.  Till today I thank 
those on the list that told me to stay away from GUI's and learn "the real 
asterisk".

If you still can't figure out the difference I can help you out but it is 
better if you learn on your own.

- Original Message - 
From: "C F" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Monday, September 10, 2007 2:42 AM
Subject: Re: [asterisk-users] What is the difference between increasing 
theverbose level and the debug level?


> In general keep in mind, asterisk is very user friendly and wont bite
> :). Trial and error is a good friend to get to know asterisk so that
> you know what all of these mean.
>
> On 9/9/07, bilal ghayyad <[EMAIL PROTECTED]> wrote:
>> Hi List;
>>
>> What is the difference between increasing the verbose
>> level and the debug level?
>>
>> By increasing the verbose level, then I will get more
>> traces messages and by increasing the debug level, I
>> will also get more traces messages. So what is the
>> difference?
>>
>> Any help?
>> Regards
>> Bilal Ghayad
>>
>>
>>
>> 
>> Yahoo! oneSearch: Finally, mobile search
>> that gives answers, not web links.
>> http://mobile.yahoo.com/mobileweb/onesearch?refer=1ONXIC
>>
>> ___
>>
>> Sign up now for AstriCon 2007!  September 25-28th. 
>> http://www.astricon.net/
>>
>> --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
>>
>
> ___
>
> Sign up now for AstriCon 2007!  September 25-28th. 
> http://www.astricon.net/
>
> --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
> 



___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

Re: [asterisk-users] Cascading queues & calls not joining unavailable queues.

[Sander Smeenk]

Quoting James FitzGibbon ([EMAIL PROTECTED]):

> Unfortunately, the patches weren't done against trunk or the head of 1.4,
> and the author didn't file a disclaimer with Mantis, so the bug (
> http://bugs.digium.com/view.php?id=9165) was recently closed.

That's just too bad, as this might be a solution to our 'problems'. :)

-- 
| The less hair I have, the more head I get!
| 1024D/08CEC94D - 34B3 3314 B146 E13C 70C8  9BDB D463 7E41 08CE C94D

___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

Re: [asterisk-users] Cascading queues & calls not joining unavailable queues.

[Sander Smeenk]

Quoting Mark Michelson ([EMAIL PROTECTED]):

> > -- Called SCCP/231
> > -- Called SCCP/220
> > -- SCCP/220-009b is busy
> > -- SCCP/231-009a is busy
> > I'd like asterisk to quit trying when all agents are busy, but i don't
> > think it's possible without scripting it yourself with some AGI-script
> > that checks 'show queues' output.

> It sounds as though skinny devices may not be reporting their device 
> state correctly, and so the queue believes that the devices are 
> available.

Looking at the output of 'show queues' everything looks completely OK
when i put the phone in various states of 'being available'. I think
it's more an opinion on what 'unavailable' is.

> Or perhaps they are reporting a state that the queue does not know
> about. If this is the case, we may be dealing with a bug. I will test
> locally when I can get access to a Skinny phone and see what's going on.

We're using chan_sccp.so in combination with Cisco 796x phones (With CTU
ringtone! Whee! :P). Maybe it doesn't really work right because of this,
but as Asterisk *tells me* it knows nobody is answering a queue, i
wonder why it keeps trying ;-)

Kind regards,
Sander.
-- 
| If you jog backwards, will you gain weight?
| 1024D/08CEC94D - 34B3 3314 B146 E13C 70C8  9BDB D463 7E41 08CE C94D

___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

[asterisk-users] 56k modem configuration

[Andrea Spadaccini]

Hello everybody,
I've got a 56k usb modem, lsusb says:

Bus 002 Device 002: ID 0572:130 Conexant Systems (Rockwell), Inc. 

I'd like to let it work with Asterisk. I think that I should use chan_modem
and/or chan_modem_bestdata, but I found little or no documentation.

Can anybody please post some instructions?

Thanks in advance,

-- 
Dr. Andrea Spadaccini
Multimedia Technologies Institute - MTI S.r.l.
Web: www.x-voice.it - Tel: +39 (0) 95 7224945

___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

Re: [asterisk-users] Which cause less CPU usage: GSM or wav??

[Thomas Kenyon]

Barton Fisher wrote:
> Thanks, OK, a bit confused  The cards are TE410P.  I really don't
> see how the set a codec for this, other than it might default to
> something in code like ulaw.  Any clue on how to verify codec in use
> during a call?
> 
G.711ulaw and G.711alaw are the audio transmission methods used for
ISDN. If you have a T1 line then the transmission method is G.711ulaw.

I've been told that if you play a ulaw signal down an alaw line (T1
signal down E1) then at the other end the voice sounds a bit like a
dalek. (Iit's very hard to do this with asterisk since it automatically
transcodes between endpoints).

The lack of a performance hit is quite striking when you have a
recording playing back as a native format rather than being transcoded.
(well, it's quite striking when you have thousands of them running
simultaneously).

> Bart
> 
> Steve Totaro wrote:
>> Michiel van Baak wrote:
>>  
>>> On 10:28, Sun 09 Sep 07, Barton Fisher wrote:
>>>  
 I have 4 TDM T1's going in to a IVR system.  The IVR messages are
 recorded .wav format - The system appears to crap out at about 40
 calls - Would using GSM or some other format help save CPU cycles?
 Using 1.2, Dual Xeon and 2GB ram
   
>>> depends on what codec the T1 is using.
>>> Best to transcode the ivr sounds to the same codec to
>>> prevent on-the-fly transcoding by asterisk.
>>>
>>>   
>> The answer is going to ulaw or alaw depending where you live.  T1
>> should most likely be using ulaw so make everything ulaw, end to end.
>>
>> Thanks,
>> Steve Totaro
>>
>> ___
>>
>> Sign up now for AstriCon 2007!  September 25-28th. 
>> http://www.astricon.net/
>> --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
>>
>>
>>
>> __ NOD32 2516 (20070909) Information __
>>
>> This message was checked by NOD32 antivirus system.
>> http://www.eset.com
>>
>>
>>
>>   
> 
> ___
> 
> Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 
> 
> --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

___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

Re: [asterisk-users] online active call watching

[Doug Lytle]

satish patel wrote:
> Dear all
>
>I have asterisk 1.4.11 i am new in asterisk i want 
> to see online call list how it is possible to see how man call 
> currently active is there any command or tool to see online call ?? 
> from --- to
Flash Operator Panel is what you'd want to look at:

http://www.asternic.org

Doug

-- 
Ben Franklin quote:

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



___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

[asterisk-users] USA Termination

[Claude Cunningham]

Send us your traffic, we can terminate it in the USA  for you ---

$.00475 US  TERMINATION. International Origination Traffic  sent with
international CLI*  1/1 Billing  50,000/day $.006/minute 100,000/day
$.00575/minute   250,000/day  $.00555/minute   500,000/day
$.0050/minute  1,000,000/day $.00475/minute

"off-net" traffic$.011/minute

 statsASR 87%   ACD 9+   G711/729SIP or H323.
Therefore on-net % will increase.

Unlimited Port Capacity. Our  footprint is largest.

Unlimited Port Capacity. Our  footprint is largest.

 Send us your CDRs &  we will analyze for "on-net" and
"off-net"traffic ratio. US CLI can replace  international CLI sent,
Unlimited capacity, SIP or H323,  G711 or G729.


Email me off list - [EMAIL PROTECTED]

Claude +1 954 905 8612

___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

Re: [asterisk-users] online active call watching

[Tzafrir Cohen]

On Sun, Sep 09, 2007 at 11:37:03PM -0700, satish patel wrote:
> Dear all
> 
>I have asterisk 1.4.11 i am new in asterisk i want 
> to see online call list how it is possible to see how man call 
> currently active is there any command or tool to see online call ?? from 
> --- to 

You can list the channels of Asterisk. While channels are not exactly
calls (a call can span over two channels), it gives you a good idea.

An occasional 'show channels' from the CLI, a terminal with:

  watch "asterisk -n -rx 'show channels'"

and the astman tool included with Asterisk are basically that.

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

Re: [asterisk-users] Strange Behaviour

[Il Neofita]

Thank you I will try tonight

On 9/10/07, Anselm Martin Hoffmeister <[EMAIL PROTECTED]> wrote:
>
> Am Montag, den 10.09.2007, 05:14 +0200 schrieb Il Neofita:
> > On 9/9/07, Anselm Martin Hoffmeister <[EMAIL PROTECTED]>
> > wrote:
> > Am Sonntag, den 09.09.2007, 20:16 +0200 schrieb Il Neofita:
> >
> > Well, it seems there are differences between those accounts
> > then.
> >
> > You might want to post your sip.conf, and -if that is
> > possible- the ATA
> > conf file; or at least a writedown of the configuration there.
> >
> > First of all, thank you for you reply
> > The ATA is the Fritz!Box and I tried with different FW version but I
> > have the same behaviour
>
> I have been using FritzBoxes for quite a while, and have not found such
> strange bugs - except after a Firmware Upgrade. It seems after some
> upgrades you need to do a "factory reset" (via the web interface) and
> enter your data again, else they behave stupidly.
>
> > this is part of the sip.conf
> > [180]
> > type=peer
> > username=180
> > secret=aa
> > callerid=First<180>
> > canreinvite = yes
> > host = dynamic
> > dtmfmode = rfc2833
> > qualify = yes
> > nat = yes
> > context = mycont
> > disallow = all
> > allow = g726
> > allow = g723
> > allow = ulaw
> > allow = alaw
> > allow = g729
> > allow = gsm
> >
> > [181]
> > type=peer
> > username=181
> > secret=bb
> > callerid=Second<181>
> > canreinvite = yes
> > host = dynamic
> > dtmfmode = rfc2833
> > qualify = yes
> > nat = yes
> > context = mycont
> > disallow = all
> > allow = g726
> > allow = g723
> > allow = ulaw
> > allow = alaw
> > allow = g729
> > allow = gsm
>
> Looks pretty OK to me. Just a stupid idea: Do you have a [general]
> section before those two?
>
> And then, I use type=friend, not type=peer, that _might_ make a
> difference in how asterisk matches sip.conf contexts to registered
> clients.
>
> 8< From my sip.conf:
> [sip501]
> mailbox=01
> callerid=501
> type=friend
> username=sip501
> secret=lk1j2eu89
> context=sipclient
> host=dynamic
> nat=yes
> disallow=all
> allow=alaw
> allow=gsm
> allow=ulaw
>
> [sip502]
> mailbox=02
> callerid=502
> type=friend
> username=sip502
> secret=1092jd0
> context=sipclient
> host=dynamic
> nat=yes
> disallow=all
> allow=alaw
> allow=gsm
> allow=ulaw
> =>8
>
> Note: Those two accounts belong to the same FritzBox.
>
> > I tried to switch the account for the two ports but what it is
> > important is only the order in the sip.conf
>
> That made me think about that friend/peer thingy.
>
> > I found some information in german and I do not know it
>
> The FritzBoxes are popular here in Germany - no wonder, being a German
> manufactured product and being given away for (nearly) free with any
> 2-year DSL contract... I like them nevertheless :)
>
> BR, HTH
>
> Anselm
>
>
>
> ___
>
> Sign up now for AstriCon 2007!  September 25-28th.
> http://www.astricon.net/
>
> --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
>
___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

Re: [asterisk-users] online active call watching

[ram]

On 9/10/07, satish patel <[EMAIL PROTECTED]> wrote:
>
> Dear all
>
>I have asterisk 1.4.11 i am new in asterisk i want to
> see online call list how it is possible to see how man call currently active
> is there any command or tool to see online call ?? from --- to



Hi

with the CDR+mysql

you can make query Invite+ack

ram
___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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

[asterisk-users] New Project: AskoziaPBX

[Michael Iedema]

Greetings everyone,

I've been working on a (yet another) "all-in-one" Asterisk based
project. It is aimed at embedded / low power systems (but scales fine
on more capable hardware) and is based on Asterisk 1.4.x and FreeBSD
6.2. Because of this, I've mostly been hanging out on the asterisk-bsd
list as bugs rolled in and the system's features were improved. We're
currently at public beta 10 after releasing pb1 in June and, I hope,
ready to announce this to a bit larger audience.

This is not a live-cd but rather an image that must initially be
written to a disk, so a dedicated machine is needed. After that, the
entire system is upgradeable through the webGUI. Anyone familiar with
the m0n0wall project (http://m0n0.ch/wall) will feel right at home as
AskoziaPBX was forked from it.

Here are the quick facts from the website and a link to the page:

* ~11 MB firmware image
* PHP based GUI accessible via http(s)
* based on Asterisk 1.4 and FreeBSD 6.2
* designed for embedded / low resource systems
* images available for the following platforms:
  * generic pc
  * pc engines wrap
  * soekris net48xx
  * VMware player
* GUI currently configures:
  * SIP, IAX, ISDN and Analog phones and providers
  * Conferencing
  * Voicemail (forwarded as e-mail attachment)
  * Call Groups
  * Call Parking
  * ...as well as all system settings (ntp, GUI port, etc.)
* all configuration stored in a single XML file
* Multilingual audio-prompts:
  * Dutch, English, French, German, Italian, Japanese,
Russian, Spanish, Swedish
* Multilingual voicemail notification e-mails:
  * Dutch, English, French, German, Italian, Polish, Spanish, Swedish

site: http://askozia.com/pbx

Thanks goes out to everyone in asterisk-bsd and pbx-users for testing
/ reporting and quite a few people in IRC who helped troubleshoot bugs
as they popped up! (Also, please remember that this is still a beta.)

Regards,
-Michael I.

___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--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