[Asterisk-Users] ASTCC: customer wants 100 accounts

[Ronald Wiplinger]

I got a request for one customers to set-up 100 accounts.

I use usually the Caller-ID as the card number.
Is there a way to make it for 100 accounts easier?

To generate 100 cards is not a problem, but if it would work with one 
account number  would be even better


I could use a different context for this customer and use only his 
account code as card number.


Any advice would be appreciated.


bye

Ronald Wiplinger
___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [Asterisk-Users] using variable

[unplug]

How? Can u show me?

On 6/27/06, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:

On Tue, Jun 27, 2006 at 12:13:31PM +0800, unplug wrote:
> Hi,
>
>  How can I access the variable in marco?  Say, there is a dial plan
> below.  In line 4, it will show the variable FOO=1234.  However, the
> variable in line 2 is nothing.  Can I assign a varilable in macro and
> access it outside a macro in the same session only?

Yes.

>
>
> [dialplan]
> 1: exten => 1234,1,Macro(test)
> 2: exten => 1234,2,NoOp(${FOO})
>
> [macro-test]
> 3: exten => s,1,Set(FOO=1234)
> 4: exten => s,2,NoOp(${FOO})

--
Tzafrir Cohen  sip:[EMAIL PROTECTED]
icq#16849755   iax:[EMAIL PROTECTED]
+972-50-7952406
[EMAIL PROTECTED]  http://www.xorcom.com
___
--Bandwidth and Colocation provided by Easynews.com --

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


___
--Bandwidth and Colocation provided by Easynews.com --

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

RE: [Asterisk-Users] Oh oh. Micro$oft just noticed VoIP

[Francesco Peeters (Asterisk)]

On Tue, June 27, 2006 0:26, shadowym said:
> They have been talking about this for awhile.  If you look at the real
> time
> and embedded operating system world they have not really done so well over
> the many years they have been trying. Just throwing money at the problem
> has
> never worked for them in the past either.

Perhaps because people expect devices like that to Just Work(tm),
something Embedded Linux is better known for than Embedded Windows is?...

> The Asterisk community has nothing to worry about in the near term if ever
> IMHO.
>

Unless they buy Digium... That'd give them a serious amount of code to
obfuscate and hide in closed source products!   ;-)

-- 
F Peeters
  PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch
  2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0
  AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN
  2 Sweex HFC-PCI cards
___
--Bandwidth and Colocation provided by Easynews.com --

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

RE : [Asterisk-Users] x100p buying advice

[f6hqz-m]

Hello,

Ridiculous business argumentation... 
By changing 2 resistors maping on the same card you can say to system that
is any response as X100P, X101P, or Clone.
No proof to good quality or if it realy run !
Take a look to voip-info.org about X100P and X101P, you will learn more
about the chipsets, which is only the available information. 

Use this kind of cards (WinModem) for tests or curses only, no production. 

Good Luck.
Francois BERGERET,
France.

-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Rod Morison
Envoyé : mardi 27 juin 2006 02:16
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : [Asterisk-Users] x100p buying advice


I'm looking to get an x100p off ebay and am not particularly familar 
with the "life cycle" of the card.. An "Authentic X100P" listing has a 
buy it now of $29.95 and says

There are 3 types of cards Asterisk would recognize: *Screenshots from the
official, original driver install
Cheap "OEM X100P","Clones", "Compatibles", Knock-Offs
   Found a Wildcard FXO: Generic Clone
The X101P (note the 101, not 100) is a Low-end version of X100P 
which uses low grade chips
   Found a Wildcard FXO: X101P
Authentic, Original X100P Speaks for Itself!
   Found a Wildcard FXO: X100P


 From what I gather clone's and knockoffs will have trouble with 
callerid. Is the "Found a Wildcard FXO: X100P" enough to establish full 
featured hardware (assuming an honest seller)? Is there another 
recommended source besides ebay in this price range?

Thanks


___
--Bandwidth and Colocation provided by Easynews.com --

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

___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [Asterisk-Users] using variable

[Tzafrir Cohen]

On Tue, Jun 27, 2006 at 12:13:31PM +0800, unplug wrote:
> Hi,
> 
>  How can I access the variable in marco?  Say, there is a dial plan
> below.  In line 4, it will show the variable FOO=1234.  However, the
> variable in line 2 is nothing.  Can I assign a varilable in macro and
> access it outside a macro in the same session only?

Yes. 

> 
> 
> [dialplan]
> 1: exten => 1234,1,Macro(test)
> 2: exten => 1234,2,NoOp(${FOO})
> 
> [macro-test]
> 3: exten => s,1,Set(FOO=1234)
> 4: exten => s,2,NoOp(${FOO})

-- 
Tzafrir Cohen  sip:[EMAIL PROTECTED]
icq#16849755   iax:[EMAIL PROTECTED]
+972-50-7952406   
[EMAIL PROTECTED]  http://www.xorcom.com
___
--Bandwidth and Colocation provided by Easynews.com --

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

[Asterisk-Users] using variable

[unplug]

Hi,

 How can I access the variable in marco?  Say, there is a dial plan
below.  In line 4, it will show the variable FOO=1234.  However, the
variable in line 2 is nothing.  Can I assign a varilable in macro and
access it outside a macro in the same session only?


[dialplan]
1: exten => 1234,1,Macro(test)
2: exten => 1234,2,NoOp(${FOO})

[macro-test]
3: exten => s,1,Set(FOO=1234)
4: exten => s,2,NoOp(${FOO})
___
--Bandwidth and Colocation provided by Easynews.com --

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

RE: [Asterisk-Users] Question about ring groups and ext. busy in call

[Cullin J. Wible]

There is a difference between call waiting (a single line 
with multiple call instances) and a multi-line phone.
 
We use a Polycom phone that has 3 lines with a single call 
instance per line. The configuration is set so that once a line has a single 
call it will return busy.
 
Therefore, we have use the local channels to acheive the 
functionality that you are looking for as outlined below. It results in a number 
of macro's being run in parallel, but as long as you have enough horse power it 
shouldn't be a problem.
 
Hope that helps.
 
 

Cullin J. 
Wible
Co-Founder & 
CTO
Email Data Source, 
Inc.
212-514-8900 
x1006
 
 
 
 
exten => 
all,1,Dial(local/SIP-0004f2026a53&local/SIP-0004f2035e5f)
 
exten => 
SIP-0004f2026a53,1,Macro(exten-chain,SIP/0004f2026a53-1,SIP/0004f2026a53-2,SIP/0004f2026a53-3)
exten => 
SIP-0004f2035e5f,1,Macro(exten-chain,SIP/0004f2035e5f-1,SIP/0004f2035e5f-2,SIP/0004f2035e5f-3)
 
;; Ring a chain of two devices (no 
voicemail):;   ${ARG1} - Device(s) to ring;   
${ARG2} - Device(s) to ring (when busy);   ${ARG3} - Device(s) to 
ring (when busy);[macro-exten-chain]exten => 
s,1,Playtones(ring)exten => s,2,Dial(${ARG1}, 30, 
r)   
; do the callexten => 
s,3,Goto(207)  
; error, busy
 
exten => s,103,GotoIf($[ "${ARG2}" != "" ]?104:207); try 
the 2nd deviceexten => s,104,Dial(${ARG2}, 30, 
r) 
; do the callexten => 
s,105,Playtones(busy)  
; play busyexten => 
s,106,Busy()   
; mark busy
 
exten => s,205,GotoIf($[ "${ARG3}" != "" ]?206:207); try 
the 3rd deviceexten => s,206,Dial(${ARG3}, 30, 
jr)    
; do the callexten => 
s,207,Playtones(busy)  
; play busyexten => 
s,208,Busy()   
; mark busy
 
exten => i,1,Playtones(busy)exten => 
i,2,Busy()
 
exten => t,1,Playtones(busy)exten => 
t,2,Busy()


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Chris 
SuttonSent: Monday, June 26, 2006 11:15 PMTo: 
asterisk-users@lists.digium.comSubject: [Asterisk-Users] Question 
about ring groups and ext. busy in call


I have a ring group set up with 3 
extensions… we’ll use 14, 15 and 16.
 
When a call comes in, it rings all 
three extensions.  If one particular extension already is on the phone, it 
completely skips that phone and only rings the other 2.  Example to 
explanation sake is:
 
Call comes in, ext. 14 is already in 
the middle of a call, 15 and 16 will ring normally, but 14 does not have any 
indication that another call came in.
 
What I’m trying to accomplish 
is:
 
I would like the ring group to 
always ring all 3 phones, even if one is on the phone.  Similar to call 
waiting I guess… on the “multi-line” phones, it could ring line 2 or 3 or which 
ever line is available if that phone is already in 
progress.
 
This is necessary because some times 
there is only 1 person in the office and may not always be able to hear the 
other phones ringing… 
 
Call pick up is not what I’m looking 
for.. (mainly because again, the person may not hear/know the other phones are 
ringing).
 
Thank you for any 
help!
 
Chris
 
___
--Bandwidth and Colocation provided by Easynews.com --

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

RE: [Asterisk-Users] best hardphone for Asterisk?

[Cullin J. Wible]

We've used a number of the polycom 301 and 501 phones in our office.

We have also deployed a dozen of the Linksys SPA-1001 single-line FXS
adapters using G726, SIP, NAT and STUN. They are extremely reliable and easy
to deploy - $60-$70 US each.

We tested a number of IAX hard phones and didn't find anything that was
reliable and/or suitable for our corporate setting. We really wanted to run
IAX for remote users, but eventually decided that SIP/STUN was easier to
support.

We also tested the IAXy device and found that it's inability to use DNS
resolution, only be configured on Linux, and only run ulaw/alaw made and
that it cost more then the SPA-1001, which can use DNS, G726/G729 and has
web-based configuration for less money the more attractive option.

We also tested the IAX hard phone made by AT-COM only to find that a number
of features such as call transfer do not work.

For home/remote users: setup STUN, and use a SPA-1001. For a corporate
setting I highly recommend the Polycom phones.

Cheers,

Cullin J. Wible
Co-Founder & CTO
Email Data Source, Inc.
212-514-8900 x1006
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Doug Crompton
Sent: Monday, June 26, 2006 11:49 PM
To: Iain Barker
Cc: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] best hardphone for Asterisk?

Iain,

 Thanks for the repsonse but you are kidding me right? From what I can see
if I bought this phone and two remotes my outlay would be close to $800 US.
This is NOT a home device unless you have nothing better to do with your
money!

You can buy a lot of single line wireless phones and FXS devices for that
amount!

Doug

On Mon, 26 Jun 2006, Iain Barker wrote:

> Doug,
>
> What you are describing sounds like the Aastra 480-CT, a base 
> Ethernet/SIP screenphone supporting multiple wireless handsets [but as 
> this is a non-commercial list I won't go into more detail here, google 
> for the above model number if you're interested in more info.]
>
> - Iain
>
> ---
> Message: 4
> Date: Mon, 26 Jun 2006 00:08:48 -0400 (EDT)
> From: Doug Crompton <[EMAIL PROTECTED]>
> Subject: RE: [Asterisk-Users] best hardphone for Asterisk?
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>   
> Message-ID:
>   <[EMAIL PROTECTED]>
> Content-Type: TEXT/PLAIN; charset=US-ASCII
>
> Still awfully pricey for home use and the styling is not there for a 
> bedroom or many other areas of a modern home. What we need is a 
> wireless sip phone modeled like the panasonic or uniden which allow 
> multiple extension off of one base. The base would connect to the 
> internet. The other problem is many of these phones require power, so 
> even if you have backup for your central system the phone still needs 
> to be on it. Power over ethernet would help.
>
> Doug
>


"Those that sacrifice essential liberty to obtain a little temporary safety
deserve neither liberty nor safety."  -- Ben Franklin (1759)


*  Doug Crompton   *
*  Richboro, PA 18954  *
*  215-431-6307*
*  *
* [EMAIL PROTECTED]*
* http://www.crompton.com  *



___
--Bandwidth and Colocation provided by Easynews.com --

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

___
--Bandwidth and Colocation provided by Easynews.com --

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

RE: [Asterisk-Users] best hardphone for Asterisk?

[Doug Crompton]

Iain,

 Thanks for the repsonse but you are kidding me right? From what I can see
if I bought this phone and two remotes my outlay would be close to $800
US. This is NOT a home device unless you have nothing better to do with
your money!

You can buy a lot of single line wireless phones and FXS devices for that
amount!

Doug

On Mon, 26 Jun 2006, Iain Barker wrote:

> Doug,
>
> What you are describing sounds like the Aastra 480-CT, a base Ethernet/SIP 
> screenphone supporting multiple wireless handsets [but as this is a 
> non-commercial list I won't go into more detail here, google for the above 
> model number if you're interested in more info.]
>
> - Iain
>
> ---
> Message: 4
> Date: Mon, 26 Jun 2006 00:08:48 -0400 (EDT)
> From: Doug Crompton <[EMAIL PROTECTED]>
> Subject: RE: [Asterisk-Users] best hardphone for Asterisk?
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>   
> Message-ID:
>   <[EMAIL PROTECTED]>
> Content-Type: TEXT/PLAIN; charset=US-ASCII
>
> Still awfully pricey for home use and the styling is not there for a
> bedroom or many other areas of a modern home. What we need is a wireless
> sip phone modeled like the panasonic or uniden which allow multiple
> extension off of one base. The base would connect to the internet. The
> other problem is many of these phones require power, so even if you have
> backup for your central system the phone still needs to be on it. Power
> over ethernet would help.
>
> Doug
>


"Those that sacrifice essential liberty to obtain a little temporary safety
 deserve neither liberty nor safety."  -- Ben Franklin (1759)


*  Doug Crompton   *
*  Richboro, PA 18954  *
*  215-431-6307*
*  *
* [EMAIL PROTECTED]*
* http://www.crompton.com  *



___
--Bandwidth and Colocation provided by Easynews.com --

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

[Asterisk-Users] Question about ring groups and ext. busy in call

[Chris Sutton]

I have a ring group set up with 3 extensions… we’ll
use 14, 15 and 16.

 

When a call comes in, it rings all three extensions.  If one
particular extension already is on the phone, it completely skips that phone
and only rings the other 2.  Example to explanation sake is:

 

Call comes in, ext. 14 is already in the middle of a call,
15 and 16 will ring normally, but 14 does not have any indication that another
call came in.

 

What I’m trying to accomplish is:

 

I would like the ring group to always ring all 3 phones,
even if one is on the phone.  Similar to call waiting I guess… on the “multi-line”
phones, it could ring line 2 or 3 or which ever line is available if that phone
is already in progress.

 

This is necessary because some times there is only 1 person
in the office and may not always be able to hear the other phones ringing…


 

Call pick up is not what I’m looking for.. (mainly
because again, the person may not hear/know the other phones are ringing).

 

Thank you for any help!

 

Chris

 






___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [Asterisk-Users] Testing a FastAGI script

[didier]

Bonjour,

En attendant la naturalisation Espagnole, es tu rentré dans les arcanes 
des DEADAGI ? (qui dit Fastagi dit un peu.)

J'avoue ne pas etre sur des limites ou non de ces bestioles.
-Juste une AGI qui ne s'interromp pas sur hangup ?
-une AGI qui permet d'accèder aux $var d'un channel meme en cas de hangup?
-Les 2 ? (probable)
- ou encore un peu plus mais là, euh, peu documenté et même les sources.

on l'utilise sur l'extension h, ok et même ailleurs si l'on veut, 
mais bon, hum, des bribes à droite à gauche (interagit avec le cdr pour 
ne pas effacer)


Bon, on ne sait jamais, un francophone à peut être des vues plus claires 
que les miennes (parce que les explications anglophones, parfois 
'intraduisibles' )


Bonne journée jusqu'à 20 h

Didier









Olivier a écrit :


Hi,

(I tried to post this message a week ago but I don't think it could 
reach the list. Please forgive me if you already received it).


I would like to develop my first FastAGI script.
I would like to test it independently from Asterisk for the sake of 
simplicity.


Which linux (or cygwin) tool is the best for that ?
Using this tool, I will open a FastAGI connection, throw data in and 
read data from.


With AGI script, echo or cat commands are enough.
But what are the simplest ones for FastAGI ?

Regards



___
--Bandwidth and Colocation provided by Easynews.com --

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



___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [Asterisk-Users] SRST type functionality

[Bruce Reeves]

We looked at this before going to * as our main system at the corporate headquarters. We have not rolled * out to our other sites yet, but in our case each site has local phone lines and wan contectivity so we are planing a full pbx at each site and then trunking. In theory you could dial the remote site via your prefered trunk, like IAX and if you detected a channel unavaliable message have a backup route in the dial plan like a DID into the system somewhere. Like I said we did not plan on using the main * servers at corporate to handle all calling.
On 6/26/06, Curt Shaffer <[EMAIL PROTECTED]> wrote:















Has anyone out there figured out how to emulate the Cisco
SRST functionality with *? If so would you mind letting me know the best
practices for this?





 Thanks



Curt







___--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users
-- BruceNortex Networks
___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [Asterisk-Users] Re: fail to make call

[unplug]

Thanks for your reply.  As you said, my expectation is wrong in the
current release.  How can I implement load balancing using multiple
asterisk of current release
with a centralized database in another way?  Is it possible?  Any
clue?  Thanks!

On 6/23/06, Kevin P. Fleming <[EMAIL PROTECTED]> wrote:

- unplug <[EMAIL PROTECTED]> wrote:
> BTW, do you mean this function will be included in next release?
> When
> will be the next release available?

No, it will not be in the next release (which is Asterisk 1.4). It may be in 
Asterisk 1.6, scheduled for January or so of 2007, but as I said in my other 
reply, even with the database support set up properly, there are still 
significant problems with operating this way, depending on how the network is 
structured.

--
Kevin P. Fleming
Senior Software Engineer
Digium, Inc.

___
--Bandwidth and Colocation provided by Easynews.com --

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


___
--Bandwidth and Colocation provided by Easynews.com --

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

[Asterisk-Users] SRST type functionality

[Curt Shaffer]

Has anyone out there figured out how to emulate the Cisco
SRST functionality with *? If so would you mind letting me know the best
practices for this?

 

 

Thanks

 

Curt






___
--Bandwidth and Colocation provided by Easynews.com --

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

[Asterisk-Users] x100p buying advice

[Rod Morison]

I'm looking to get an x100p off ebay and am not particularly familar 
with the "life cycle" of the card.. An "Authentic X100P" listing has a 
buy it now of $29.95 and says


There are 3 types of cards Asterisk would recognize:
*Screenshots from the official, original driver install
   Cheap "OEM X100P","Clones", "Compatibles", Knock-Offs
  Found a Wildcard FXO: Generic Clone
   The X101P (note the 101, not 100) is a Low-end version of X100P 
which uses low grade chips

  Found a Wildcard FXO: X101P
   Authentic, Original X100P Speaks for Itself!
  Found a Wildcard FXO: X100P


From what I gather clone's and knockoffs will have trouble with 
callerid. Is the "Found a Wildcard FXO: X100P" enough to establish full 
featured hardware (assuming an honest seller)? Is there another 
recommended source besides ebay in this price range?


Thanks


___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [Asterisk-Users] ASTCC: How to reset periodically all "card in use" flag back?

[JP Carballo]

Ronald Wiplinger wrote:


Nicolás Gudiño wrote:


You should install php-pcntl (or compile php to add support for
process control functions). The inuse problem will be fixed then.

Regards,



Can you please give us more info about that?
What is php-pcntl? What should it do? How can it be used to be a 
solution?



Nicolas already said and I quote:

" Replying to myself... I was thinking on a2billing, not astcc, so
php-pcntl will make no difference."

ASTCC uses Perl-AGI. That should have clued you in.

--
JP Carballo

http://www.netfone2x.com
Bringing the world closer.

It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. 


___
--Bandwidth and Colocation provided by Easynews.com --

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

[Asterisk-Users] M() option to Dial

[Eric \"ManxPower\" Wieling]

I'm using the M() option to Dial() and having problems.  In the 
following dialplan example ANY digit exits the macro.  When the callee 
presses 1 the Noop(Reset AbsoluteTimeout(0)) does not get run.  Does 
anyone have any ideas as to what I'm doing wrong?  Asterisk 1.2.x


[extensions]
exten => 2998,1,Dial(Zap/1/5551212,,wM(answer-confirmation^20))

[macro-answer-confirmation]
exten => s,1,Noop(Set AbsoluteTimeout(${ARG1})
exten => s,n,Background(/etc/asterisk/call-from-campground)
exten => s,n,Goto(2)

exten => 1,1,Noop(Reset AbsoluteTimeout(0))

--
Now accepting new clients in Birmingham, Atlanta, Huntsville, 
Chattanooga, and Montgomery.

___
--Bandwidth and Colocation provided by Easynews.com --

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

RE: [Asterisk-Users] SE Michigan asterisk users group

[Carlos Alperin]

OK,

This is what I get till now:

Ron Kushner Sterling Heigths[EMAIL PROTECTED]
Rusty DekemaAnn Arbor   [EMAIL PROTECTED]
Jon Radon   Southfield  [EMAIL PROTECTED]
Steven BerkHolz [EMAIL PROTECTED]
Tom Hayden  Livonia [EMAIL PROTECTED]
Michael George  Lansing [EMAIL PROTECTED]
Tim Sharp   Livonia [EMAIL PROTECTED]
Bradley Watkins ??  [EMAIL PROTECTED]
albeit  ??  [EMAIL PROTECTED]
Carlos Alperin  Southfield  [EMAIL PROTECTED]

If you guys want to start planning on how to start, then is time to start
exchanging direct mails with more info
In order to arrange future steps.

I can create the mailing list, for that I can use my own domain
calperin.com, or we going to need to register a domain on Network Solutions

Waiting for suggestions,

Regards,

Carlos Alperin

___
--Bandwidth and Colocation provided by Easynews.com --

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

[Asterisk-Users] 1.2.9.1 SIP/Local/Queue behaviours weird

[Isaac Xiao]

Our phone all Polycom phone and we use *'s transfer function rather than
phone's one. We also has canreinvite=no. I believe that it is something
wrong with Call Bridge between two channels(ZAP/SIP/Local). Before we
didn't disable autofallthrough (default is yes), we also experienced
call drop.

>I have seen this when Polycom has to communicate with none polycom
>phones and a transfer is initiated to a polycom, unless the Polycom
>presses Hold and then unhold, there is only one way audio, this is
>without NAT involved. There might also be other cases when this
>happens. My workaround is to add canreinvite=no

Isaac Xiao
___
--Bandwidth and Colocation provided by Easynews.com --

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

RE: [Asterisk-Users] DUNDi Not Able to HandleComplexFailoverSituations

[Patrick]

On Mon, 2006-06-26 at 15:33 -0700, Michael Collins wrote:
> > > I get annoyed Stephen when Digium goes around calling Asterisk
> > 'enterprise grade', which in my opinion it really isn't. I'd consider
> > distributed ACD queues to be a requirement for an enterprise grade
> > product, but it's becoming apparent that there is no mechanism for
> > implementing this. I'm being told that DUNDi isn't the right man for
> the
> > job.
> > 
> > I'd suggest you ask Digium for your money back.
> > 
> > Leif.
> 
> 
> Question: isn't there a bigger picture issue here?  I've seen a lot of
> bashing going on in this thread but not very much useful dialogue.
> (Doug bashing DUNDi and Digium, other people bashing Doug for his
> "annoying" posts, etc.)  
> 
> Whether or not we like or dislike Doug's tone is, IMHO, irrelevant.  How
> about we tackle the REAL questions: If DUNDi isn't the answer, what is?
> Is Asterisk even capable of doing what Doug needs, namely, distributed
> ACD queues?  If so, how?  If not, why?  Is it even feasible to try to do
> it?  Will it require an Asterisk add-on, or can the core be modified to
> do this? (This leads to the question for the dev list...)
> 
> These questions, of course, lead to other questions: If * can be
> programmed to do distributed ACD queues, does that mean there are other
> features that might benefit from a "distributed" model?  Etc., etc.  I'm
> just throwing out ideas because maybe one of these ideas can turned into
> a killer app, just like Asterisk itself.  Just think of the advantage
> you would have if you wanted to sell Asterisk against one of the big
> boys.  How much would a fully redundant, HA Asterisk system cost
> compared to the same thing by Cisco, Avaya, Nortel, NEC...  You get the
> idea.  
> 
> The moral of this post: a little good-natured bashing is just fine, but
> let's not lose sight of the ultimate goal, which is to keep making
> Asterisk a better product.

Nod. There have been many posts on the mailinglist about redundancy/HA
but I have never seen a detailed explanation how to create a fully
redundant HA Asterisk system. Would be nice if that recent DUNDi article
posted on the list were expanded by Digium and/or the community with
some of this magic. There's "clustering & scaling" mentioned at
http://edvina.net/training/schedule.shtm (day 5 at the end) so I guess
it's possible but you have to pay a few thousand euros to get the
info :)

Regards,
Patrick



___
--Bandwidth and Colocation provided by Easynews.com --

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

RE: [Asterisk-Users] best hardphone for Asterisk?

[Iain Barker]

Doug,

What you are describing sounds like the Aastra 480-CT, a base Ethernet/SIP 
screenphone supporting multiple wireless handsets [but as this is a 
non-commercial list I won't go into more detail here, google for the above 
model number if you're interested in more info.]

- Iain

---
Message: 4
Date: Mon, 26 Jun 2006 00:08:48 -0400 (EDT)
From: Doug Crompton <[EMAIL PROTECTED]>
Subject: RE: [Asterisk-Users] best hardphone for Asterisk?
To: Asterisk Users Mailing List - Non-Commercial Discussion

Message-ID:
<[EMAIL PROTECTED]>
Content-Type: TEXT/PLAIN; charset=US-ASCII

Still awfully pricey for home use and the styling is not there for a
bedroom or many other areas of a modern home. What we need is a wireless
sip phone modeled like the panasonic or uniden which allow multiple
extension off of one base. The base would connect to the internet. The
other problem is many of these phones require power, so even if you have
backup for your central system the phone still needs to be on it. Power
over ethernet would help.

Doug
___
--Bandwidth and Colocation provided by Easynews.com --

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

[Asterisk-Users] Microsoft unified communications

[Dean Collins]

http://www.eweek.com/article2/0,1895,1981625,00.asp

 

Unified
messaging in Exchange Server 2007,
expected to ship in late 2006 or early 2007, will deliver a unified in-box
experience that includes e-mail, voice mail and faxing functionality, as well
as new capabilities such as speech-based auto attendant allowing users to
access their communications from any phone. (Click here to read an interview with Raikes on the
company's approach to the emerging market for real-time business
communications.)

 

 

I hope we can get integration into
Asterisk as I’d like to reply to emails from my handset from time to
time.

 



Regards,

 

 

Dean Collins

Cognation Pty Ltd

[EMAIL PROTECTED]

+1-212-203-4357 Ph

+1-917-207-3420 Mb

+61-2-9016-5642 (Sydney
in-dial).



 






___
--Bandwidth and Colocation provided by Easynews.com --

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

RE: [Asterisk-Users] GXP-2000 and Shared Line Appearances

[shadowym]

Thanks Dustin,

Can you give an example of how this would be used when a call comes in?

How is this different from the existing call parking feature in Asterisk?
Looks very similar except the existing Asterisk call parking automatically
assigns an extension and then announces it to the person who parked the
call.

Features.conf file included with Freepbx 2.1.1 using Asterisk 1.2.9.1

;
; Sample Parking configuration
;

[general]
parkext => 70   ; What ext. to dial to park
parkpos => 71-79; What extensions to park calls on
context => parkedcalls  ; Which context parked calls are in
;parkingtime => 60  ; Number of seconds a call can be
parked for (default is 45 seconds)

[featuremap]
;blindxfer => ##; Blind Transfer
;disconnect => **   ; Disconnect Call
automon => *1   ; One Touch Record
;atxfer => *2   ; Attended Xfer

> -Original Message-
> From: Dustin Wildes [mailto:[EMAIL PROTECTED] 
> Sent: Monday, June 26, 2006 11:55 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] GXP-2000 and Shared Line Appearances
> 
> Daniel Salama wrote:
> 
> > Dustin,
> >
> > any updates on this?
> >
> > Thanks,
> > Daniel
> >
> Hey Daniel!
> Yes - just posted the link.
> I appologize for the delay.
> 
> Here's the link to the forum as well, if anyone is 
> interested. This should compile and run on Asterisk-1.2.4 and higher.
> http://www.vecsector.com/phonecall/valet/
> 
> Enjoy!
> 
> 
> Dustin Wildes
> VecSector, LLC
> 1.912.422.7082 x101
> 
> 
> 
___
--Bandwidth and Colocation provided by Easynews.com --

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

RE: [Asterisk-Users] DUNDi Not Able to HandleComplexFailoverSituations

[Michael Collins]

> > I get annoyed Stephen when Digium goes around calling Asterisk
> 'enterprise grade', which in my opinion it really isn't. I'd consider
> distributed ACD queues to be a requirement for an enterprise grade
> product, but it's becoming apparent that there is no mechanism for
> implementing this. I'm being told that DUNDi isn't the right man for
the
> job.
> 
> I'd suggest you ask Digium for your money back.
> 
> Leif.


Question: isn't there a bigger picture issue here?  I've seen a lot of
bashing going on in this thread but not very much useful dialogue.
(Doug bashing DUNDi and Digium, other people bashing Doug for his
"annoying" posts, etc.)  

Whether or not we like or dislike Doug's tone is, IMHO, irrelevant.  How
about we tackle the REAL questions: If DUNDi isn't the answer, what is?
Is Asterisk even capable of doing what Doug needs, namely, distributed
ACD queues?  If so, how?  If not, why?  Is it even feasible to try to do
it?  Will it require an Asterisk add-on, or can the core be modified to
do this? (This leads to the question for the dev list...)

These questions, of course, lead to other questions: If * can be
programmed to do distributed ACD queues, does that mean there are other
features that might benefit from a "distributed" model?  Etc., etc.  I'm
just throwing out ideas because maybe one of these ideas can turned into
a killer app, just like Asterisk itself.  Just think of the advantage
you would have if you wanted to sell Asterisk against one of the big
boys.  How much would a fully redundant, HA Asterisk system cost
compared to the same thing by Cisco, Avaya, Nortel, NEC...  You get the
idea.  

The moral of this post: a little good-natured bashing is just fine, but
let's not lose sight of the ultimate goal, which is to keep making
Asterisk a better product.

-MC
___
--Bandwidth and Colocation provided by Easynews.com --

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

RE: [Asterisk-Users] Oh oh. Micro$oft just noticed VoIP

[shadowym]

They have been talking about this for awhile.  If you look at the real time
and embedded operating system world they have not really done so well over
the many years they have been trying. Just throwing money at the problem has
never worked for them in the past either.  

The Asterisk community has nothing to worry about in the near term if ever
IMHO.

> -Original Message-
> From: Brian Capouch [mailto:[EMAIL PROTECTED] 
> Sent: Monday, June 26, 2006 10:16 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Oh oh. Micro$oft just noticed VoIP
> 
> It will be interesting to see how many standards get broken, 
> and how many proprietary hooks get thrown into the pot.  The 
> bean counters smell some money, and their OS franchise is waning:
> 
> http://www.nytimes.com/2006/06/26/technology/26soft.html
> 
> --
> This message has been scanned for viruses and dangerous 
> content by MailScanner, and is believed to be clean.
> 
> 
> 
___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [Asterisk-Users] asterisk-stat display problems

[Mojo with Horan & Company, LLC]

Chris, on redhat-flavored systems this log Julian mentioned could be any 
of the following filenames under /var/log/httpd/:


error_log
ssl_error_log if this was an ssl connection
www.domain.com-error_log if this was a name-based virtual host and some 
semblence of sanity was used for the log file names :)


Moj

Julian J. M. wrote:

Check /var/log/http/error.log

Usually, asterisk-stat fails because it tries to use more memory than
allowed in php.ini.

Julian J. M.

On 6/26/06, Chris Earle (CBL) <[EMAIL PROTECTED]> wrote:

yep

I don't know exactly which things the php-gd is used for, but like I said,
someof the pages work, like the main record page, the little red bars
showing call volume work fine


Really annoying, cause it looks so good at that point, then you go to use
the other pages/features and it's broken

Thanks for the reply,

--
Chris



- Original Message -
From: "Mojo with Horan & Company, LLC" <[EMAIL PROTECTED]>
To: "Chris Earle (CBL)" <[EMAIL PROTECTED]>; "Asterisk Users Mailing List -
Non-Commercial Discussion" 
Sent: Monday, June 26, 2006 1:49 PM
Subject: Re: [Asterisk-Users] asterisk-stat display problems



do you have the php-gd package installed on your * server?

Chris Earle (CBL) wrote:

Hey all,

having a terrible time with asterisk-stat -- it runs, server is fine,

but

some of the pages don't display properly/at all --- I think this is a

code

problem with them, but not sure.  I thought everyone loved the

asterisk-stat

package?

See below problems.  Any ideas?  Areski hasn't replied to me since

--
Chris


- Original Message -
From: "Chris Earle (CBL)"
To: "Areski"
Sent: Tuesday, June 13, 2006 6:15 PM
Subject: Re: CDR-Analyser version question



Thank you for the reply;

I see now that the main file cdr.php does work with argument ?s=1, 2,
etc
but when s=0, does not load

I get an Apache error :

 relocation error: /usr/lib/php4/20020429/gd.so: undefined symbol:
gdFontCacheShutdown

Not sure if that means anything important;




Also, in the new Asterisk-Stat feature pages like Calls Compare (s=2),

the

pages do not complete their output -- no search button displayed, stops
outputting radio buttons for UserField row etc

So at this point, only the main Call-log page (s=1) works.


I am using Debian with php 4.4.1
Mysql ver 12.22, Distrib 4.0.24
GD Library is 2.0.33 I think


Any input you can pass along would be much appreciated!  I am

comfortable

with php so if you want me to modify sourcecode that is fine

Thanks!




- Original Message -
From: "Areski"
To: "Chris Earle (CBL)"
Sent: Sunday, May 28, 2006 7:11 PM
Subject: Re: CDR-Analyser version question



No there is no asterisk requirement to make asterisk-stat.
Indeed the soft is only based on the cdr database. If you have an

error

you can give me more info, I may help you.

Rgds, Areski

On 5/25/06, Chris Earle (CBL) wrote:

Hi there,

quick question:

Does asterisk-stat v2.0.1 require Asterisk 1.2+ ?  I am using

Asterisk

1.0.x

and can't get it to load the cdr.php properly

so I downgraded to v1.3 and it works...

Let me know if there's an asterisk version requirement for each

version

of

the CDR Analyser

Thanks!



--
Chris Earle







--
Mojo <[EMAIL PROTECTED]>
Office Manger, Horan & Company, LLC
(907) 747- x112

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


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

___
--Bandwidth and Colocation provided by Easynews.com --

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


___
--Bandwidth and Colocation provided by Easynews.com --

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

!DSPAM:500,44a0511e207772014189408!



--
Mojo <[EMAIL PROTECTED]>
Office Manger, Horan & Company, LLC
(907) 747- x112
___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [Asterisk-Users] ASTCC: How to reset periodically all "card in use" flag back?

[Ronald Wiplinger]

Nicolás Gudiño wrote:

Hi Ronald,

If a user calls and hangs up before the destination party rings, than
the in-use flag remains set! This is one case, but maybe there are many
other cases.


You should install php-pcntl (or compile php to add support for
process control functions). The inuse problem will be fixed then.

Regards,



Can you please give us more info about that?
What is php-pcntl? What should it do? How can it be used to be a solution?


bye

Ronald Wiplinger
___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [Asterisk-Users] SE Michigan asterisk users group

[RE Kushner List Account]

Carlos Alperin wrote:


I live in Southfield, our main office is in Pontiac, but our Colo is in
Southfield.

 



I'm here in Sterling Heights, have a call center in Clinton Twp that's 
100% Cisco/Linksys phones (7940s and SPA942s) and rent in a colo down in 
Southfield as well where I connect to the PSTN and run other type of 
calling services based on Asterisk. I've been working on building 
diskless Asterisk servers to improve reliability in some of my applications.


I've been in all kinds of user groups in the past, MCUG, AACS, WCAU, 
etc, as well as First Tuesdays. It was through these groups I have 
gained quite a few contacts at Michigan Bell/Ameritech/SBC/AT&T and 
other fortune 500 companies.


Depends on where an Asterisk user group meets, I might be interested.

-Ron

___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!

[Jean-Michel Hiver]

Ja dat kun je wel zeggen ja... Maar goed dat Nederlanders vrij aardig
Engels praten!
 ;-)



Pues my punto fue que un poquito de correo en otro idioma no hace 
daño, y si ayuda mucho y molesta poco, ¿por qué quejarse?


Quel bordel, sacrebleu!

--
Jean-Michel Hiver - http://ykoz.net/
Découvrez la Réunion des Technologies IP & Telecom
TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE

___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [Asterisk-Users] asterisk-stat display problems

[Julian J. M.]

Check /var/log/http/error.log

Usually, asterisk-stat fails because it tries to use more memory than
allowed in php.ini.

Julian J. M.

On 6/26/06, Chris Earle (CBL) <[EMAIL PROTECTED]> wrote:

yep

I don't know exactly which things the php-gd is used for, but like I said,
someof the pages work, like the main record page, the little red bars
showing call volume work fine


Really annoying, cause it looks so good at that point, then you go to use
the other pages/features and it's broken

Thanks for the reply,

--
Chris



- Original Message -
From: "Mojo with Horan & Company, LLC" <[EMAIL PROTECTED]>
To: "Chris Earle (CBL)" <[EMAIL PROTECTED]>; "Asterisk Users Mailing List -
Non-Commercial Discussion" 
Sent: Monday, June 26, 2006 1:49 PM
Subject: Re: [Asterisk-Users] asterisk-stat display problems


> do you have the php-gd package installed on your * server?
>
> Chris Earle (CBL) wrote:
> > Hey all,
> >
> > having a terrible time with asterisk-stat -- it runs, server is fine,
but
> > some of the pages don't display properly/at all --- I think this is a
code
> > problem with them, but not sure.  I thought everyone loved the
asterisk-stat
> > package?
> >
> > See below problems.  Any ideas?  Areski hasn't replied to me since
> >
> > --
> > Chris
> >
> >
> > - Original Message -
> > From: "Chris Earle (CBL)"
> > To: "Areski"
> > Sent: Tuesday, June 13, 2006 6:15 PM
> > Subject: Re: CDR-Analyser version question
> >
> >
> >> Thank you for the reply;
> >>
> >> I see now that the main file cdr.php does work with argument ?s=1, 2,
> >> etc
> >> but when s=0, does not load
> >>
> >> I get an Apache error :
> >>
> >>  relocation error: /usr/lib/php4/20020429/gd.so: undefined symbol:
> >> gdFontCacheShutdown
> >>
> >> Not sure if that means anything important;
> >>
> >>
> >>
> >>
> >> Also, in the new Asterisk-Stat feature pages like Calls Compare (s=2),
the
> >> pages do not complete their output -- no search button displayed, stops
> >> outputting radio buttons for UserField row etc
> >>
> >> So at this point, only the main Call-log page (s=1) works.
> >>
> >>
> >> I am using Debian with php 4.4.1
> >> Mysql ver 12.22, Distrib 4.0.24
> >> GD Library is 2.0.33 I think
> >>
> >>
> >> Any input you can pass along would be much appreciated!  I am
comfortable
> >> with php so if you want me to modify sourcecode that is fine
> >>
> >> Thanks!
> >>
> >>
> >>
> >>
> >> - Original Message -
> >> From: "Areski"
> >> To: "Chris Earle (CBL)"
> >> Sent: Sunday, May 28, 2006 7:11 PM
> >> Subject: Re: CDR-Analyser version question
> >>
> >>
> >>> No there is no asterisk requirement to make asterisk-stat.
> >>> Indeed the soft is only based on the cdr database. If you have an
error
> >>> you can give me more info, I may help you.
> >>>
> >>> Rgds, Areski
> >>>
> >>> On 5/25/06, Chris Earle (CBL) wrote:
>  Hi there,
> 
>  quick question:
> 
>  Does asterisk-stat v2.0.1 require Asterisk 1.2+ ?  I am using
Asterisk
> >> 1.0.x
>  and can't get it to load the cdr.php properly
> 
>  so I downgraded to v1.3 and it works...
> 
>  Let me know if there's an asterisk version requirement for each
> > version
> >> of
>  the CDR Analyser
> 
>  Thanks!
> 
> 
> 
>  --
>  Chris Earle
> 
> 
> > 
> >
> >
>
> --
> Mojo <[EMAIL PROTECTED]>
> Office Manger, Horan & Company, LLC
> (907) 747- x112
>
> --
> This message has been scanned for viruses and dangerous content by
> MailScanner, and is believed to be clean.


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

___
--Bandwidth and Colocation provided by Easynews.com --

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


___
--Bandwidth and Colocation provided by Easynews.com --

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

RE: [Asterisk-Users] AGI script can not print out error message toconsole

[Douglas Garstang]

> -Original Message-
> From: Moises Silva [mailto:[EMAIL PROTECTED]
> Sent: Monday, June 26, 2006 2:44 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] AGI script can not print out 
> error message
> toconsole
> 
> 
> what do you mean by "could not print out message to stderr"???
> 
> Try being more descriptive about your problem. Error messages, how
> have you tried etc.
> 
> On 6/26/06, Zichao Wu <[EMAIL PROTECTED]> wrote:
> >
> > Hi, guys, I used  /usr/src/asterisk/agi/eagi-test.c script 
> to test AGI API,
> > but that script could not print out message to stderr.
> >
> > any ideas?

He may be referring to the fact that when you run asterisk in non-console mode, 
stderr goes nowhere (not even /var/log/asterisk/messages). Considering that in 
a production environment, your going to want to run it like this, it means that 
if, say, an AGI script encounters a syntax error, you can't see what the 
problem was, unless you shut asterisk run, re-run it in console mode, debug, 
and restart it again. Not very convenient!

Doug.
___
--Bandwidth and Colocation provided by Easynews.com --

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

[Asterisk-Users] "Say" Applications fail

[Jon Mosier]

All of the Asterisk "Say" applications have stopped working.

Example: SayDigits(), SayNumber(), etc...

CLI output:

-- Executing SayDigits("SIP/209.247.17.5-b7901508", "12356") in new  
stack
  == Spawn extension (facloc-english, 12356, 2) exited non-zero on  
'SIP/209.247.17.5-b7901508'


This is driving me crazy.
___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!

[Josué Conti]

Hi All.   Please, we need to have more respect with the list.    Regards
Josué
2006/6/26, Francesco Peeters <[EMAIL PROTECTED]>:
On Mon, June 26, 2006 21:39, Brian Capouch said:> Francesco Peeters (Asterisk) wrote:>>>
 Ja dat kun je wel zeggen ja... Maar goed dat Nederlanders vrij aardig>> Engels praten!>>  ;-) Pues my punto fue que un poquito de correo en otro idioma no hace daño,
> y si ayuda mucho y molesta poco, ¿por qué quejarse?>> B.>Ningunas quejas aquí... Apenas una explicación en el 'netiquette'--FP___
--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:  
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [Asterisk-Users] AGI script can not print out error message to console

[Moises Silva]

what do you mean by "could not print out message to stderr"???

Try being more descriptive about your problem. Error messages, how
have you tried etc.

On 6/26/06, Zichao Wu <[EMAIL PROTECTED]> wrote:


Hi, guys, I used  /usr/src/asterisk/agi/eagi-test.c script to test AGI API,
but that script could not print out message to stderr.

any ideas?
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:

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






--
"Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org";
___
--Bandwidth and Colocation provided by Easynews.com --

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

[Asterisk-Users] AGI script can not print out error message to console

[Zichao Wu]

Hi, guys, I used  /usr/src/asterisk/agi/eagi-test.c script to test AGI API, but that script could not print out message to stderr.
 
any ideas? 
___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!

[Francesco Peeters]

On Mon, June 26, 2006 21:39, Brian Capouch said:
> Francesco Peeters (Asterisk) wrote:
>
>>
>>
>> Ja dat kun je wel zeggen ja... Maar goed dat Nederlanders vrij aardig
>> Engels praten!
>>  ;-)
>>
>
> Pues my punto fue que un poquito de correo en otro idioma no hace daño,
> y si ayuda mucho y molesta poco, ¿por qué quejarse?
>
> B.
>

Ningunas quejas aquí... Apenas una explicación en el 'netiquette'

--FP
___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!

[Ralph Liebessohn]

On 6/26/06, Josué Conti <[EMAIL PROTECTED]> wrote:
OK Marco, irei efetuar os testes.
Se você quiser, posso lhe ajudar no forum, estou a disposição.
Assim que você criar as contas avise para podermos já ir colaborando.
 
Saudações
 
JosuéThe differences of licenses are here: https://www.nch.com.au/cgi-bin/register.exe?software=uplink
The site only says that support is different.-- Ralph LiebessohnICQ: 74835911Skype: liebessohn
___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!

[Brian Capouch]

Francesco Peeters (Asterisk) wrote:




Ja dat kun je wel zeggen ja... Maar goed dat Nederlanders vrij aardig
Engels praten!
 ;-)



Pues my punto fue que un poquito de correo en otro idioma no hace daño, 
y si ayuda mucho y molesta poco, ¿por qué quejarse?


B.

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

___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [Asterisk-Users] GXP-2000 and Shared Line Appearances

[Daniel Salama]

Beautiful. Will test and give you comments.

Nice work.

- Daniel

On Jun 26, 2006, at 2:55 PM, Dustin Wildes wrote:


Daniel Salama wrote:


Dustin,

any updates on this?

Thanks,
Daniel


Hey Daniel!
Yes - just posted the link.
I appologize for the delay.

Here's the link to the forum as well, if anyone is interested. This  
should compile and run on Asterisk-1.2.4 and higher.

http://www.vecsector.com/phonecall/valet/

Enjoy!


Dustin Wildes
VecSector, LLC
1.912.422.7082 x101

___
--Bandwidth and Colocation provided by Easynews.com --

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


___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!

[Francesco Peeters (Asterisk)]

On Mon, June 26, 2006 20:06, Brian Capouch said:
> Tzafrir Cohen wrote:
>> On Mon, Jun 26, 2006 at 09:39:11AM -0300, Josué Conti wrote:
>>
>>>Marco, bom dia.
>>>Essa interligação entre o Skype e Asterisk, é feito atavés de um módulo
>>>externo?
>>>É freeware?
>>>Podemos seguir com o projeto Asterisk-PT?
>>
>>
>> English, please, folks.
>>
>
> Let them talk.  What's it hurt the rest of us?

It is more a question of netiquette... If you're on an English
mailinglist, you should speak English (Not attacking Josué and Marco, just
answering the question here). It is not only more productive (If you keep
to English, more people understand and can contribute to *and* profit from
the discussion), but speaking a different language not spoken by the
majority on list is generally considered akin whispering in company: not
quite rude, but also not-done...

> We have seen the wages of tortured English sometimes unleashed on the
> list.  If they're getting the job done, I say hit the "Delete" button
> and get on with your life.

You can hit the delete button for bad English too, you know!  ;-)

> If 80% of the list traffic were in foreign languages, then I would say
> we would have an issue.

Ja dat kun je wel zeggen ja... Maar goed dat Nederlanders vrij aardig
Engels praten!
 ;-)



-- 
F Peeters
  PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch
  2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0
  AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN
  2 Sweex HFC-PCI cards
___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!

[Josué Conti]

Sorry to all.
Speaking English only.
 
Regards
 
Josué 
2006/6/26, Marco Mouta <[EMAIL PROTECTED]>:
Sorry  to all,Now only English speaking :)Your translation was perfect.Thanks once more
On 6/26/06, Mike Fedyk <[EMAIL PROTECTED]> wrote:> Tzafrir Cohen wrote:> > On Mon, Jun 26, 2006 at 09:39:11AM -0300, Josué Conti wrote:> >
> >> Marco, bom dia.> >> Essa interligação entre o Skype e Asterisk, é feito atavés de um módulo> >> externo?> >> É freeware?> >> Podemos seguir com o projeto Asterisk-PT?
> >>> >> > English, please, folks.> >> >> I don't know Portuguese and my Spanish is terrible, but I understood> that Josue wanted to know if he needed any external modules.  Marco
> pointed him to the right place to get skype-to-sip and now they're going> to collaborate.>> So, please guys English please or you'll get more of my bad translations. ;)>> Mike
> ___> --Bandwidth and Colocation provided by Easynews.com -->> Asterisk-Users mailing list> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users>--Com os melhores cumprimentos,Marco Mouta___
--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:  
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [Asterisk-Users] GXP-2000 and Shared Line Appearances

[Dustin Wildes]

Daniel Salama wrote:


Dustin,

any updates on this?

Thanks,
Daniel


Hey Daniel!
Yes - just posted the link.
I appologize for the delay.

Here's the link to the forum as well, if anyone is interested. This 
should compile and run on Asterisk-1.2.4 and higher.

http://www.vecsector.com/phonecall/valet/

Enjoy!


Dustin Wildes
VecSector, LLC
1.912.422.7082 x101

___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [Asterisk-Users] '500 Internal Server' Error on SIP NOTIFY

[Ben Chennat]

Yes we have been getting this error message "'500 Internal Server' errors back from their Polycom IP-601 (normally IP address)".  Do not know why. Was able to regenerate the same issue some times, but not all the time. It is not consistent. 


 

Symptom:
If you have several phones online (10 extns) if for some reason all the phones start to sent the message because several people in the office are transferring and answering new calls and existing calls in a certain manor, after a while the Asterisk reboots, and if at that instance, if you have any lines on park or on hold, all those lines gets dropped, and then light gets stuck on the Polycom IP601 phone. The only way you could get rid of this light on the Polycom phone is by rebooting the phones where the lights are stuck (Almost all phones). 

 
Symptom regeneration:
It happens when a person is talking, then multiple calls come in and then the person tries to transfer the call to some one. If only one or two error message is coming from the IP601 it will not cause any problem. 

 

 

Solution:

We do not have any solutions for it yet. Hope that Asterisk or Polycom will come up with a solution/ Patch/ Firmware upgrade soon. If you do find a solution please let us know.

Thanks,

Ben K. Chennat

On 6/26/06, Doug Lytle <[EMAIL PROTECTED]> wrote:
Douglas Garstang wrote:> Is anyone getting '500 Internal Server' errors back from their Polycom phones when Asterisk sends a SIP NOTIFY message to them?
>Yes, for quite a while.  Happens for us, when you do a transfer via thePolycom's transfer button.  Doesn't seem to cause any problems though.Doug--Ben Franklin quote:"Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety."
___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [Asterisk-Users] STUN?

[Moises Silva]

please type in google.com:

STUN server ALG

The fourth result is a good and small explanation.

On 6/26/06, Martin Joseph <[EMAIL PROTECTED]> wrote:


On Jun 26, 2006, at 9:32 AM, Raymond Tant wrote:

> Hi all,
>
> Could someone point at resources for running Asterisk behind a
> firewall.
> STUN keeps coming up but, alas, I'm easily confused. J

STUN is just a way to discover the true address of a machine behind a
NAT.

Firewalls aren't really an issue per se,  other then needing to open
particular ports for asterisk to use. For example, udp port 4569 for
IAX2 traffic, and 5060 for SIP signaling, as well as ports in the
1-2 range RTP traffic.




___
--Bandwidth and Colocation provided by Easynews.com --

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






--
"Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org";
___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [Asterisk-Users] Oh oh. Micro$oft just noticed VoIP

[trixter aka Bret McDanel]

On Mon, 2006-06-26 at 13:16 -0400, Brian Capouch wrote:
> It will be interesting to see how many standards get broken, and how 
> many proprietary hooks get thrown into the pot.  The bean counters smell 
> some money, and their OS franchise is waning:
> 
> http://www.nytimes.com/2006/06/26/technology/26soft.html
> 

and they have been working with cisco on ice (which is standards based,
although ice is more of an extension to sip than anything else).  But
shhh that doesnt help the people that want to bash for no other reason
than they can!


-- 
Trixter http://www.0xdecafbad.com Bret McDanel
Belfast IE +44 28 9099 6461DE +49 801 777 555 3402
Utrecht NL +31 306 553058  US WA +1 360 207 0479
US NY +1 516 687 5200  FreeWorldDialup: 635378
http://www.trxtel.com the VoIP provider that pays you!


signature.asc
Description: This is a digitally signed message part
___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!

[Marco Mouta]

Sorry  to all,

Now only English speaking :)

Your translation was perfect.

Thanks once more

On 6/26/06, Mike Fedyk <[EMAIL PROTECTED]> wrote:

Tzafrir Cohen wrote:
> On Mon, Jun 26, 2006 at 09:39:11AM -0300, Josué Conti wrote:
>
>> Marco, bom dia.
>> Essa interligação entre o Skype e Asterisk, é feito atavés de um módulo
>> externo?
>> É freeware?
>> Podemos seguir com o projeto Asterisk-PT?
>>
>
> English, please, folks.
>
>
I don't know Portuguese and my Spanish is terrible, but I understood
that Josue wanted to know if he needed any external modules.  Marco
pointed him to the right place to get skype-to-sip and now they're going
to collaborate.

So, please guys English please or you'll get more of my bad translations. ;)

Mike
___
--Bandwidth and Colocation provided by Easynews.com --

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




--
Com os melhores cumprimentos,

Marco Mouta
___
--Bandwidth and Colocation provided by Easynews.com --

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

[Asterisk-Users] Email notification

[Roger Workman]

Is there a way to get asterisk to send you a email when it looses or an 
extension doesn’t re-register

Roger Workman
Business Development
Upperclassman/Universal Holdings LLC
Voice: 304.324.3800
 Fax:   304.324.3801
ICQ: 4447584
Website: http://www.upperclassman.net
Billing Questions: billing at upperclassman.net
Rental Questions: rentals at upperclassman.net
Maintenance: help at upperclassman.net



This e-mail and any of its attachments may contain sensitive information, which 
is privileged, confidential, or subject to copyright belonging to RW Management 
Inc, Universal Holdings LLC or Upperclassman LLC. This e-mail is intended 
solely for the use of the individual or entity to which it is addressed. If you 
are not the intended recipient of this e-mail, you are hereby notified that any 
dissemination, distribution, copying, or action taken in relation to the 
contents of and attachments to this e-mail is strictly prohibited and may be 
unlawful. If you have received this e-mail in error, please notify the sender 
immediately and permanently delete the original and any copy of or printout of 
this e-mail.

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric "ManxPower" 
Wieling
Sent: Monday, June 26, 2006 1:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] '500 Internal Server' Error on SIP NOTIFY

Yes.  It does not seem to cause any problems.

Douglas Garstang wrote:
> Is anyone getting '500 Internal Server' errors back from their Polycom phones 
> when Asterisk sends a SIP NOTIFY message to them?
> I called Polycom tech support, who where utterly useless.
> Of course Polycom won't officially support it anyway, as they only support 
> Asterisk Business Edition. We're using 1.2.9, but it's been ocurring for 
> quite some time. We have about 35 phones and it's happening on most (also on 
> the few running SIP software 1.6.6).
>
> SIP Software version: 1.6.3.0067
> BootROM version: 2.6.2.0032
>
> Reliably Transmitting (no NAT) to xxx.187.128.95:5060:
> NOTIFY sip:[EMAIL PROTECTED] SIP/2.0
> Via: SIP/2.0/UDP xxx.187.142.203:5060;branch=z9hG4bK4d777013;rport
> From: ;tag=as6fd80d1b
> To: "Front Desk" ;tag=3B576862-120A3007
> Contact: 
> Call-ID: [EMAIL PROTECTED]
> CSeq: 114 NOTIFY
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Event: presence
> Content-Type: application/xpidf+xml
> Subscription-State: active
> Content-Length: 371
>
> 
> 
> 
> 
> 
>  DEFANGED_priority="0.80">
> 
> 
> 
> 
> 
>
>
> <-- SIP read from xxx.187.128.95:5060:
> SIP/2.0 500 Internal Server Error
> Via: SIP/2.0/UDP xxx.187.142.203:5060;branch=z9hG4bK4d777013;rport
> From: ;tag=as6fd80d1b
> To: "Front Desk" ;tag=3B576862-120A3007
> CSeq: 114 NOTIFY
> Call-ID: [EMAIL PROTECTED]
> Contact: 
> Event: presence
> User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.6.0036
> Content-Length: 0
>
> Doug.
> ___
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>


--
Now accepting new clients in Birmingham, Atlanta, Huntsville,
Chattanooga, and Montgomery.
___
--Bandwidth and Colocation provided by Easynews.com --

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


___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!

[Brian Capouch]

Tzafrir Cohen wrote:

On Mon, Jun 26, 2006 at 09:39:11AM -0300, Josué Conti wrote:


Marco, bom dia.
Essa interligação entre o Skype e Asterisk, é feito atavés de um módulo
externo?
É freeware?
Podemos seguir com o projeto Asterisk-PT?



English, please, folks.



Let them talk.  What's it hurt the rest of us?

We have seen the wages of tortured English sometimes unleashed on the 
list.  If they're getting the job done, I say hit the "Delete" button 
and get on with your life.


If 80% of the list traffic were in foreign languages, then I would say 
we would have an issue.


MO.

B.

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

___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [Asterisk-Users] asterisk-stat display problems

[Chris Earle \(CBL\)]

yep

I don't know exactly which things the php-gd is used for, but like I said,
someof the pages work, like the main record page, the little red bars
showing call volume work fine


Really annoying, cause it looks so good at that point, then you go to use
the other pages/features and it's broken

Thanks for the reply,

--
Chris



- Original Message - 
From: "Mojo with Horan & Company, LLC" <[EMAIL PROTECTED]>
To: "Chris Earle (CBL)" <[EMAIL PROTECTED]>; "Asterisk Users Mailing List -
Non-Commercial Discussion" 
Sent: Monday, June 26, 2006 1:49 PM
Subject: Re: [Asterisk-Users] asterisk-stat display problems


> do you have the php-gd package installed on your * server?
>
> Chris Earle (CBL) wrote:
> > Hey all,
> >
> > having a terrible time with asterisk-stat -- it runs, server is fine,
but
> > some of the pages don't display properly/at all --- I think this is a
code
> > problem with them, but not sure.  I thought everyone loved the
asterisk-stat
> > package?
> >
> > See below problems.  Any ideas?  Areski hasn't replied to me since
> >
> > --
> > Chris
> >
> >
> > - Original Message - 
> > From: "Chris Earle (CBL)"
> > To: "Areski"
> > Sent: Tuesday, June 13, 2006 6:15 PM
> > Subject: Re: CDR-Analyser version question
> >
> >
> >> Thank you for the reply;
> >>
> >> I see now that the main file cdr.php does work with argument ?s=1, 2,
> >> etc
> >> but when s=0, does not load
> >>
> >> I get an Apache error :
> >>
> >>  relocation error: /usr/lib/php4/20020429/gd.so: undefined symbol:
> >> gdFontCacheShutdown
> >>
> >> Not sure if that means anything important;
> >>
> >>
> >>
> >>
> >> Also, in the new Asterisk-Stat feature pages like Calls Compare (s=2),
the
> >> pages do not complete their output -- no search button displayed, stops
> >> outputting radio buttons for UserField row etc
> >>
> >> So at this point, only the main Call-log page (s=1) works.
> >>
> >>
> >> I am using Debian with php 4.4.1
> >> Mysql ver 12.22, Distrib 4.0.24
> >> GD Library is 2.0.33 I think
> >>
> >>
> >> Any input you can pass along would be much appreciated!  I am
comfortable
> >> with php so if you want me to modify sourcecode that is fine
> >>
> >> Thanks!
> >>
> >>
> >>
> >>
> >> - Original Message - 
> >> From: "Areski"
> >> To: "Chris Earle (CBL)"
> >> Sent: Sunday, May 28, 2006 7:11 PM
> >> Subject: Re: CDR-Analyser version question
> >>
> >>
> >>> No there is no asterisk requirement to make asterisk-stat.
> >>> Indeed the soft is only based on the cdr database. If you have an
error
> >>> you can give me more info, I may help you.
> >>>
> >>> Rgds, Areski
> >>>
> >>> On 5/25/06, Chris Earle (CBL) wrote:
>  Hi there,
> 
>  quick question:
> 
>  Does asterisk-stat v2.0.1 require Asterisk 1.2+ ?  I am using
Asterisk
> >> 1.0.x
>  and can't get it to load the cdr.php properly
> 
>  so I downgraded to v1.3 and it works...
> 
>  Let me know if there's an asterisk version requirement for each
> > version
> >> of
>  the CDR Analyser
> 
>  Thanks!
> 
> 
> 
>  --
>  Chris Earle
> 
> 
> > 
> >
> >
>
> -- 
> Mojo <[EMAIL PROTECTED]>
> Office Manger, Horan & Company, LLC
> (907) 747- x112
>
> -- 
> This message has been scanned for viruses and dangerous content by
> MailScanner, and is believed to be clean.


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

___
--Bandwidth and Colocation provided by Easynews.com --

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

RE: [Asterisk-Users] '500 Internal Server' Error on SIP NOTIFY

[Douglas Garstang]

> -Original Message-
> From: Doug Lytle [mailto:[EMAIL PROTECTED]
> Sent: Monday, June 26, 2006 11:08 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] '500 Internal Server' Error on 
> SIP NOTIFY
> 
> 
> Douglas Garstang wrote:
> > Is anyone getting '500 Internal Server' errors back from 
> their Polycom phones when Asterisk sends a SIP NOTIFY message to them?
> >   
> 
> Yes, for quite a while.  Happens for us, when you do a 
> transfer via the 
> Polycom's transfer button.  Doesn't seem to cause any problems though.

It's bloody annoying though, especially for those type-A's that don't like to 
see the console cluttered up with junk. :)

Doug
___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [Asterisk-Users] asterisk-stat display problems

[Mojo with Horan & Company, LLC]

do you have the php-gd package installed on your * server?

Chris Earle (CBL) wrote:

Hey all,

having a terrible time with asterisk-stat -- it runs, server is fine, but
some of the pages don't display properly/at all --- I think this is a code
problem with them, but not sure.  I thought everyone loved the asterisk-stat
package?

See below problems.  Any ideas?  Areski hasn't replied to me since

--
Chris


- Original Message - 
From: "Chris Earle (CBL)"

To: "Areski"
Sent: Tuesday, June 13, 2006 6:15 PM
Subject: Re: CDR-Analyser version question



Thank you for the reply;

I see now that the main file cdr.php does work with argument ?s=1, 2,
etc
but when s=0, does not load

I get an Apache error :

 relocation error: /usr/lib/php4/20020429/gd.so: undefined symbol:
gdFontCacheShutdown

Not sure if that means anything important;




Also, in the new Asterisk-Stat feature pages like Calls Compare (s=2), the
pages do not complete their output -- no search button displayed, stops
outputting radio buttons for UserField row etc

So at this point, only the main Call-log page (s=1) works.


I am using Debian with php 4.4.1
Mysql ver 12.22, Distrib 4.0.24
GD Library is 2.0.33 I think


Any input you can pass along would be much appreciated!  I am comfortable
with php so if you want me to modify sourcecode that is fine

Thanks!




- Original Message - 
From: "Areski"

To: "Chris Earle (CBL)"
Sent: Sunday, May 28, 2006 7:11 PM
Subject: Re: CDR-Analyser version question



No there is no asterisk requirement to make asterisk-stat.
Indeed the soft is only based on the cdr database. If you have an error
you can give me more info, I may help you.

Rgds, Areski

On 5/25/06, Chris Earle (CBL) wrote:

Hi there,

quick question:

Does asterisk-stat v2.0.1 require Asterisk 1.2+ ?  I am using Asterisk

1.0.x

and can't get it to load the cdr.php properly

so I downgraded to v1.3 and it works...

Let me know if there's an asterisk version requirement for each

version

of

the CDR Analyser

Thanks!



--
Chris Earle








--
Mojo <[EMAIL PROTECTED]>
Office Manger, Horan & Company, LLC
(907) 747- x112
___
--Bandwidth and Colocation provided by Easynews.com --

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

[Asterisk-Users] registering a Motorola VT1005

[Brandon Warner]

I am trying to 
register a motorola VT1005. I have many supura ata's that work fine. Anyhelp, 
would be great.
Brandon
___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [Asterisk-Users] '500 Internal Server' Error on SIP NOTIFY

[Bruce Reeves]

I have been seeing the same errors here with Polycom 501 and 601 phones. Asterisk version is 1.2.9.1 and Polycom SIP version 1.6.3On 6/26/06, 
Douglas Garstang <[EMAIL PROTECTED]> wrote:
Is anyone getting '500 Internal Server' errors back from their Polycom phones when Asterisk sends a SIP NOTIFY message to them?I called Polycom tech support, who where utterly useless.Of course Polycom won't officially support it anyway, as they only support Asterisk Business Edition. We're using 
1.2.9, but it's been ocurring for quite some time. We have about 35 phones and it's happening on most (also on the few running SIP software 1.6.6).SIP Software version: 1.6.3.0067
BootROM version: 2.6.2.0032Reliably Transmitting (no NAT) to xxx.187.128.95:5060:NOTIFY sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP xxx.187.142.203:5060;branch=z9hG4bK4d777013;rportFrom: ;tag=as6fd80d1bTo: "Front Desk" <
sip:[EMAIL PROTECTED]>;tag=3B576862-120A3007Contact: Call-ID: 
[EMAIL PROTECTED]CSeq: 114 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: presenceContent-Type: application/xpidf+xml
Subscription-State: activeContent-Length: 371
sip:[EMAIL PROTECTED];method=SUBSCRIBE" />
sip:[EMAIL PROTECTED];user=ip" priority="0.80">
<-- SIP read from xxx.187.128.95:5060:SIP/2.0 500 Internal Server ErrorVia: SIP/2.0/UDP xxx.187.142.203:5060;branch=z9hG4bK4d777013;rportFrom: <
sip:[EMAIL PROTECTED]>;tag=as6fd80d1bTo: "Front Desk" ;tag=3B576862-120A3007CSeq: 114 NOTIFY
Call-ID: [EMAIL PROTECTED]Contact: Event: presence
User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.6.0036Content-Length: 0Doug.___--Bandwidth and Colocation provided by Easynews.com
 --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users
-- BruceNortex Networks
___
--Bandwidth and Colocation provided by Easynews.com --

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

[Asterisk-Users] Oh oh. Micro$oft just noticed VoIP

[Brian Capouch]

It will be interesting to see how many standards get broken, and how 
many proprietary hooks get thrown into the pot.  The bean counters smell 
some money, and their OS franchise is waning:


http://www.nytimes.com/2006/06/26/technology/26soft.html

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

___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [Asterisk-Users] STUN?

[Martin Joseph]

On Jun 26, 2006, at 9:32 AM, Raymond Tant wrote:

Hi all, Could someone point at resources for running Asterisk behind a firewall.STUN keeps coming up but, alas, I’m easily confused. J

STUN is just a way to discover the true address of a machine behind a NAT.

Firewalls aren't really an issue per se,  other then needing to open particular ports for asterisk to use. For example, udp port 4569 for IAX2 traffic, and 5060 for SIP signaling, as well as ports in the 1-2 range RTP traffic.


___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [Asterisk-Users] '500 Internal Server' Error on SIP NOTIFY

[Eric \"ManxPower\" Wieling]

Yes.  It does not seem to cause any problems.

Douglas Garstang wrote:

Is anyone getting '500 Internal Server' errors back from their Polycom phones 
when Asterisk sends a SIP NOTIFY message to them?
I called Polycom tech support, who where utterly useless.
Of course Polycom won't officially support it anyway, as they only support 
Asterisk Business Edition. We're using 1.2.9, but it's been ocurring for quite 
some time. We have about 35 phones and it's happening on most (also on the few 
running SIP software 1.6.6).

SIP Software version: 1.6.3.0067
BootROM version: 2.6.2.0032
 
Reliably Transmitting (no NAT) to xxx.187.128.95:5060:

NOTIFY sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP xxx.187.142.203:5060;branch=z9hG4bK4d777013;rport
From: ;tag=as6fd80d1b
To: "Front Desk" ;tag=3B576862-120A3007
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 114 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: presence
Content-Type: application/xpidf+xml
Subscription-State: active
Content-Length: 371
 












 
 
<-- SIP read from xxx.187.128.95:5060: 
SIP/2.0 500 Internal Server Error

Via: SIP/2.0/UDP xxx.187.142.203:5060;branch=z9hG4bK4d777013;rport
From: ;tag=as6fd80d1b
To: "Front Desk" ;tag=3B576862-120A3007
CSeq: 114 NOTIFY
Call-ID: [EMAIL PROTECTED]
Contact: 
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.6.0036
Content-Length: 0
 
Doug.

___
--Bandwidth and Colocation provided by Easynews.com --

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





--
Now accepting new clients in Birmingham, Atlanta, Huntsville, 
Chattanooga, and Montgomery.

___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [Asterisk-Users] '500 Internal Server' Error on SIP NOTIFY

[Doug Lytle]

Douglas Garstang wrote:

Is anyone getting '500 Internal Server' errors back from their Polycom phones 
when Asterisk sends a SIP NOTIFY message to them?
  


Yes, for quite a while.  Happens for us, when you do a transfer via the 
Polycom's transfer button.  Doesn't seem to cause any problems though.


Doug

--

Ben Franklin quote:

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


___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [Asterisk-Users] SE Michigan asterisk users group

[Rusty Dekema]

On 6/22/06, BerkHolz, Steven <[EMAIL PROTECTED]> wrote:

I am thinking of getting an asterisk user group together for either SE
Michigan or just Metro-Detroit.


I'm in Ann Arbor and would be interested in such a group; if you
create a mailing list for it, could you please add me?

Thanks,
Rusty
___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!

[Mike Fedyk]

Tzafrir Cohen wrote:

On Mon, Jun 26, 2006 at 09:39:11AM -0300, Josué Conti wrote:
  

Marco, bom dia.
Essa interligação entre o Skype e Asterisk, é feito atavés de um módulo
externo?
É freeware?
Podemos seguir com o projeto Asterisk-PT?



English, please, folks.

  
I don't know Portuguese and my Spanish is terrible, but I understood 
that Josue wanted to know if he needed any external modules.  Marco 
pointed him to the right place to get skype-to-sip and now they're going 
to collaborate.


So, please guys English please or you'll get more of my bad translations. ;)

Mike
___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [Asterisk-Users] best hardphone for Asterisk?

[Martin Joseph]

On Jun 26, 2006, at 7:14 AM, Doug Crompton wrote:


I guess I did not make my point clearly enough. I already do have just
that. An spa-3000 with ALL internal analog phones on it's on FXO.

That's wrong,  phones hook to an FXS.

 But this
gives just ONE extension for all phones. Yes I could get more FXS's and
run seperate wires.
I am using a HT-488 as my secondary FXS, which works ok,  but still has 
problems with DTMF unless I use inband through  the gateway a wellgate 
3701a in my case.


So with that background what would be nice is a wireless device like 
the

Panasonic cordless with one base and multiple phone capability that
connected via ethernet and serves the phones. Just wishful thinking. I
will stick with what I have until something useful, sylish, and less
expensive arrives.

Yeah, that does sound nice...

I have a panasonic cordless hooked to the HT-488,  this gives me 
mobility with my preferred fxs, and also allows for multiple calls to 
occur which is very slick.  ie I can call out long distance and calls 
can still arrive through the fxo to the other house phones.  This is 
kind of disorienting to housemates who are used to standard phone 
systems ;~)


Marty

___
--Bandwidth and Colocation provided by Easynews.com --

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

[Asterisk-Users] EuroISDN and Sangoma Card

[Tristan]

Hi List,

Just a little question about a notice from asterisk I don't understand:

Here is what I have as soon as I place a call on a E1 line with an a104D 
Sangoma Card ( asterisk 1.2.9.1 ) :


Jun 26 18:57:24 NOTICE[16489] channel.c: Don't know what to do with 
control frame 15


Does Anyone has a clue of how to get rid of that ?

May it's because I use the HDLC decoding in hardware ?

Thanks in advance for your help !!!

Cheers,

Tristan
___
--Bandwidth and Colocation provided by Easynews.com --

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

[Asterisk-Users] Soekris net4801-50 + IAXY

[Juan Luis Moyano]

Hi, I'm having an issue with a soekris net4801 board and a S101i "IAXy" 
device. When I connect a successfully provisioned IAXy directly via a 
crossover cable into an ethernet port of the soekris, the link led turns 
on orange so i'ts 10Mb and the activity led blinks like if there is some 
action going on but  when I try 'tcpdump -nettti sis1' I see nothing 
going on, no received packets. When I plug a regular PC on the same 
ethernet port there I can see all the traffic going on. I'm really stuck 
on this one. Help me please! Regards.


Juan Luis Moyano
___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [Asterisk-Users] This is getting really annoying - re: POSTFIX

[Martin Joseph]

On Jun 26, 2006, at 5:51 AM, Matt wrote:


What on earth is going on with the list?!?!   Some of my messages
never make it... then days later I get something like this back:
I'm hoping this was a transient issue.  I saw this too with a couple of 
posts,  but it's been ok since.


Marty

___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [Asterisk-Users] Meetme max users

[Bartosz Wegrzyn - asterisk]

thanks

we are planing to have around 50-60 users in 1 room.

> We've had over 100 participants spread across 30 meetme rooms on a
> single server before,  and the most we've had in a single meetme room
> is 46. I don't know of a hard limit for meetme participants and I
> haven't seen a limit in the code. You would most likely be limited by
> the resources on your server I would guess.
>
> MATT---
>
> On 6/23/06, Bartosz Wegrzyn - asterisk <[EMAIL PROTECTED]> wrote:
>> Does anyone knows what is the max of users that meefme can handle.
>> I am using Iax2 clients to connect to the conference.
>>
>> Thanks
>>
>>
>> ___
>> --Bandwidth and Colocation provided by Easynews.com --
>>
>> Asterisk-Users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
> ___
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>

___
--Bandwidth and Colocation provided by Easynews.com --

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

[Asterisk-Users] STUN?

[Raymond Tant]

Hi all,

 

Could someone point at resources for running Asterisk behind
a firewall.

STUN keeps coming up but, alas, I’m easily confused. J

 

Ray






___
--Bandwidth and Colocation provided by Easynews.com --

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

[Asterisk-Users] '500 Internal Server' Error on SIP NOTIFY

[Douglas Garstang]

Is anyone getting '500 Internal Server' errors back from their Polycom phones 
when Asterisk sends a SIP NOTIFY message to them?
I called Polycom tech support, who where utterly useless.
Of course Polycom won't officially support it anyway, as they only support 
Asterisk Business Edition. We're using 1.2.9, but it's been ocurring for quite 
some time. We have about 35 phones and it's happening on most (also on the few 
running SIP software 1.6.6).

SIP Software version: 1.6.3.0067
BootROM version: 2.6.2.0032
 
Reliably Transmitting (no NAT) to xxx.187.128.95:5060:
NOTIFY sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP xxx.187.142.203:5060;branch=z9hG4bK4d777013;rport
From: ;tag=as6fd80d1b
To: "Front Desk" ;tag=3B576862-120A3007
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 114 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: presence
Content-Type: application/xpidf+xml
Subscription-State: active
Content-Length: 371
 











 
 
<-- SIP read from xxx.187.128.95:5060: 
SIP/2.0 500 Internal Server Error
Via: SIP/2.0/UDP xxx.187.142.203:5060;branch=z9hG4bK4d777013;rport
From: ;tag=as6fd80d1b
To: "Front Desk" ;tag=3B576862-120A3007
CSeq: 114 NOTIFY
Call-ID: [EMAIL PROTECTED]
Contact: 
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.6.0036
Content-Length: 0
 
Doug.
___
--Bandwidth and Colocation provided by Easynews.com --

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

[Asterisk-Users] registering a Motorola vt1005

[Brandon Warner]

Has anyone 
successfully registered a moto vt1005 to asterisk. If so, 
how?
 

Brandon Warner
Assistant Director of NOC Services
Dark Fiber Solutions
600 1/2 Grant Ave.
York, NE 68467
Office: 402-362-3334
Cell:402-366-2087
 
"The information 
transmitted is intended only for the person or entity to which it is addressed 
and may contain confidential and/or privileged material. Any review, 
retransmission, dissemination or other use of, or taking of any action in 
reliance upon, this information by persons or entities other than the intended 
recipient is prohibited. If you received this in error, please contact the 
sender and delete the material from all computers."
 
___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [Asterisk-Users] Signaling and media

[Olle E Johansson]

26 jun 2006 kl. 10.54 skrev Jean-Michel Hiver:


Johansson Olle E a écrit :



26 jun 2006 kl. 07.10 skrev Martin Joseph:



On Jun 25, 2006, at 4:11 PM, Jean-Michel Hiver wrote:


Hi List,

Is there a way to tell asterisk to only accept SIP streams from   
the same IP address that is used for signaling?



"SIP streams" are signalling...


Sorry, I was talking about the media.


Have you tested the ACL features in  sip.conf - accept/deny ?


Any pointers on these ACLs?


Check "permit" and "deny" in sip.conf.

/O

---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk Training http://edvina.net/training/



___
--Bandwidth and Colocation provided by Easynews.com --

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

RE: [Asterisk-Users] RE: Voice calls sent to fax extension

[Colin Anderson]

>Surely once the call has been bridged the fax detection should turn off ?

I'd like to find out a way it can be done, can anyone else comment?

___
--Bandwidth and Colocation provided by Easynews.com --

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

[Asterisk-Users] asterisk-stat display problems

[Chris Earle \(CBL\)]

Hey all,

having a terrible time with asterisk-stat -- it runs, server is fine, but
some of the pages don't display properly/at all --- I think this is a code
problem with them, but not sure.  I thought everyone loved the asterisk-stat
package?

See below problems.  Any ideas?  Areski hasn't replied to me since

--
Chris


- Original Message - 
From: "Chris Earle (CBL)"
To: "Areski"
Sent: Tuesday, June 13, 2006 6:15 PM
Subject: Re: CDR-Analyser version question


> Thank you for the reply;
>
> I see now that the main file cdr.php does work with argument ?s=1, 2,
> etc
> but when s=0, does not load
>
> I get an Apache error :
>
>  relocation error: /usr/lib/php4/20020429/gd.so: undefined symbol:
> gdFontCacheShutdown
>
> Not sure if that means anything important;
>
>
>
>
> Also, in the new Asterisk-Stat feature pages like Calls Compare (s=2), the
> pages do not complete their output -- no search button displayed, stops
> outputting radio buttons for UserField row etc
>
> So at this point, only the main Call-log page (s=1) works.
>
>
> I am using Debian with php 4.4.1
> Mysql ver 12.22, Distrib 4.0.24
> GD Library is 2.0.33 I think
>
>
> Any input you can pass along would be much appreciated!  I am comfortable
> with php so if you want me to modify sourcecode that is fine
>
> Thanks!
>
>
>
>
> - Original Message - 
> From: "Areski"
> To: "Chris Earle (CBL)"
> Sent: Sunday, May 28, 2006 7:11 PM
> Subject: Re: CDR-Analyser version question
>
>
> > No there is no asterisk requirement to make asterisk-stat.
> > Indeed the soft is only based on the cdr database. If you have an error
> > you can give me more info, I may help you.
> >
> > Rgds, Areski
> >
> > On 5/25/06, Chris Earle (CBL) wrote:
> > > Hi there,
> > >
> > > quick question:
> > >
> > > Does asterisk-stat v2.0.1 require Asterisk 1.2+ ?  I am using Asterisk
> 1.0.x
> > > and can't get it to load the cdr.php properly
> > >
> > > so I downgraded to v1.3 and it works...
> > >
> > > Let me know if there's an asterisk version requirement for each
version
> of
> > > the CDR Analyser
> > >
> > > Thanks!
> > >
> > >
> > >
> > > --
> > > Chris Earle
> > >
> > >



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

___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [Asterisk-Users] Asterisk-1.2.9.1 with Siemens HiPath 4000

[Josué Conti]

Jun 26 12:43:16 WARNING[31148]: chan_zap.c:8386 pri_dchannel: Ring requested on unconfigured channel 0/16 span 1I noticed this message in the CLI, when I tried to effect one call of HiPath 4000 for asterisk. Ring occurred, however when the voicemail of asterisk took care of call it was dumb, without no sound. I thank the attention

RegardsJosué
 
2006/6/26, Josué Conti <[EMAIL PROTECTED]>:


Hi Richard.
Thank you very much for its attention. In the reality what is occurring is that in some originated calls of the HiPath with destination to the Asterisk they are being without the dumb and rings. I do not have this parameter in my HiPath 4000, what I have seemed in the COT is TR6T (1tr6 isdn tie link) would be this parameter?      Best Regards 
Josué
2006/6/26, richard Coco <[EMAIL PROTECTED]>: 

Hi Josuéif the Siemens phone calls Asterisk, it didn't get adial tone from Asterisk? Is it correct? 
if yes, this is depending of Asterisk which didn'tgenerates a ringback messages as it expexts dial tongeneration localy. So try this workaround for HiPathlocal dial ton generation:-> Add option TR6Q(TRGT) to the class of trunk (COT) 
parametershope it will help...rich--- Josué Conti <[EMAIL PROTECTED]
> wrote:>   Hello all.>  I have installed and functioning asterisk-1.2.9.1> where I effected one> upgrade in asterisk-1.0.9, is interconnected with a> PABX Siemens HiPath 4000
> in ISDN PRI with protocol QSIG, the one that is> happening he is that the > calls originated for PABX Siemens and destined to> SIP phones asterisk are> being without audio, nor Ring, is dumb. They could
> help in this case me?> Best Regards>> Josué > > ___> --Bandwidth and Colocation provided by 
Easynews.com> -->> Asterisk-Users mailing list> To UNSUBSCRIBE or update options visit: >>
http://lists.digium.com/mailman/listinfo/asterisk-users>__Do You Yahoo!? Tired of spam?  Yahoo! Mail has the best spam protection around
http://mail.yahoo.com___--Bandwidth and Colocation provided by 
Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:  
http://lists.digium.com/mailman/listinfo/asterisk-users 
 
___
--Bandwidth and Colocation provided by Easynews.com --

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

[Asterisk-Users] MeetMe Volume Issues

[Justin Tunney]

Hi,

I'm using the latest 1.2 release of Asterisk and I've noticed that one of the 
releases of Zaptel or Asterisk in the past few months seems to have 
introduced a problem with MeetMe.  Here are the symptoms:

 - Very high volume for internal IP phone users
 - Very low volume for incoming analog callers
 - Analog callers can not hear each other in conference

This seems to happen with the 4-port and 24-port TDM cards sold by Digium.  
Has anyone else experienced similar problems?

Thanks!

--
Justin Tunney
___
--Bandwidth and Colocation provided by Easynews.com --

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

RE: [Asterisk-Users] SE Michigan asterisk users group

[Carlos Alperin]

Ok, I count at least 4. Just lets propose when & 
where for the first meeting group, and start to think about issues 
discussion.
 
Tom, what to we need for the mailing list? I can do 
something about that.
 
Carlos Alperin
 
 



From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Jon 
RadonSent: Monday, June 26, 2006 11:12 AMTo: Asterisk 
Users Mailing List - Non-Commercial DiscussionSubject: Re: 
[Asterisk-Users] SE Michigan asterisk users group
I'm also in the area, near Southfield.  I'd be interested as 
well.
On 6/22/06, BerkHolz, 
Steven <[EMAIL PROTECTED]> 
wrote: 
I 
  am thinking of getting an asterisk user group together for either 
  SEMichigan or just Metro-Detroit.How much interest in asterisk in 
  Michigan is there on this list?I am already on the board of 
  glimasoutheast, with is a group fortechnology professionals. (very broad 
  range)It is a spin-off from Automation Alley, which is SE Michigan's 
  version of Silicone Valley.--Stevenhttp://www.glimasoutheast.org___--Bandwidth 
  and Colocation provided by Easynews.com 
  --Asterisk-Users mailing listTo UNSUBSCRIBE or update options 
  visit:  http://lists.digium.com/mailman/listinfo/asterisk-users 
  -- Is it something someone 
said, was it something someone said? 
___
--Bandwidth and Colocation provided by Easynews.com --

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

[Asterisk-Users] Pickup zap issue

[Fredrik von Kantzow]

Hi,

As you can see in this log the problem is very new to me..

Connected to Asterisk 1.2.5 currently running on volcano (pid = 7874)
volcano*CLI> set verbose 4
Verbosity was 0 and is now 4
-- Starting simple switch on 'Zap/2-1'
-- Executing Dial("Zap/2-1", "SIP/180|60") in new stack
-- Called 180
-- SIP/180-d11c is ringing
-- SIP/180-d11c answered Zap/2-1

Now everything looks good, BUT when I pickup the handset on extension 
180 nothing happens as you can see in the log Asterisk notices that 180 
answers but the person calling in on the ZAP interface still hears the 
ringing tone, it seems that Asterisk does not bridge the call or 
physically answers it on the ZAP interface.


Anybody had issues like this or could have an idea where to start 
looking for the problem?


Fred
___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [Asterisk-Users] RE: Voice calls sent to fax extension

[Julian Lyndon-Smith]

Surely once the call has been bridged the fax detection should turn off ?

Julian
Colin Anderson wrote:

yes. Wind whistling in a car, female voices at a particular pitch and
volume, fax machine running in the background of a voice call with the
speaker on. It happens. Whether this is a problem or not depends on your
pain threshold. I get a couple reports a week, which means that it actually
happens ten times a couple times a week, so twenty times a week, and I
process ~20K calls a week, so it happens to me .1 % of the time. Is this a
problem for me? Nah. Is it a problem for you? Maybe - what's your pain
threshold?

ps fwiw, this behavior will happen with any device that listens inline for a
CNG tone, so it's not just an Asterisk thing

-Original Message-
From: Paul A. Pringle [mailto:[EMAIL PROTECTED]
Sent: Monday, June 26, 2006 8:54 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] RE: Voice calls sent to fax extension


I thought it might be an inadvertent button press, but none of the keys
(on my phone at least) are recognized by Asterisk as fax tones.  This
has happened to two different users getting calls from different people
using different equipment.  Does anyone else see this behavior
occasionally?

Paul

-Original Message-
Date: Fri, 23 Jun 2006 15:29:31 -0400
From: "Bill Gibbs" <[EMAIL PROTECTED]>
Subject: RE: [Asterisk-Users] Voice calls sent to fax extension
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Message-ID:
<[EMAIL PROTECTED]>
Content-Type: text/plain;   charset="us-ascii"

Maybe their fat jowls hit a few buttons on the keypad and sent the fax
tone down the line and they didn't realize it?

Bill

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul A.
Pringle
Sent: Friday, June 23, 2006 2:51 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Voice calls sent to fax extension

I have a situation that has repeated itself a few times.  Someone calls
into Asterisk and is connected with a voice extension.  At some point
during the call, the log shows "chan_zap.c: DTMF digit: f on Zap/2-1".
At this point, the call is redirected to receive a fax and the Asterisk
voice extension is hung up.  The users report that there were no
noticable tones heard just before the cutover, so I'm not sure what's
going on.  Is there a way to disable detection of faxes after the
voicecall is initiated?  We're running a Digium card to convert our
analog trunks if that makes any difference.

Thanks!

Paul
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --

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




___
--Bandwidth and Colocation provided by Easynews.com --

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

RE: [Asterisk-Users] RE: Voice calls sent to fax extension

[Colin Anderson]

>Yes, which is why I disable faxdetect entirely.  My sister-in-law was 
>constantly being detected as a fax machine several minutes into 
>conversations with my wife.  As funny as that may seem at first ... 
>those two eventually make it a not-so-funny situation for me.

lol, Spousal Acceptance Factor, I have found, is the cornerstone of any
Asterisk feature. Seriously, if I have a great idea and I am going to
introduce it to my users, I put it on my Asterisk box at home and let my
wife use it. If she says: "That doesn't suck" then I am golden with my
users. Note that she doesn't say: "Wow thats a great feature" she simply
says: "That doesn't suck" which to her is high praise. 
___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [Asterisk-Users] Asterisk-1.2.9.1 with Siemens HiPath 4000

[Josué Conti]

Hi Richard.
Thank you very much for its attention. In the reality what is occurring is that in some originated calls of the HiPath with destination to the Asterisk they are being without the dumb and rings. I do not have this parameter in my HiPath 4000, what I have seemed in the COT is TR6T (1tr6 isdn tie link) would be this parameter?      Best Regards
Josué
2006/6/26, richard Coco <[EMAIL PROTECTED]>:
Hi Josuéif the Siemens phone calls Asterisk, it didn't get adial tone from Asterisk? Is it correct?
if yes, this is depending of Asterisk which didn'tgenerates a ringback messages as it expexts dial tongeneration localy. So try this workaround for HiPathlocal dial ton generation:-> Add option TR6Q(TRGT) to the class of trunk (COT)
parametershope it will help...rich--- Josué Conti <[EMAIL PROTECTED]> wrote:>   Hello all.>  I have installed and functioning 
asterisk-1.2.9.1> where I effected one> upgrade in asterisk-1.0.9, is interconnected with a> PABX Siemens HiPath 4000> in ISDN PRI with protocol QSIG, the one that is> happening he is that the
> calls originated for PABX Siemens and destined to> SIP phones asterisk are> being without audio, nor Ring, is dumb. They could> help in this case me?> Best Regards>> Josué
> > ___> --Bandwidth and Colocation provided by Easynews.com> -->> Asterisk-Users mailing list> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users>__Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection aroundhttp://mail.yahoo.com___--Bandwidth and Colocation provided by 
Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:  http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation provided by Easynews.com --

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

[Asterisk-Users] AEL scripting, CUT use and string concatenation

[Marcello Lupo]

Hi to all,
i'm wondering to realize a dynamic macro that can take the number of 
extensions to RING,the ring type and all the parameter in a dynamic way.


I have done this code to test it:

macro pbx-ring-group-ael(pbx_id,num_int,ring_type,timeout,ext_string) {
//; pbx_id = Id of PBX in the DB
//; num_int = Quantity of extensions to ring
//; ring_type = Kind of RING (C=contemporaneous S=sequential)
//; timeout = Amount of time to ring
//; ext_string = String with extension numbers like 101-102-103-104-105

if(${ring_type}=C) {
for (x=1 ; ${x} <= ${num_int} ; x=${x} + 1) {
int=${CUT(ext_string,,${x})};
if(${x} = 1) {
dialstring=SIP/${pbx_id}-${int};
} else {

dialstring=${dialstring}&SIP/${pbx_id}-${int};
};
if(${x} = ${num_int}) {
dialstring=${dialstring}|${timeout};
};

NoOp(STRING ${dialstring});
};
};
Hangup();
};

I'm getting problems both in the CUT expression and the concatenation of 
strings due to the presence of &,/,- in it. I think something can be 
done with double quote but it will be inserted as part of the string, so 
the concatenation will fail.
For the CUT i don't know what is the problem. I tried with 
CUT(int=(ext_string,,${x}) too but without success.


This is the dialplan resulting from the expansion of the ael script:

show dialplan macro-pbx-ring-group-ael
[ Context 'macro-pbx-ring-group-ael' created by 'pbx_ael' ]
  's' =>1. Set(pbx_id=${ARG1})[pbx_ael]
2. Set(num_int=${ARG2})   [pbx_ael]
3. Set(ring_type=${ARG3}) [pbx_ael]
4. Set(timeout=${ARG4})   [pbx_ael]
5. Set(ext_string=${ARG5})[pbx_ael]
6. GotoIf($[ ${ring_type}=C ]?7:22)   [pbx_ael]
7. Set(x=$[ 1 ])  [pbx_ael]
8. GotoIf($[ ${x} <= ${num_int} ]?9:21)   [pbx_ael]
9. Set(x=$[ ${x} + 1 ])   [pbx_ael]
10. Set(int=$[ ${CUT(ext_string,,${x})} ]) [pbx_ael]
11. GotoIf($[ ${x} = 1 ]?12:14)   [pbx_ael]
12. Set(dialstring=$[ SIP/${pbx_id}-${int} ]) [pbx_ael]
13. Goto(15)  [pbx_ael]
14. Set(dialstring=$[ 
${dialstring}&SIP/${pbx_id}-${int} ]) [pbx_ael]
15. NoOp(Finish 
if-for-if-pbx-ring-group-ael-6-7-11) [pbx_ael]

16. GotoIf($[ ${x} = ${num_int} ]?17:18) [pbx_ael]
17. Set(dialstring=$[ ${dialstring}|${timeout} ]) 
[pbx_ael]
18. NoOp(Finish 
if-for-if-pbx-ring-group-ael-6-7-16) [pbx_ael]

19. NoOp(STRING ${dialstring}) [pbx_ael]
20. Goto(8)[pbx_ael]
21. NoOp(Finish for-if-pbx-ring-group-ael-6-7) 
[pbx_ael]

22. NoOp(Finish if-pbx-ring-group-ael-6)  [pbx_ael]
23. Hangup()[pbx_ael]

-= 1 extension (23 priorities) in 1 context. =-

This is the log of errors:

-- Executing Macro("SIP/1234-100-b263", 
"pbx-ring-group-ael|1234|5|C|20|101-102-103-104-105") in new stack

-- Executing Set("SIP/1234-100-b263", "pbx_id=1234") in new stack
-- Executing Set("SIP/1234-100-b263", "num_int=5") in new stack
-- Executing Set("SIP/1234-100-b263", "ring_type=C") in new stack
-- Executing Set("SIP/1234-100-b263", "timeout=20") in new stack
-- Executing Set("SIP/1234-100-b263", 
"ext_string=101-102-103-104-105") in new stack

-- Executing GotoIf("SIP/1234-100-b263", "1?7:22") in new stack
-- Goto (macro-pbx-ring-group-ael,s,7)
-- Executing Set("SIP/1234-100-b263", "x=1") in new stack
-- Executing GotoIf("SIP/1234-100-b263", "1?9:21") in new stack
-- Goto (macro-pbx-ring-group-ael,s,9)
-- Executing Set("SIP/1234-100-b263", "x=2") in new stack
Jun 26 17:17:24 WARNING[31282]: ast_expr2.fl:176 ast_yyerror: 
ast_yyerror(): syntax error: syntax error, unexpected $end, expecting 
TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input:


  ^
Jun 26 17:17:24 WARNING[31282]: ast_expr2.fl:180 ast_yyerror: If you 
have questions, please refer to doc/README.variables in the asterisk source.

-- Executing Set("SIP/1234-100-b263", "int=0") in new stack
-- Executing GotoIf("SIP/1234-100-b263", "0?12:14") in new stack
-- Goto (macro-pbx-ring-group-ael,s,14)
Jun 26 17:17:24 WARNING[31282]: ast_expr2.fl:176 ast_yyerror: 
ast_yyerror(): syntax error: syntax error, unexpected TOK_AND, expecting 
TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input:

 &SIP/1234-0
 ^
Jun 26 17:17:24 WARNING[31282]: ast_expr2.fl:180

RE: [Asterisk-Users] RE: Voice calls sent to fax extension

[Tim Sharp]

I also had an intermittent problem, on average one or two faxes a week, that 
were not recongnzed as a fax.  Then I switched phone companies and have not had 
that problem since.  It has been over 2 months.  I addition, my echo problem 
has been practically eliminated and overall voice quality is better. I believe 
that it is because of better levels of RX gain for fax recognition, but don't 
know for sure. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Paul A.
Pringle
Sent: Monday, June 26, 2006 10:54 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] RE: Voice calls sent to fax extension


I thought it might be an inadvertent button press, but none of the keys
(on my phone at least) are recognized by Asterisk as fax tones.  This
has happened to two different users getting calls from different people
using different equipment.  Does anyone else see this behavior
occasionally?

Paul

-Original Message-
Date: Fri, 23 Jun 2006 15:29:31 -0400
From: "Bill Gibbs" <[EMAIL PROTECTED]>
Subject: RE: [Asterisk-Users] Voice calls sent to fax extension
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Message-ID:
<[EMAIL PROTECTED]>
Content-Type: text/plain;   charset="us-ascii"

Maybe their fat jowls hit a few buttons on the keypad and sent the fax
tone down the line and they didn't realize it?

Bill

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul A.
Pringle
Sent: Friday, June 23, 2006 2:51 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Voice calls sent to fax extension

I have a situation that has repeated itself a few times.  Someone calls
into Asterisk and is connected with a voice extension.  At some point
during the call, the log shows "chan_zap.c: DTMF digit: f on Zap/2-1".
At this point, the call is redirected to receive a fax and the Asterisk
voice extension is hung up.  The users report that there were no
noticable tones heard just before the cutover, so I'm not sure what's
going on.  Is there a way to disable detection of faxes after the
voicecall is initiated?  We're running a Digium card to convert our
analog trunks if that makes any difference.

Thanks!

Paul
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --

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

RE: [Asterisk-Users] RE: Voice calls sent to fax extension

[Colin Anderson]

yes. Wind whistling in a car, female voices at a particular pitch and
volume, fax machine running in the background of a voice call with the
speaker on. It happens. Whether this is a problem or not depends on your
pain threshold. I get a couple reports a week, which means that it actually
happens ten times a couple times a week, so twenty times a week, and I
process ~20K calls a week, so it happens to me .1 % of the time. Is this a
problem for me? Nah. Is it a problem for you? Maybe - what's your pain
threshold?

ps fwiw, this behavior will happen with any device that listens inline for a
CNG tone, so it's not just an Asterisk thing

-Original Message-
From: Paul A. Pringle [mailto:[EMAIL PROTECTED]
Sent: Monday, June 26, 2006 8:54 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] RE: Voice calls sent to fax extension


I thought it might be an inadvertent button press, but none of the keys
(on my phone at least) are recognized by Asterisk as fax tones.  This
has happened to two different users getting calls from different people
using different equipment.  Does anyone else see this behavior
occasionally?

Paul

-Original Message-
Date: Fri, 23 Jun 2006 15:29:31 -0400
From: "Bill Gibbs" <[EMAIL PROTECTED]>
Subject: RE: [Asterisk-Users] Voice calls sent to fax extension
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Message-ID:
<[EMAIL PROTECTED]>
Content-Type: text/plain;   charset="us-ascii"

Maybe their fat jowls hit a few buttons on the keypad and sent the fax
tone down the line and they didn't realize it?

Bill

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul A.
Pringle
Sent: Friday, June 23, 2006 2:51 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Voice calls sent to fax extension

I have a situation that has repeated itself a few times.  Someone calls
into Asterisk and is connected with a voice extension.  At some point
during the call, the log shows "chan_zap.c: DTMF digit: f on Zap/2-1".
At this point, the call is redirected to receive a fax and the Asterisk
voice extension is hung up.  The users report that there were no
noticable tones heard just before the cutover, so I'm not sure what's
going on.  Is there a way to disable detection of faxes after the
voicecall is initiated?  We're running a Digium card to convert our
analog trunks if that makes any difference.

Thanks!

Paul
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [Asterisk-Users] RE: Voice calls sent to fax extension

[Lee Howard]

Paul A. Pringle wrote:


I thought it might be an inadvertent button press, but none of the keys
(on my phone at least) are recognized by Asterisk as fax tones.  This
has happened to two different users getting calls from different people
using different equipment.  Does anyone else see this behavior
occasionally?



Yes, which is why I disable faxdetect entirely.  My sister-in-law was 
constantly being detected as a fax machine several minutes into 
conversations with my wife.  As funny as that may seem at first ... 
those two eventually make it a not-so-funny situation for me.


The fax detection should, in theory, only be looking for CNG tones and 
should, in theory, only be looking for them during the first several 
seconds of a call.  Analogue DTMF tones should not be detectable as CNG, 
in theory.


Lee.
___
--Bandwidth and Colocation provided by Easynews.com --

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

[Asterisk-Users] Re: Voice calls sent to fax extension

[Steven]

I do get random DTMF tones.
They have been to sparse to diagnose if there was anything common with those 
calls.

When it is noticed and I look it up in the logs, it may be any digits.

We see this on zap(PRI) to zap(PRI) bridged calls too.

We are using a TE411P.



-- 
-- 
Steven

http://www.glimasoutheast.org



"Paul A. Pringle" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED]
I thought it might be an inadvertent button press, but none of the keys
(on my phone at least) are recognized by Asterisk as fax tones.  This
has happened to two different users getting calls from different people
using different equipment.  Does anyone else see this behavior
occasionally?

Paul

-Original Message-
Date: Fri, 23 Jun 2006 15:29:31 -0400
From: "Bill Gibbs" <[EMAIL PROTECTED]>
Subject: RE: [Asterisk-Users] Voice calls sent to fax extension
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Message-ID:
<[EMAIL PROTECTED]>
Content-Type: text/plain; charset="us-ascii"

Maybe their fat jowls hit a few buttons on the keypad and sent the fax
tone down the line and they didn't realize it?

Bill

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul A.
Pringle
Sent: Friday, June 23, 2006 2:51 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Voice calls sent to fax extension

I have a situation that has repeated itself a few times.  Someone calls
into Asterisk and is connected with a voice extension.  At some point
during the call, the log shows "chan_zap.c: DTMF digit: f on Zap/2-1".
At this point, the call is redirected to receive a fax and the Asterisk
voice extension is hung up.  The users report that there were no
noticable tones heard just before the cutover, so I'm not sure what's
going on.  Is there a way to disable detection of faxes after the
voicecall is initiated?  We're running a Digium card to convert our
analog trunks if that makes any difference.

Thanks!

Paul
___
--Bandwidth and Colocation provided by Easynews.com --

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



___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [Asterisk-Users] caller id

[sdgesa gaeharth]

I am not sure, I will check.  If I dont', and get it started, will it just start working? If not, what do I need to do?ThanksJoshua West <[EMAIL PROTECTED]> wrote:  Do you have the Caller ID feature with your telephone service package?sdgesa gaeharth wrote:> How can I get the external caller id to show on the polycom 501> phones. Currently, when someone calls our office, we only see the word> "asterisk" in the caller id.>> This is our set up:>> VOIP(polycom)<--->Asterisk 1.2.4<--->PSTN>> Thanks>> > Yahoo! Groups gets better. Check out the new email design.> > Plus there’s much more to come.> >> ___> --Bandwidth and Colocation provided by Easynews.com -->> Asterisk-Users mailing list> To UNSUBSCRIBE or update options visit:>http://lists.digium.com/mailman/listinfo/asterisk-users>   -- Joshua WestLinux Infrastructure EngineerBoston Engineering Corporationhttp://www.boston-engineering.com___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users 
		Do you Yahoo!? Next-gen email? Have it all with the  all-new Yahoo! Mail Beta.___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [Asterisk-Users] SE Michigan asterisk users group

[Jon Radon]

I'm also in the area, near Southfield.  I'd be interested as well.
On 6/22/06, BerkHolz, Steven <[EMAIL PROTECTED]> wrote:
I am thinking of getting an asterisk user group together for either SEMichigan or just Metro-Detroit.
How much interest in asterisk in Michigan is there on this list?I am already on the board of glimasoutheast, with is a group fortechnology professionals. (very broad range)It is a spin-off from Automation Alley, which is SE Michigan's version
of Silicone Valley.--Stevenhttp://www.glimasoutheast.org___--Bandwidth and Colocation provided by 
Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:  http://lists.digium.com/mailman/listinfo/asterisk-users
-- Is it something someone said, was it something someone said? 
___
--Bandwidth and Colocation provided by Easynews.com --

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

[Asterisk-Users] Is there a way to reinstall the AMP

[Yrving Rivas]

Hi all.

Today I have tried to connect to the AMP with http://myserverip but I can
not connect to the AMP (it sends me out of my network).
What would be happening?.
The last thing I did is to try to change the digital receptionist manually.
Is there a way to re-install the amp?

Thanks

__
Correo Yahoo!
Espacio para todos tus mensajes, antivirus y antispam ¡gratis! 
Regístrate ya - http://correo.yahoo.com.mx/ 
___
--Bandwidth and Colocation provided by Easynews.com --

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

[Asterisk-Users] RE: Voice calls sent to fax extension

[Paul A. Pringle]

I thought it might be an inadvertent button press, but none of the keys
(on my phone at least) are recognized by Asterisk as fax tones.  This
has happened to two different users getting calls from different people
using different equipment.  Does anyone else see this behavior
occasionally?

Paul

-Original Message-
Date: Fri, 23 Jun 2006 15:29:31 -0400
From: "Bill Gibbs" <[EMAIL PROTECTED]>
Subject: RE: [Asterisk-Users] Voice calls sent to fax extension
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Message-ID:
<[EMAIL PROTECTED]>
Content-Type: text/plain;   charset="us-ascii"

Maybe their fat jowls hit a few buttons on the keypad and sent the fax
tone down the line and they didn't realize it?

Bill

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul A.
Pringle
Sent: Friday, June 23, 2006 2:51 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Voice calls sent to fax extension

I have a situation that has repeated itself a few times.  Someone calls
into Asterisk and is connected with a voice extension.  At some point
during the call, the log shows "chan_zap.c: DTMF digit: f on Zap/2-1".
At this point, the call is redirected to receive a fax and the Asterisk
voice extension is hung up.  The users report that there were no
noticable tones heard just before the cutover, so I'm not sure what's
going on.  Is there a way to disable detection of faxes after the
voicecall is initiated?  We're running a Digium card to convert our
analog trunks if that makes any difference.

Thanks!

Paul
___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [Asterisk-Users] 1.2.9.1 SIP/Local/Queue behaviours weird

[C F]

I have seen this when Polycom has to communicate with none polycom
phones and a transfer is initiated to a polycom, unless the Polycom
presses Hold and then unhold, there is only one way audio, this is
without NAT involved. There might also be other cases when this
happens. My workaround is to add canreinvite=no


On 6/26/06, Isaac Xiao <[EMAIL PROTECTED]> wrote:





Hi,



Does any one experience that SIP phone to SIP phone (Polycom phone) calls
can't hear each other, but Monitor application records both end's voices. It
also happens in group pickup calls. Zap calls to queue (Local channel) also
experience this problem (sometimes, our SIP phone can't hear any voice from
incoming Zap calls when pickup, sometimes this happens after 10-50 seconds'
talk). It is weird.



Jun 26 16:53:35 VERBOSE[8290] logger.c: -- Executing
Dial("Local/[EMAIL PROTECTED],2", "SIP/7188|30|trWwT") in new stack
 Jun 26 16:53:35 DEBUG[8290] chan_sip.c: Setting NAT on RTP to 0
 Jun 26 16:53:35 DEBUG[8290] chan_sip.c: Setting NAT on VRTP to 0
 Jun 26 16:53:35 DEBUG[8290] chan_sip.c: Outgoing Call for 7188
 Jun 26 16:53:35 VERBOSE[8290] logger.c: -- Called 7188
 Jun 26 16:53:35 VERBOSE[8287] logger.c: -- Local/[EMAIL PROTECTED],1
is ringing
 Jun 26 16:53:35 DEBUG[2966] chan_sip.c: (Provisional) Stopping
retransmission (but retaining packet) on
'[EMAIL PROTECTED]' Request
102: Found
 Jun 26 16:53:35 DEBUG[2966] chan_sip.c: (Provisional) Stopping
retransmission (but retaining packet) on
'[EMAIL PROTECTED]' Request
102: Found
 Jun 26 16:53:35 DEBUG[2957] channel.c: Avoiding initial deadlock for
'SIP/7188-6b1f'
 Jun 26 16:53:35 VERBOSE[8290] logger.c: -- SIP/7188-6b1f is ringing
 Jun 26 16:53:37 DEBUG[2966] chan_sip.c: Acked pending invite 102
 Jun 26 16:53:37 DEBUG[2966] chan_sip.c: Stopping retransmission on
'[EMAIL PROTECTED]' of Request
102: Match Found
 Jun 26 16:53:37 DEBUG[2966] chan_sip.c: build_route: Contact hop:
 Jun 26 16:53:37 VERBOSE[8290] logger.c: -- SIP/7188-6b1f answered
Local/[EMAIL PROTECTED],2
 Jun 26 16:53:37 DEBUG[8287] app_queue.c: Dunno what to do with control type
-1
 Jun 26 16:53:37 VERBOSE[8287] logger.c: -- Local/[EMAIL PROTECTED],1
answered Zap/13-1
 Jun 26 16:53:37 DEBUG[8287] chan_zap.c: Set option TONE VERIFY, mode:
MUTECONF(1) on Zap/13-1
 Jun 26 16:53:37 VERBOSE[8287] logger.c: -- Stopped music on hold on
Zap/13-1
 Jun 26 16:53:37 DEBUG[8287] channel.c: Scheduling timer at 0 sample
intervals
 Jun 26 16:54:02 DEBUG[8290] channel.c: Didn't get a frame from channel:
SIP/7188-6b1f
 Jun 26 16:54:02 DEBUG[8290] channel.c: Bridge stops bridging channels
Local/[EMAIL PROTECTED],2 and SIP/7188-6b1f
 Jun 26 16:54:02 DEBUG[8290] chan_sip.c: update_call_counter(7188) -
decrement call limit counter
 Jun 26 16:54:02 DEBUG[8290] app_dial.c: Exiting with DIALSTATUS=ANSWER.
 Jun 26 16:54:02 VERBOSE[8290] logger.c: == Spawn extension (macro-dial, s,
10) exited non-zero on 'Local/[EMAIL PROTECTED],2' in macro 'dial'
 Jun 26 16:54:02 VERBOSE[8290] logger.c: == Spawn extension (macro-dial, s,
10) exited non-zero on 'Local/[EMAIL PROTECTED],2' in macro 'exten-vm'
 Jun 26 16:54:02 VERBOSE[8290] logger.c: == Spawn extension (macro-dial, s,
10) exited non-zero on 'Local/[EMAIL PROTECTED],2'
 Jun 26 16:54:02 DEBUG[8290] res_monitor.c: monitor executing ( nice -n 19
soxmix
"/var/spool/asterisk/monitor/20060626-165333-1151304813.901-in.gsm"
"/var/spool/asterisk/monitor/20060626-165333-1151304813.901-out.gsm"
"/var/spool/asterisk/monitor/20060626-165333-1151304813.901.gsm"
&& rm -f
"/var/spool/asterisk/monitor/20060626-165333-1151304813.901-"*
) &
 Jun 26 16:54:02 DEBUG[8287] channel.c: Didn't get a frame from channel:
Local/[EMAIL PROTECTED],1
 Jun 26 16:54:02 DEBUG[8287] channel.c: Bridge stops bridging channels
Zap/13-1 and Local/[EMAIL PROTECTED],1
 Jun 26 16:54:02 VERBOSE[8287] logger.c: == Spawn extension (ext-queues,
7141, 6) exited non-zero on 'Zap/13-1'
 Jun 26 16:54:02 DEBUG[8287] chan_zap.c: Set option AUDIO MODE, value: ON(1)
on Zap/13-1
 Jun 26 16:54:02 DEBUG[8287] chan_zap.c: Hangup: channel: 13 index = 0,
normal = 27, callwait = -1, thirdcall = -1
 Jun 26 16:54:02 DEBUG[8287] chan_zap.c: Not yet hungup... Calling hangup
once with icause, and clearing call
 Jun 26 16:54:02 DEBUG[8287] chan_zap.c: disabled echo cancellation on
channel 13
 Jun 26 16:54:02 DEBUG[8287] chan_zap.c: Set option TDD MODE, value: OFF(0)
on Zap/13-1
 Jun 26 16:54:02 DEBUG[8287] chan_zap.c: Updated conferencing on 13, with 0
conference users
 Jun 26 16:54:02 DEBUG[8287] chan_zap.c: Set option AUDIO MODE, value:
OFF(0) on Zap/13-1
 Jun 26 16:54:02 DEBUG[8287] chan_zap.c: disabled echo cancellation on
channel 13
 Jun 26 16:54:02 VERBOSE[8287] logger.c: -- Hungup 'Zap/13-1'



Isaac Xiao


__

RE: [Asterisk-Users] SE Michigan asterisk users group

[Carlos Alperin]

I live in Southfield, our main office is in Pontiac, but our Colo is in
Southfield.



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael George
Sent: Monday, June 26, 2006 8:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: [EMAIL PROTECTED]; [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] SE Michigan asterisk users group

Our main office is near Lansing, but we have a person who lives in the AA
are that would like to attend such a group.

On Thu, Jun 22, 2006 at 04:27:02PM -0400, BerkHolz, Steven wrote:
> I am thinking of getting an asterisk user group together for either SE 
> Michigan or just Metro-Detroit.
> 
> How much interest in asterisk in Michigan is there on this list?
> 
> I am already on the board of glimasoutheast, with is a group for 
> technology professionals. (very broad range) It is a spin-off from 
> Automation Alley, which is SE Michigan's version of Silicone Valley.

--
-M

There are 10 kinds of people in this world:
Those who can count in binary and those who cannot.
___
--Bandwidth and Colocation provided by Easynews.com --

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

___
--Bandwidth and Colocation provided by Easynews.com --

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

[Asterisk-Users] 1.2.9.1 SIP/Local/Queue behaviours weird

[Isaac Xiao]

Hi,

 

Does any one experience that SIP phone to SIP phone
(Polycom phone) calls can’t hear each other, but Monitor application
records both end’s voices. It also happens in group pickup calls. Zap
calls to queue (Local channel) also experience this problem (sometimes, our SIP
phone can’t hear any voice from incoming Zap calls when pickup, sometimes
this happens after 10-50 seconds’ talk). It is weird.

 

Jun 26 16:53:35 VERBOSE[8290] logger.c: -- Executing
Dial("Local/[EMAIL PROTECTED],2",
"SIP/7188|30|trWwT") in new stack
Jun 26 16:53:35 DEBUG[8290] chan_sip.c: Setting NAT on RTP to 0
Jun 26 16:53:35 DEBUG[8290] chan_sip.c: Setting NAT on VRTP to 0
Jun 26 16:53:35 DEBUG[8290] chan_sip.c: Outgoing Call for 7188
Jun 26 16:53:35 VERBOSE[8290] logger.c: -- Called 7188
Jun 26 16:53:35 VERBOSE[8287] logger.c: -- Local/[EMAIL PROTECTED],1 is
ringing
Jun 26 16:53:35 DEBUG[2966] chan_sip.c: (Provisional) Stopping retransmission
(but retaining packet) on '[EMAIL PROTECTED]'
Request 102: Found
Jun 26 16:53:35 DEBUG[2966] chan_sip.c: (Provisional) Stopping retransmission
(but retaining packet) on '[EMAIL PROTECTED]'
Request 102: Found
Jun 26 16:53:35 DEBUG[2957] channel.c: Avoiding initial deadlock for
'SIP/7188-6b1f'
Jun 26 16:53:35 VERBOSE[8290] logger.c: -- SIP/7188-6b1f is ringing
Jun 26 16:53:37 DEBUG[2966] chan_sip.c: Acked pending invite 102
Jun 26 16:53:37 DEBUG[2966] chan_sip.c: Stopping retransmission on
'[EMAIL PROTECTED]' of Request 102: Match Found
Jun 26 16:53:37 DEBUG[2966] chan_sip.c: build_route: Contact hop: 
Jun 26 16:53:37 VERBOSE[8290] logger.c: --
SIP/7188-6b1f answered Local/[EMAIL PROTECTED],2
Jun 26 16:53:37 DEBUG[8287] app_queue.c: Dunno what to do with control type -1
Jun 26 16:53:37 VERBOSE[8287] logger.c: -- Local/[EMAIL PROTECTED],1
answered Zap/13-1
Jun 26 16:53:37 DEBUG[8287] chan_zap.c: Set option TONE VERIFY, mode:
MUTECONF(1) on Zap/13-1
Jun 26 16:53:37 VERBOSE[8287] logger.c: -- Stopped music on hold on Zap/13-1
Jun 26 16:53:37 DEBUG[8287] channel.c: Scheduling timer at 0 sample intervals
Jun 26 16:54:02 DEBUG[8290] channel.c: Didn't get a frame from channel:
SIP/7188-6b1f
Jun 26 16:54:02 DEBUG[8290] channel.c: Bridge stops bridging channels
Local/[EMAIL PROTECTED],2 and SIP/7188-6b1f
Jun 26 16:54:02 DEBUG[8290] chan_sip.c: update_call_counter(7188) - decrement
call limit counter
Jun 26 16:54:02 DEBUG[8290] app_dial.c: Exiting with DIALSTATUS=ANSWER.
Jun 26 16:54:02 VERBOSE[8290] logger.c: == Spawn extension (macro-dial, s, 10)
exited non-zero on 'Local/[EMAIL PROTECTED],2' in macro 'dial'
Jun 26 16:54:02 VERBOSE[8290] logger.c: == Spawn extension (macro-dial, s, 10)
exited non-zero on 'Local/[EMAIL PROTECTED],2' in macro 'exten-vm'
Jun 26 16:54:02 VERBOSE[8290] logger.c: == Spawn extension (macro-dial, s, 10)
exited non-zero on 'Local/[EMAIL PROTECTED],2'
Jun 26 16:54:02 DEBUG[8290] res_monitor.c: monitor executing ( nice -n 19
soxmix
"/var/spool/asterisk/monitor/20060626-165333-1151304813.901-in.gsm"
"/var/spool/asterisk/monitor/20060626-165333-1151304813.901-out.gsm"
"/var/spool/asterisk/monitor/20060626-165333-1151304813.901.gsm"
&& rm -f
"/var/spool/asterisk/monitor/20060626-165333-1151304813.901-"* )
&
Jun 26 16:54:02 DEBUG[8287] channel.c: Didn't get a frame from channel:
Local/[EMAIL PROTECTED],1
Jun 26 16:54:02 DEBUG[8287] channel.c: Bridge stops bridging channels Zap/13-1
and Local/[EMAIL PROTECTED],1
Jun 26 16:54:02 VERBOSE[8287] logger.c: == Spawn extension (ext-queues, 7141,
6) exited non-zero on 'Zap/13-1'
Jun 26 16:54:02 DEBUG[8287] chan_zap.c: Set option AUDIO MODE, value: ON(1) on
Zap/13-1
Jun 26 16:54:02 DEBUG[8287] chan_zap.c: Hangup: channel: 13 index = 0, normal =
27, callwait = -1, thirdcall = -1
Jun 26 16:54:02 DEBUG[8287] chan_zap.c: Not yet hungup... Calling hangup once
with icause, and clearing call
Jun 26 16:54:02 DEBUG[8287] chan_zap.c: disabled echo cancellation on channel
13
Jun 26 16:54:02 DEBUG[8287] chan_zap.c: Set option TDD MODE, value: OFF(0) on
Zap/13-1
Jun 26 16:54:02 DEBUG[8287] chan_zap.c: Updated conferencing on 13, with 0
conference users
Jun 26 16:54:02 DEBUG[8287] chan_zap.c: Set option AUDIO MODE, value: OFF(0) on
Zap/13-1
Jun 26 16:54:02 DEBUG[8287] chan_zap.c: disabled echo cancellation on channel
13
Jun 26 16:54:02 VERBOSE[8287] logger.c: -- Hungup 'Zap/13-1'



 

Isaac Xiao

 






___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [Asterisk-Users] Asterisk ACD with Polycom IP501

[BJ Weschke]

Hi Dean -

It should be working. If not, please email me a sip debug trace along
with your /etc/asterisk/agents.conf and your /etc/asterisk/sip.conf.

Thanks.

BJ

On 6/26/06, Dean @ INKnBITs <[EMAIL PROTECTED]> wrote:

Hi,

Has anybody got the polycom acd function to work? I have the following
setup:

Debian 3.1 - 2.6.8 linux
zlib-1.1.4
libpri-1.2.3
zaptel- 1.2.6
Asterisk - the bweschke/polycom_acd_funtions branch version - I get one
error when doing a make install about needing a newer version of libpri and
zaptel, I got the above versions from asterisk.org, are there newer version
anywhere else?

In the sip.conf file I have set the agentlogin=yes and agentcbcontext=demo
(demo as from extensions.conf context)

I have setup an agent in agents.conf as ,1234,Name

I have changed in the sip.cfg of the polycom phone:
feature.15.name="acd-login-logout" feature.15.enabled="1"
feature.16.name="acd-agent-availability" feature.16.enabled="1"

and in the phone1.cfg of the polycom I'm only using line1 so made the
changes below:
reg.1.acd-login-logout="1"
reg.1.acd-agent-available="1"


I get the login button on the phone, and when I try and login with the 
agent it just goes back to login.


Any help would be appreciated.

Thanks,
Dean Bath




___
--Bandwidth and Colocation provided by Easynews.com --

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




--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
___
--Bandwidth and Colocation provided by Easynews.com --

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

RE: [Asterisk-Users] Asterisk Startups

[Douglas Garstang]

Yeah, that's what I like about Oz. Everyone knows everyone... miss you guys too!

> -Original Message-
> From: Rob Thomas [mailto:[EMAIL PROTECTED]
> Sent: Monday, June 26, 2006 2:58 AM
> To: asterisk-users
> Subject: RE: [Asterisk-Users] Asterisk Startups
> 
> 
> Well now would be a great time to come back, Doug! We miss you! 8)
> 
> --Rob
> 
> 
> > -Original Message-
> > From: Douglas Garstang [mailto:[EMAIL PROTECTED] 
> > Sent: Monday, 26 June 2006 3:22 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion; 
> > Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: RE: [Asterisk-Users] Asterisk Startups
> > 
> > 
> > Paul,
> >  
> > D'oh. The fact I left Sydney 5 years ago for the US might be 
> > a teeny complication. :P
> >  
> > Doug.
> > 
> > -Original Message- 
> > From: Paul Hales [mailto:[EMAIL PROTECTED] 
> > Sent: Sun 6/25/2006 11:01 PM 
> > To: Asterisk Users Mailing List - Non-Commercial Discussion 
> > Cc: 
> > Subject: Re: [Asterisk-Users] Asterisk Startups
> > 
> > 
> > 
> > Douglas Garstang wrote:
> > > Does anyone know of any startups using Asterisk? What 
> > about established companies? Ones that are hiring would be 
> nice :)
> > > 
> > > Doug.
> > > 
> > > 
> > > 
> > > 
> > >  
> > > 
> > --
> > --
> > >
> > > ___
> > > --Bandwidth and Colocation provided by Easynews.com --
> > >
> > > Asterisk-Users mailing list
> > > To UNSUBSCRIBE or update options visit:
> > >http://lists.digium.com/mailman/listinfo/asterisk-users
> > >  
> > We are always looking for good people - here in Melbourne.
> > 
> > PaulH
> > 
> > --
> > Paul Hales
> > Technical Manager
> > AsteriskIT
> > www.asteriskit.com.au
> > bus: 03 8320 8100
> > mob: 0434 673 529
> > 
> > ___
> > --Bandwidth and Colocation provided by Easynews.com --
> > 
> > Asterisk-Users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> > 
> > 
> > 
> ___
> --Bandwidth and Colocation provided by Easynews.com --
> 
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
> 
___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [Asterisk-Users] best hardphone for Asterisk?

[Doug Crompton]

I guess I did not make my point clearly enough. I already do have just
that. An spa-3000 with ALL internal analog phones on it's on FXO. But this
gives just ONE extension for all phones. Yes I could get more FXS's and
run seperate wires.

So with that background what would be nice is a wireless device like the
Panasonic cordless with one base and multiple phone capability that
connected via ethernet and serves the phones. Just wishful thinking. I
will stick with what I have until something useful, sylish, and less
expensive arrives.

Doug


On Mon, 26 Jun 2006, Michael George wrote:

> On Mon, Jun 26, 2006 at 12:08:48AM -0400, Doug Crompton wrote:
> > Still awfully pricey for home use and the styling is not there for a
> > bedroom or many other areas of a modern home. What we need is a wireless
> > sip phone modeled like the panasonic or uniden which allow multiple
> > extension off of one base. The base would connect to the internet. The
> > other problem is many of these phones require power, so even if you have
> > backup for your central system the phone still needs to be on it. Power
> > over ethernet would help.
>
> 1. If you have *, you don't necessarily need multiple handsets off of one
>   base.
> 2. Cordless phones also require power
> 3. If the multi-handset cordless phone does suit your needs best, then
>   get a SIP ATA device like a Sipura or IAXy and you should have your
>   needs met.
>
> --
> -M
>
> There are 10 kinds of people in this world:
>   Those who can count in binary and those who cannot.
> ___
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>


"Those that sacrifice essential liberty to obtain a little temporary safety
 deserve neither liberty nor safety."  -- Ben Franklin (1759)


*  Doug Crompton   *
*  Richboro, PA 18954  *
*  215-431-6307*
*  *
* [EMAIL PROTECTED]*
* http://www.crompton.com  *



___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [Asterisk-Users] Asterisk-1.2.9.1 with Siemens HiPath 4000

[richard Coco]

Hi Josué

if the Siemens phone calls Asterisk, it didn't get a
dial tone from Asterisk? Is it correct?

if yes, this is depending of Asterisk which didn't
generates a ringback messages as it expexts dial ton
generation localy. So try this workaround for HiPath
local dial ton generation:
-> Add option TR6Q(TRGT) to the class of trunk (COT)
parameters

hope it will help...

rich





--- Josué Conti <[EMAIL PROTECTED]> wrote:

>   Hello all.
>  I have installed and functioning asterisk-1.2.9.1
> where I effected one
> upgrade in asterisk-1.0.9, is interconnected with a
> PABX Siemens HiPath 4000
> in ISDN PRI with protocol QSIG, the one that is
> happening he is that the
> calls originated for PABX Siemens and destined to
> SIP phones asterisk are
> being without audio, nor Ring, is dumb. They could
> help in this case me?
> Best Regards
> 
> Josué
> > ___
> --Bandwidth and Colocation provided by Easynews.com
> --
> 
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>   
>
http://lists.digium.com/mailman/listinfo/asterisk-users
> 


__
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 
___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [Asterisk-Users] This is getting really annoying - re: POSTFIX

[Warren]

Michiel van Baak wrote:

  On 08:51, Mon 26 Jun 06, Matt wrote:
  
  
What on earth is going on with the list?!?!   Some of my messages
never make it... then days later I get something like this back:


Final-Recipient: rfc822; asterisk-users@lists.digium.com
Action: failed
Status: 5.0.0
Diagnostic-Code: X-Postfix; mail forwarding loop for
  asterisk-users@lists.digium.com

  
  

Thank god!
I have been looking and grepping and reconfiguring my
postfix during the last week because of this messages.
I now know it's really not me.

  

Nope - me too.  Some of my messages make it, but all get bounced, even
the ones that do make it.

W


___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [Asterisk-Users] This is getting really annoying - re: POSTFIX

[Josué Conti]

Hello All.

Accurately, my messages also are not being received for the list and the traffic of messages really is very low. It will be a problem of the list?    Best Regards
Josué 
2006/6/26, Michiel van Baak <[EMAIL PROTECTED]>:
On 08:51, Mon 26 Jun 06, Matt wrote:> What on earth is going on with the list?!?!   Some of my messages
> never make it... then days later I get something like this back:>>> Final-Recipient: rfc822; asterisk-users@lists.digium.com> Action: failed
> Status: 5.0.0> Diagnostic-Code: X-Postfix; mail forwarding loop for>   asterisk-users@lists.digium.comThank god!I have been looking and grepping and reconfiguring my
postfix during the last week because of this messages.I now know it's really not me.--Michiel van Baak[EMAIL PROTECTED]
http://michiel.vanbaak.euGnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD"Why is it drug addicts and computer afficionados are both called users?"
___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [Asterisk-Users] SE Michigan asterisk users group

[Tom Hayden]

Count me in, my office is in Livonia, but I currently reside in the D.  Someone should set up a mailing list for this.--Tom HaydenOn 6/26/06, Michael George
 <[EMAIL PROTECTED]> wrote:Our main office is near Lansing, but we have a person who lives in the
AA are that would like to attend such a group.On Thu, Jun 22, 2006 at 04:27:02PM -0400, BerkHolz, Steven wrote:> I am thinking of getting an asterisk user group together for either SE> Michigan or just Metro-Detroit.
>> How much interest in asterisk in Michigan is there on this list?>> I am already on the board of glimasoutheast, with is a group for> technology professionals. (very broad range)> It is a spin-off from Automation Alley, which is SE Michigan's version
> of Silicone Valley.---MThere are 10 kinds of people in this world:Those who can count in binary and those who cannot.___--Bandwidth and Colocation provided by 
Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation provided by Easynews.com --

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

[Asterisk-Users] struggling with the "g" flag

[Julian Lyndon-Smith]

If I have in my dialplan

[AgentQ]
exten => _XX.,1,Dial(Sip/{$exten},120,g)
exten => _XX.,2,NoOP(here we are)

where [AgentQ] is called by the queue command to a member added by

addqueuemember(Local/[EMAIL PROTECTED])

why don't I get to the NoOp if the agent hangs up during the 
announcement message (to the agent) ?


I see in the app_dial.c program that the "g" flag is tested thus:

if ((ast_test_flag(peerflags, OPT_GO_ON)) && (!chan->_softhangup) && 
(res != AST_PBX_KEEPALIVE))

res = 0;

So this would indicate that if all three of these conditions are met 
then res would be set to 0, and things would behave how I want.


In chan_agent.c, the following line is where the agent has hung up

if (peer->_softhangup) {
/* Agent must have hung up */
ast_log(LOG_WARNING, "Agent on %s hungup on the customer.  They're 
going to be pissed.\n", peer->name);


I see that in chan_agent the peer->_softhangup is true (I get the 
message on the console) but the test in app_dial specifically tests to 
see if chan->_softhangup is *not* true.


Why is that ?

Julian
___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [Asterisk-Users] PCI or MiniPCI Hardware DSP for G.729, G.723.1 and/or GSM

[Andrew Kohlsmith]

On Saturday 24 June 2006 09:44, Paul Hewlett wrote:
> I would imagine that this would not solve any problems - the extra overhead
> of piping the data over the PCI bus would very quickly negate any speed
> gains of the DSP over the native Intel FPU. Additionally you would probably
> introduce extra latency. I did quite a lot of work on DSP co-processor
> boards and there was always a considerable startup time when all the data
> pipes had to be filled.

Talk to Digium, they're releasing a PCI DSP board that does exactly this.  I 
guess that the PCI overhead isn't as great as was first thought?  Interesting 
times...

-A.
___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [Asterisk-Users] This is getting really annoying - re: POSTFIX

[Michiel van Baak]

On 08:51, Mon 26 Jun 06, Matt wrote:
> What on earth is going on with the list?!?!   Some of my messages
> never make it... then days later I get something like this back:
> 
> 
> Final-Recipient: rfc822; asterisk-users@lists.digium.com
> Action: failed
> Status: 5.0.0
> Diagnostic-Code: X-Postfix; mail forwarding loop for
>   asterisk-users@lists.digium.com


Thank god!
I have been looking and grepping and reconfiguring my
postfix during the last week because of this messages.
I now know it's really not me.

-- 
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?"

___
--Bandwidth and Colocation provided by Easynews.com --

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

[Asterisk-Users] Asterisk ACD with Polycom IP501

[Dean @ INKnBITs]

Hi,

Has anybody got the polycom acd function to work? I have the following
setup:

Debian 3.1 - 2.6.8 linux
zlib-1.1.4
libpri-1.2.3
zaptel- 1.2.6
Asterisk - the bweschke/polycom_acd_funtions branch version - I get one
error when doing a make install about needing a newer version of libpri and
zaptel, I got the above versions from asterisk.org, are there newer version
anywhere else?

In the sip.conf file I have set the agentlogin=yes and agentcbcontext=demo
(demo as from extensions.conf context)

I have setup an agent in agents.conf as ,1234,Name

I have changed in the sip.cfg of the polycom phone:
feature.15.name="acd-login-logout" feature.15.enabled="1"
feature.16.name="acd-agent-availability" feature.16.enabled="1"

and in the phone1.cfg of the polycom I'm only using line1 so made the
changes below:
reg.1.acd-login-logout="1"
reg.1.acd-agent-available="1"


I get the login button on the phone, and when I try and login with the 
agent it just goes back to login.


Any help would be appreciated.

Thanks,
Dean Bath




___
--Bandwidth and Colocation provided by Easynews.com --

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