Re: [Asterisk-Users] Good W2K softphone

[God Knows Well]

Hi

U can visit the http://iaxclient.sf.net for some opensource underdevelopment 
softphones.

Take Care

Obaid Amin Syed

From: Chris Albertson [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Good W2K softphone
Date: Fri, 3 Oct 2003 23:00:13 -0700 (PDT)
I haven't found any open source/freeware software phones that run under
Windows 2000 that I like or that even work well. What are other people 
using?

=
Chris Albertson
  Home:   310-376-1029  [EMAIL PROTECTED]
  Cell:   310-990-7550
  Office: 310-336-5189  [EMAIL PROTECTED]
  KG6OMK
__
Do you Yahoo!?
The New Yahoo! Shopping - with improved product search
http://shopping.yahoo.com
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
_
The new MSN 8: advanced junk mail protection and 2 months FREE* 
http://join.msn.com/?page=features/junkmail

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: DB virtualization for multiple database support - Was Re: [Asterisk-Users] How to use vmdb.sql in voicemail.conf/extension.conf

[Chris Albertson]

I think ODBC is the way to go.  There is really nothing to write.


  You can't anymore MySQL was ripped from Asterisk because the client
 libs
  are GPL.
 
 I would be more than happy to help write a DB Virtualization
 function for *.
 
 I *love* the way it works in Java, but that's not a real possibility.
  It wouldn't need to be as complicated as JDBC but it's a nice model.



=
Chris Albertson
  Home:   310-376-1029  [EMAIL PROTECTED]
  Cell:   310-990-7550
  Office: 310-336-5189  [EMAIL PROTECTED]
  KG6OMK

__
Do you Yahoo!?
The New Yahoo! Shopping - with improved product search
http://shopping.yahoo.com
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Anyone else use Audacity for prompts?

[Brian Capouch]

I am using Audacity to record some voice prompts.

The .wav files I'm producing are of stellar quality.  However, once I 
turn them into .gsm, they sound buzzy and muffled.

I know that some of this comes with the territory, but I wonder if there 
is anyone out there who does this routinely, and who can advise me as to 
the MO I could use that results in the highest quality in the resulting 
playback files.

Thanks.

B.

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Anyone else use Audacity for prompts?

[Shaun Ewing]

- Original Message - 
From: Brian Capouch [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, October 06, 2003 5:21 PM
Subject: [Asterisk-Users] Anyone else use Audacity for prompts?


 I am using Audacity to record some voice prompts.

 The .wav files I'm producing are of stellar quality.  However, once I
 turn them into .gsm, they sound buzzy and muffled.

 I know that some of this comes with the territory, but I wonder if there
 is anyone out there who does this routinely, and who can advise me as to
 the MO I could use that results in the highest quality in the resulting
 playback files.

 Thanks.

 B.

What are you using to convert the wav files to gsm? I've been using 'sox'
under Linux and have had no quality issues whatsoever.

An example line to convert:

sox file.wav -r 8000 -c 1 file.gsm

-Shaun

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Anyone else use Audacity for prompts?

[Brian Capouch]

Shaun Ewing wrote:
I know that some of this comes with the territory, but I wonder if there
is anyone out there who does this ( wav - gsm) routinely, and who can advise me as to
the MO I could use that results in the highest quality in the resulting
playback files.
What are you using to convert the wav files to gsm? I've been using 'sox'
under Linux and have had no quality issues whatsoever.
An example line to convert:

sox file.wav -r 8000 -c 1 file.gsm

That's the exact command I'm using, but it sure does sound crappy to my 
ears.

Perhaps that's the best I can do w/gsm, and of course I expect once the 
sound is sent through the PSTN most of the highs and bottom are gone 
anyways.

I was just hoping there was something I could do to make the resulting 
files a bit clearer.

Thx.

B.

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Anyone else use Audacity for prompts?

[Michael T Farnworth]

I have had been recording my gsm files by getting through to the Asterisk
answering service using a GrandStream BudgeTone 102 phone.  I then copy
the file which is stored in voicemail and use sox to increase the volume.  
Results are okay but nothing to write home about particularly (or maybe 
that is just my lack of a good telephone voice).

Michael

On Mon, 6 Oct 2003, Brian Capouch wrote:

 Shaun Ewing wrote:
 
 I know that some of this comes with the territory, but I wonder if
 there is anyone out there who does this ( wav - gsm) routinely, and
 who can advise me as to the MO I could use that results in the highest
 quality in the resulting playback files.
  
  What are you using to convert the wav files to gsm? I've been using 'sox'
  under Linux and have had no quality issues whatsoever.
  
  An example line to convert:
  
  sox file.wav -r 8000 -c 1 file.gsm
  
 
 That's the exact command I'm using, but it sure does sound crappy to my 
 ears.
 
 Perhaps that's the best I can do w/gsm, and of course I expect once the 
 sound is sent through the PSTN most of the highs and bottom are gone 
 anyways.
 
 I was just hoping there was something I could do to make the resulting 
 files a bit clearer.
 
 Thx.
 
 B.
 

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Anyone else use Audacity for prompts?

[wasim]

On Mon, 6 Oct 2003, Brian Capouch wrote:

 I was just hoping there was something I could do to make the resulting 
 files a bit clearer.

put them through baudline and see what's happening
also you might wanna try bandpass filter using ecasound

- wasim
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Noise with Grandstream/PSTN

[Dave Cotton]

Up until yesterday I've had a lot of high pitched noise when connecting
a BT101 to the PSTN via the X100P. I was using an Asus A7V133 with raid
motherboard and an 850 AMD Duron.  Over the weekend I thought I'd try
another machine. I had an HP Vectra 400 Mhz PII MMX with 128 Mb RAM
available, today no noise at all.

Now I must see if the Vectra is up to the job.
 
-- 
Dave Cotton [EMAIL PROTECTED]

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Anyone else use Audacity for prompts?

[Alastair Maw]

On 06/10/03 08:25, Shaun Ewing wrote:

The .wav files I'm producing are of stellar quality.  However, once I
turn them into .gsm, they sound buzzy and muffled.

An example line to convert:

sox file.wav -r 8000 -c 1 file.gsm
It'll sound much better if you go:

sox file.wav -r 8000 -c 1 file.gsm resample

Of course, there's only so much you can do to make 8kHz prompts sound 
any good. Doing the original recording at 8kHz is a good start.

--
Alastair Maw
System Analyst @ MX Telcom
www.mxtelecom.com
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Noise with Grandstream/PSTN

[WipeOut]

Dave Cotton wrote:

Up until yesterday I've had a lot of high pitched noise when connecting
a BT101 to the PSTN via the X100P. I was using an Asus A7V133 with raid
motherboard and an 850 AMD Duron.  Over the weekend I thought I'd try
another machine. I had an HP Vectra 400 Mhz PII MMX with 128 Mb RAM
available, today no noise at all.
Now I must see if the Vectra is up to the job.

 

I have been using a P2 400 for a while and its been fine for everything 
I have thrown at it.. I have to upgrade now becasue the PCI slots don't 
supply 3.3v so the TDM10B card that I just got does now work.. :(

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Good W2K softphone

[Joe Dennick]

I've had good luck with X-Lite (www.xten.com).

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of God Knows
Well
Sent: Monday, October 06, 2003 1:03 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Good W2K softphone



Hi


U can visit the http://iaxclient.sf.net for some opensource
underdevelopment 
softphones.

Take Care

Obaid Amin Syed

From: Chris Albertson [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Good W2K softphone
Date: Fri, 3 Oct 2003 23:00:13 -0700 (PDT)


I haven't found any open source/freeware software phones that run under

Windows 2000 that I like or that even work well. What are other people 
using?

=
Chris Albertson
   Home:   310-376-1029  [EMAIL PROTECTED]
   Cell:   310-990-7550
   Office: 310-336-5189  [EMAIL PROTECTED]
   KG6OMK

__
Do you Yahoo!?
The New Yahoo! Shopping - with improved product search 
http://shopping.yahoo.com 
___
Asterisk-Users mailing list
[EMAIL PROTECTED] 
http://lists.digium.com/mailman/listinfo/asterisk-users

_
The new MSN 8: advanced junk mail protection and 2 months FREE* 
http://join.msn.com/?page=features/junkmail

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] IVR Questions?

[duncan]

I failed to mention in my original post that I've looked at perl through 
AGI, but haven't yet found a function that allows me to capture digits to 
a variable that I can then manipulate.  I probably should also mention 
that I'm not a programmer-type, although I can usually muddle through 
simple scripts for smaller uses like this one.  Can you suggest which 
function I would use?
using the asterisk-perl module from http://asterisk.gnuinter.net/

you would use the get_data function:

my $captured_dtmfs = 
$AGI-get_data('/var/lib/asterisk/sounds/whatever','5','4');

from the show agi get data help

show agi get data
Usage: GET DATA file to be streamed [timeout] [max digits]
 Stream the given file, and recieve DTMF data. Returns the digits 
recieved from the channel at the other end.

hope this helps

duncan

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Ascom Ascotel 2050 Fritz PCI Card (Capi)

[Dave Sykes]

kapejod,

Thanks for the prompt reply, that didn't work, although I managed to work
it out, and the msn number I need to put in is the position number in the
S0 bus, for example for X616 it would 1, X617 2 and so on. Thanks for the
response it allowed me to work it out.

Thanks

Dave Sykes
Head of Research
+44 (0)1252 740721



|-+-
| |   Klaus-Peter Junghanns |
| |   [EMAIL PROTECTED]   |
| |   Sent by:  |
| |   [EMAIL PROTECTED]|
| |   .digium.com   |
| | |
| | |
| |   03/10/2003 18:03  |
| |   Please respond to |
| |   asterisk-users|
| | |
|-+-
  
|
  |
|
  |   To:   [EMAIL PROTECTED]  
  |
  |   cc:   (bcc: Dave Sykes/UK/Elateral)  
|
  |   Subject:  Re: [Asterisk-Users] Ascom Ascotel 2050  Fritz PCI Card (Capi)
|
  
|




Hi Dave,

try :

Dial(CAPI/${CALLERIDNUM}:${EXTEN},10)

(make sure you have a msn= line in capi.conf that allows
this for ${CALLERIDNUM})

regards

kapejod


Am Fre, 2003-10-03 um 18.58 schrieb Dave Sykes:
 Hello,

 We have been trying to add asterisk to our Ascom Ascotel 2050 PBX. We
have
 a AVM Fritz!PCI Card connected to an S0 bus extension from the PBX. The
 fritz card is configured to use chan_capi, and we can make calls SIP-SIP
 SIP-PBX extension PBX extension-SIP all successfully, we have assigned
 more than one PBX extension number to the S0 port in the Ascom PBX (it
has
 8 positions) calls from the PBX to the SIP phones work fine and are
 directed correctly. However when we dial from a SIP phone to an Ascom
 extension it always comes up on the Ascom handset with the number of the
 first position in the S0 setup.

 For Example

 S0 setup in Ascom
 position 1: 616
 position 2: 617
 position 3: 618

 in extensions.conf

 exten = 1,1,Dial(Sip/X616,10)
 exten = 2,1,Dial(Sip/X617,10)
 exten = 3,1,Dial(Sip/X618,10)

 calls from an Ascom extension to 616,617,618 get directed correctly to
the
 Sip phone X616,X617,X618 respectively

 for outgoing calls we have

 exten = _XXX,1,Dial(CAPI/@01234567${CALLERIDNUM}:${EXTEN},10)

 and dialling for instance 765 from a Sip phone correctly routes the call
to
 the Ascom and that then connects through to extension 765, however no
 matter which Sip phone dials it always displays 616 on the Ascom phone as
 the Caller ID. I can see from the * console that it has picked up the
 caller id correctly
 for example to dial 765 from Sip/X617
 -- Executing Dial(SIP/X617-5ee5, CAPI/@01234567617:765|10) in new
stack

 So the question is: is it possible to tell the Ascom exchange that the
call
 is coming from extension 617, as I guess it does not use the caller id as
 that seems to be set up correctly?






e-solutions for marketing
http://www.elateral.com/uk

Elateral Limited, Crosby Way, Farnham, GU9 7XX, UK
Tel. +44 (0)1252 740740
Fax. +44 (0)1252 740741
***
This e-mail is confidential and may be privileged.  It may be read,
copied and used only by the intended recipient.  If you have received
it in error please contact the sender immediately by return e-mail or
by telephone. Please then delete the e-mail and do not disclose its
contents to any person.  We believe, but do not warrant, that this
e-mail and any attachments are virus free.  You should take full
responsibility for virus checking.  Elateral reserves the right to
monitor all e-mail communications through its internal and external
networks.  The opinions or ideas expressed above belong to their
author and do not necessarily reflect the views of the Elateral Group.
***






___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Alternatives to FXS cards?

[Matt Lawson]

Hi everyone,

I know someone makes a product that's a POTS phone to SIP converter, 
where you just plug your POTS phone in one side and the network cable in 
the other.  Has anyone successfully used any of these with Asterisk, and 
if so how expensive were they?

I ask partly out of frustration with the FXS cards but mostly because it 
would make installation MUCH easier for what we're doing, plus it would 
be another piece of hardware that we could re-sell, plus it would free 
up some slots in the server, which is valuable real estate.

Comments?

TIA
- Matt
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Alternatives to FXS cards?

[WipeOut]

Matt Lawson wrote:

Hi everyone,

I know someone makes a product that's a POTS phone to SIP converter, 
where you just plug your POTS phone in one side and the network cable 
in the other.  Has anyone successfully used any of these with 
Asterisk, and if so how expensive were they?

I ask partly out of frustration with the FXS cards but mostly because 
it would make installation MUCH easier for what we're doing, plus it 
would be another piece of hardware that we could re-sell, plus it 
would free up some slots in the server, which is valuable real estate.

Comments?

TIA
- Matt
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Grandstream make the cheapest one I have see, they go for about $70 
each.. should work the same as the Bugetone phones so should be 
compatible with Asterisk.. but I haven't tested one..

Later

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] problems with the extensions of sip in ATA 186

[Javier Rios]

Greetings



I
am a new user of Asterisk and am beginning to make tests.



My problem is the following one:



I
have a Cisco ATA 186, already obtains that it was validated in asterisk, but I
have a doubt of like being able to declare the extenciones for the equipment.



This
it I must make in the file extensions.conf
??



You
can indicate me an example? 








 
  
  Ing Javier Rios
  
 
 
  
  Ing de Proyectos
  
 
 
  
  04167285748
  
 
 
  
  212 2637246 /2637187
  
 

[Asterisk-Users] runing asterisk and apache

[listas iPfone]

Hi All,

I´m thinking in install apache in my asterisk machine to host a litle site.

Anybody knows about problems doing that?

thanks

miklos

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] runing asterisk and apache

[Jon Stockill]

On Mon, 6 Oct 2003, listas iPfone wrote:

 I´m thinking in install apache in my asterisk machine to host a litle site.

 Anybody knows about problems doing that?

I've got apache installed on my asterisk server - it's handy for setting
up calls (a cgi just drops a call file into the outgoing call directory
with the relevant details in). I've not seen any problems. I suppose it
would depend on the amount of traffic you were expecting the site to
getnerate.

-- 
Jon Stockill
[EMAIL PROTECTED]
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] runing asterisk and apache

[Neil Stone]

listas iPfone wrote:
Hi All,

I´m thinking in install apache in my asterisk machine to host a litle site.

Anybody knows about problems doing that?

thanks

miklos

I also run a mail server on mine... among varoius other things

Neil

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Good W2K softphone

[Ali Davachi]

We are trying Asterisk for the first time, and have been unable to get
the X-Lite softphone working (send/receive).  Can anyone provide a
sip.conf that works?

Thanks

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joe Dennick
Sent: Monday, October 06, 2003 8:36 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Good W2K softphone


I've had good luck with X-Lite (www.xten.com).

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of God Knows
Well
Sent: Monday, October 06, 2003 1:03 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Good W2K softphone



Hi


U can visit the http://iaxclient.sf.net for some opensource
underdevelopment 
softphones.

Take Care

Obaid Amin Syed

From: Chris Albertson [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Good W2K softphone
Date: Fri, 3 Oct 2003 23:00:13 -0700 (PDT)


I haven't found any open source/freeware software phones that run under

Windows 2000 that I like or that even work well. What are other people 
using?

=
Chris Albertson
   Home:   310-376-1029  [EMAIL PROTECTED]
   Cell:   310-990-7550
   Office: 310-336-5189  [EMAIL PROTECTED]
   KG6OMK

__
Do you Yahoo!?
The New Yahoo! Shopping - with improved product search 
http://shopping.yahoo.com 
___
Asterisk-Users mailing list
[EMAIL PROTECTED] 
http://lists.digium.com/mailman/listinfo/asterisk-users

_
The new MSN 8: advanced junk mail protection and 2 months FREE* 
http://join.msn.com/?page=features/junkmail

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Good W2K softphone

[Joe Dennick]

Sip.conf
[7002]
type=friend
secret=blah
host=dynamic

You will also need an entry for 7002 in extensions.conf.  You can then
configure the X-lite phone to use 7002 as both its user and
authenticated user with a password of 'blah'.  Make sure you select the
'Send Internal IP' option to 'On'.  It won't register otherwise.  You'll
see the X-Lite phone connect, and also see it in the CLI if you are
monitoring that.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ali Davachi
Sent: Monday, October 06, 2003 9:15 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Good W2K softphone


We are trying Asterisk for the first time, and have been unable to get
the X-Lite softphone working (send/receive).  Can anyone provide a
sip.conf that works?

Thanks

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joe Dennick
Sent: Monday, October 06, 2003 8:36 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Good W2K softphone


I've had good luck with X-Lite (www.xten.com).

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of God Knows
Well
Sent: Monday, October 06, 2003 1:03 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Good W2K softphone



Hi


U can visit the http://iaxclient.sf.net for some opensource
underdevelopment 
softphones.

Take Care

Obaid Amin Syed

From: Chris Albertson [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Good W2K softphone
Date: Fri, 3 Oct 2003 23:00:13 -0700 (PDT)


I haven't found any open source/freeware software phones that run under

Windows 2000 that I like or that even work well. What are other people
using?

=
Chris Albertson
   Home:   310-376-1029  [EMAIL PROTECTED]
   Cell:   310-990-7550
   Office: 310-336-5189  [EMAIL PROTECTED]
   KG6OMK

__
Do you Yahoo!?
The New Yahoo! Shopping - with improved product search
http://shopping.yahoo.com 
___
Asterisk-Users mailing list
[EMAIL PROTECTED] 
http://lists.digium.com/mailman/listinfo/asterisk-users

_
The new MSN 8: advanced junk mail protection and 2 months FREE* 
http://join.msn.com/?page=features/junkmail

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] runing asterisk and apache

[Dave Cotton]

On Mon, 2003-10-06 at 15:57, listas iPfone wrote:
 Hi All,
 
 I´m thinking in install apache in my asterisk machine to host a litle site.

Asterisk + Openvpn + Apache + MySql + Postfix + Amavisd + Sophie + local
mirror of Mandrake Cooker.

This is Linux at work not M$ at play. 
 
-- 
Dave Cotton [EMAIL PROTECTED]

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] phonecore source

[James Coberly]

Hi,

Trying to compile gnophone and am having a bit of a time finding the 
source for phonecore.  Anyone know of somewhere I can pull the source from?

Thanks,

James-

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Data base structure

[Bartosz Jozwiak]

Hello,

Could somebody tell me where I can find information 
how should look
like the structure of database for dbget, dbput and 
so on

Thank you,

-- Bart

[Asterisk-Users] Asterisk, X-Lite and iLBC..still..

[WipeOut]

Hi,

Has anyone managed to get X-Lite to work with Asterisk using the iLBC 
codec.. I have just tried updating the the latest version 1079 (BTW this 
new version supports up to 10 proxy configurations, Not that I can see a 
reason to have 10 proxies setup, I would rather have the ability to 
transfer calls)..

I can make a call using iLBC but the sound that I hear is just a lot of 
pop's and crunches..

Later..

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Asterisk, X-Lite and iLBC..still..

[Leif Madsen]

WipeOut wrote:

Hi,

Has anyone managed to get X-Lite to work with Asterisk using the iLBC 
codec.. I have just tried updating the the latest version 1079 (BTW this 
new version supports up to 10 proxy configurations, Not that I can see a 
reason to have 10 proxies setup, I would rather have the ability to 
transfer calls)..

I can make a call using iLBC but the sound that I hear is just a lot of 
pop's and crunches..
Hey... they can't give you EVERYTHING.  There has to be some reason to 
buy the Pro version :)

Thanks,
Leif Madsen.
FWD: 18924
IAX: 1700-363-0761
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Re: Help with GPL license of Asterisk

[Gerald Henriksen]

On Thu, 02 Oct 2003 11:26:56 -0700, Jan Rychter [EMAIL PROTECTED]
wrote:

Having worked with GPL software quite a bit, also in the commercial
world, and having gotten some legal advice, I believe that the
anti-patent clauses in the GPL and LGPL are quite possibly the biggest
problem preventing the use of GPL'd software by commercial entities,
much bigger than the pass on the source and the rights requirement.

Not really.  Certainly it hasn't stopped lots of companies big and
small from releasing GPL software.

As I understand it (and as my legal counsel advises me) this effectively
means that if I distribute GPL/LGPL code, I have to make sure that its
distribution and re-distribution is not restricted by patents (or other
restrictions).

No, simply because that would be impossible (both because you would
never be able to program given the number of patents you would have to
search, and because it is entirely probable that no software is
entirely patent free).

What you can't do is knowingly license some source code/software under
the GPL/LGPL if you are already aware of any patent or other issues
that would in any way conflict with the redistribution of that code.

If the code in question contains parts which some patents lay claim to,
restricting distribution, then I must not distribute the code at
all. 

Correct.  The GPL/LGPL allow no further restrictions other than that
of the GPL/LGPL itself.  

It is needless to mention that it is impossible to me to verify that no
patents (worldwide!) lay claim to the code I'm distributing and impose
restrictions upon its distribution. Sooner or later I'm going to find
out that I do not comply with the GPL, because I distribute GPLd code
even though there are patent restrictions that apply to it.

Possibly.  But those same issues apply to any software whether open
source or closed source.  Regardless of the license used patents would
still apply, and would be enough to force you to stop distributing
your software without an appropriate license (and possibly fee).

An example of a particularly clear case of this problem is the XviD code
(http://www.xvid.org/), which is GPL-licensed. It seems to me that the
authors (copyright holders, to be precise) may distribute the software
under any license they choose, but nobody else is allowed to
re-distribute it, because they would be violating section 7 of the GPL,
as the MPEG-4 compression is (in some countries) covered by patents
requiring royalties to be paid.

Wrong.  The authors of xvid cannot license it under the GPL/LGPL
because MPEG-4 is known to have patent license issues.  In other words
the patent issues place a restriction on distribution that violates
the GPL, hence it cannot be GPL.

This is not unique to xvid, the same issue applies to any of the mp3
decoders (like xmms) which cannot be GPL/LPGL licensed which is why at
the very least Red Hat has removed those programs from their
distribution.

If those authors want to release open source codecs then they need to
either:

a) use another open source license that does allow restrictions on
further redistribution (I believe the BSD license falls into this
category but I could be wrong).

b) arrange for an exemption for any GPL software from those patents. 

c) implement a codec with no known patent issues (like ogg vorbis).

This is an issue which is very often overlooked in the hot GPL
debates. However, in the commercial world, it is possibly the most
important one.

Not overlooked, it is just not an issue.

Conclusion (IMHO of course): if you have the choice, use a license that
is OSI-compliant but does not have the anti-patent clause. Or has it
phrased differently.

It all depends on what your goal is.

Remember that the GPL also offers protection to companies.  One of the
reasons companies like IBM and SGI are releasing some of their stuff
under the GPL is precisely because it does protect them from having
their competitors simply take the technology and incorporating it into
their non-open source software.


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Problems with Caller ID on FXO

[Chris Hirsch]

And a followup for some debug messages for ManxPower:

*CLI -- Starting simple switch on 'Zap/1-1'
NOTICE[262161]: File callerid.c, Line 238 (callerid_feed): Caller*ID
failed checksum
ERROR[262161]: File callerid.c, Line 192 (callerid_feed): fsk_serie
made mylen  0 (-30)
WARNING[262161]: File chan_zap.c, Line 4340 (ss_thread): CallerID feed
failed: Success
WARNING[262161]: File chan_zap.c, Line 4357 (ss_thread): CallerID
returned with error on channel 'Zap/1-1'
NOTICE[262161]: File app_dial.c, Line 502 (dial_exec): Unable to create
channel of type 'IAX'
 == Everyone is busy at this time
WARNING[262161]: File pbx.c, Line 1810 (ast_pbx_run): Timeout, but no
rule 't' in context 'inbound-analog'
 -- Hungup 'Zap/1-1'

I understand the problem with the IAX stuff since I don't have that
quite configured yet

Chris Hirsch wrote:

  
  
I'm still seeing this:
  
*CLI -- Starting simple switch on 'Zap/1-1'
NOTICE[262161]: File callerid.c, Line 238 (callerid_feed): Caller*ID
failed checksum
NOTICE[262161]: File chan_zap.c, Line 4314 (ss_thread): Got event 2
(Ring/Answered)...
  
My zaptel.conf:
fxsks=1
loadzone = us
defaultzone=us
  
My zapata.conf: 
[channels]
language=en
context=inbound-analog
signalling=fxs_ks
usecallerid=yes
echocancel=yes
echocancelwhenbridged=yes
channel = 1
  
Now I just realized that I haven't set up my sound card on this
computer and I'm seeing
WARNING[98311]: File chan_oss.c, Line 232 (sound_thread): Read error
on sound device: Resource temporarily unavailable
  
Is that necessary to somehow read the caller id burst?
  
Eric Wieling wrote:
  
On Fri, 2003-10-03 at 13:28, Chris Hirsch wrote:
  

  failed checksum
NOTICE[262161]: File chan_zap.c, Line 4314 (ss_thread): Got event 2 
(Ring/Answered)...



I've only seen this message when using callprogress=yes and/or
busydetect=yes.  Set them to no.

  
  
  
  -- 
Do pediatricians play miniature golf on Wednesdays?


http://ccicolorado.org
Exceptional Dogs for Exceptional People - Help Out Today!

  


-- 
If at first you don't succeed,then skydiving isn't for you


http://ccicolorado.org
Exceptional Dogs for Exceptional People - Help Out Today!

Re: [Asterisk-Users] Asterisk, X-Lite and iLBC..still..

[WipeOut]

Leif Madsen wrote:

WipeOut wrote:

Hi,

Has anyone managed to get X-Lite to work with Asterisk using the iLBC 
codec.. I have just tried updating the the latest version 1079 (BTW 
this new version supports up to 10 proxy configurations, Not that I 
can see a reason to have 10 proxies setup, I would rather have the 
ability to transfer calls)..

I can make a call using iLBC but the sound that I hear is just a lot 
of pop's and crunches..


Hey... they can't give you EVERYTHING.  There has to be some reason to 
buy the Pro version :)

I think the pro version is over priced (if you are buying a single copy 
anyway, volume pricing is better) when you can get a hard phone for $65..

Especially when I don't need a phone with 6 lines or conferencing or 10 
proxies.. I would be happy to trade it all for transfer.. :)

In fact I would be happy to pay a resonable price for X-Basic, being 
X-Lite as is but with Transfer.. Xten should expand their product range 
to include X-Lite, X-Basic  (as above), X-Mid (include conference) and 
X-Pro (as it is) and then have tiered pricing to match each product..

By thats just my Opinion.. :)

Later..

Oh, and I would like iLBC to work.. ;)

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Problems with Caller ID on FXO

[Eric Wieling]

His Telco is ATT Broadband, Caller*ID works on an analog phone.

On Mon, 2003-10-06 at 10:11, Chris Hirsch wrote:
 And a followup for some debug messages for ManxPower:
 
 *CLI -- Starting simple switch on 'Zap/1-1'
 NOTICE[262161]: File callerid.c, Line 238 (callerid_feed): Caller*ID
 failed checksum
 ERROR[262161]: File callerid.c, Line 192 (callerid_feed): fsk_serie
 made mylen  0 (-30)
 WARNING[262161]: File chan_zap.c, Line 4340 (ss_thread): CallerID feed
 failed: Success
 WARNING[262161]: File chan_zap.c, Line 4357 (ss_thread): CallerID
 returned with error on channel 'Zap/1-1'
 NOTICE[262161]: File app_dial.c, Line 502 (dial_exec): Unable to
 create channel of type 'IAX'
   == Everyone is busy at this time
 WARNING[262161]: File pbx.c, Line 1810 (ast_pbx_run): Timeout, but no
 rule 't' in context 'inbound-analog'
 -- Hungup 'Zap/1-1'
 
 I understand the problem with the IAX stuff since I don't have that
 quite configured yet
 
 Chris Hirsch wrote:
  I'm still seeing this:
  
  *CLI -- Starting simple switch on 'Zap/1-1'
  NOTICE[262161]: File callerid.c, Line 238 (callerid_feed): Caller*ID
  failed checksum
  NOTICE[262161]: File chan_zap.c, Line 4314 (ss_thread): Got event 2
  (Ring/Answered)...
  
  My zaptel.conf:
  fxsks=1
  loadzone = us
  defaultzone=us
  
  My zapata.conf: 
  [channels]
  language=en
  context=inbound-analog
  signalling=fxs_ks
  usecallerid=yes
  echocancel=yes
  echocancelwhenbridged=yes
  channel = 1
  
  Now I just realized that I haven't set up my sound card on this
  computer and I'm seeing
   WARNING[98311]: File chan_oss.c, Line 232 (sound_thread): Read
  error on sound device: Resource temporarily unavailable
  
  Is that necessary to somehow read the caller id burst?
  
  Eric Wieling wrote:
   On Fri, 2003-10-03 at 13:28, Chris Hirsch wrote:
 
failed checksum
NOTICE[262161]: File chan_zap.c, Line 4314 (ss_thread): Got event 2 
(Ring/Answered)...

   I've only seen this message when using callprogress=yes and/or
   busydetect=yes.  Set them to no.
   
 
  
  -- 
  Do pediatricians play miniature golf on Wednesdays?
  
  
  http://ccicolorado.org
  Exceptional Dogs for Exceptional People - Help Out Today!
  

 
 -- 
 If at first you don't succeed,then skydiving isn't for you
 
 
 http://ccicolorado.org
 Exceptional Dogs for Exceptional People - Help Out Today!
 
 
 __
 This message has been 'sanitized'. This means that potentially
 dangerous content has been rewritten or removed. The following log
 describes which actions were taken.
 
 Sanitizer (start=1065454032):
   Part (pos=2920):
 SanitizeFile (filename=unnamed.txt, mimetype=text/plain):
   Match (names=unnamed.txt, rule=1):
 ScanFile (file=/tmp/att-3f8189d0-6SL-unnamed.txt):
   Scan succeeded, file is clean.
 
 Enforced policy: unknown
 
   Match (names=unnamed.txt, rule=3):
 Enforced policy: accept
 
   Part (pos=5348):
 SanitizeFile (filename=unnamed.html, mimetype=text/html):
   Match (names=unnamed.html, rule=1):
 ScanFile (file=/tmp/att-3f8189d1-N5B-unnamed.html):
   Scan succeeded, file is clean.
 
 Enforced policy: unknown
 
   Match (names=unnamed.html, rule=3):
 Enforced policy: accept
 
 Rewrote HTML tag: _pre class=moz-signature cols=72_
   as: _pre class=moz-signature DANGEROUS_cols=72_
 Rewrote HTML tag: _pre cols=72 class=moz-signature_
   as: _pre DANGEROUS_cols=72 class=moz-signature_
 Total modifications so far: 2
 
 
 
 Anomy 0.0.0 : Sanitizer.pm $Id: Sanitizer.pm,v 1.79 2003/06/19
 19:22:00 bre Exp $ 
 
-- 
Sample configs and more: http://www.fnords.org/~eric/asterisk/

BTEL Consulting
+1-850-484-4535 x2111 (Pensacola)
+1-504-595-3916 x2111 (New Orleans)
+1-877-677-9643 x2111 (Toll Free)


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Priority Voicemail

[Clif Jones]

I am relatively new at Asterisk but have a 2-machine system running with
the developer kit (FXS/FXO cards), some Cisco SIP 7960's and a Audiocodes
MP-104 SIP gateway.  I would like to fix a couple of the voicemail boxes
so someone can press some numbers such as 911 and get sent to a priority
voicemail box.  This different voicemail box would weed out the normal calls
from a special call.  How would one approach this?  My initial guess is that
you have to give an opportunity to do this before you enter voicemail or
maybe allow the user to exit the voicemail before leaving a message and then
send them to an auto-attendant.  Any good ideas or examples would be
greaty appreciated!
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] newbie question: 1 or 2 servers

[Mireia.Munoz-de-jesus]

Hi!

I have an H.323 network. I am trying to do a SIP-H.323 gateway to be able to
accept all the calls that come from other networks (SIP).

I have some questions:

- Must I use IAX? If so, how? I have not SIP servers, I only have a H.323
gatekeeper.

- If not, where is that I say that one call that come from SIP must be
translated to H.323 and vice-versa?

Thanks a lot for all your help.

Mireia


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Problems with Caller ID on FXO

[Chris Hirsch]

And a followup for some debug messages for ManxPower (2nd time since I
can't seem to get all the details :-) ):

FXO Card Installed:
[EMAIL PROTECTED] asterisk]# cat /proc/zaptel/1 
Span 1: WCFXO/0 "Wildcard X101P Board 1" 

 1 WCFXO/0/0 FXSKS (In use) 

Analog Line:
ATT Broadband coming over cable and then into standard phone
lines. Caller ID on other phones works fine.

Asterisk Version is Latest from CVS as of around 09:30 MST today:

This is the output when I receive a call:
*CLI -- Starting simple switch on 'Zap/1-1'
NOTICE[262161]: File callerid.c, Line 238 (callerid_feed): Caller*ID
failed checksum
ERROR[262161]: File callerid.c, Line 192 (callerid_feed): fsk_serie
made mylen  0 (-30)
WARNING[262161]: File chan_zap.c, Line 4340 (ss_thread): CallerID feed
failed: Success
WARNING[262161]: File chan_zap.c, Line 4357 (ss_thread): CallerID
returned with error on channel 'Zap/1-1'
NOTICE[262161]: File app_dial.c, Line 502 (dial_exec): Unable to create
channel of type 'IAX'
 == Everyone is busy at this time
WARNING[262161]: File pbx.c, Line 1810 (ast_pbx_run): Timeout, but no
rule 't' in context 'inbound-analog'
 -- Hungup 'Zap/1-1'

I understand the problem with the IAX stuff since I don't have that
quite configured yet

Chris Hirsch wrote:

  
  
I'm still seeing this:
  
*CLI -- Starting simple switch on 'Zap/1-1'
NOTICE[262161]: File callerid.c, Line 238 (callerid_feed): Caller*ID
failed checksum
NOTICE[262161]: File chan_zap.c, Line 4314 (ss_thread): Got event 2
(Ring/Answered)...
  
My zaptel.conf:
fxsks=1
loadzone = us
defaultzone=us
  
My zapata.conf: 
[channels]
language=en
context=inbound-analog
signalling=fxs_ks
usecallerid=yes
echocancel=yes
echocancelwhenbridged=yes
channel = 1
  
Now I just realized that I haven't set up my sound card on this
computer and I'm seeing
WARNING[98311]: File chan_oss.c, Line 232 (sound_thread): Read error
on sound device: Resource temporarily unavailable
  
Is that necessary to somehow read the caller id burst?
  
Eric Wieling wrote:
  
On Fri, 2003-10-03 at 13:28, Chris Hirsch wrote:
  

  failed checksum
NOTICE[262161]: File chan_zap.c, Line 4314 (ss_thread): Got event 2 
(Ring/Answered)...



I've only seen this message when using callprogress=yes and/or
busydetect=yes.  Set them to no.

  
  
  
  -- 
Do pediatricians play miniature golf on Wednesdays?


http://ccicolorado.org
Exceptional Dogs for Exceptional People - Help Out Today!

  


-- 
If at first you don't succeed,then skydiving isn't for you


http://ccicolorado.org
Exceptional Dogs for Exceptional People - Help Out Today!

Re: DB virtualization for multiple database support - Was Re: [Asterisk-Users] How to use vmdb.sql in voicemail.conf/extension.conf

[Tilghman Lesher]

On Monday 06 October 2003 01:43 am, Chris Albertson wrote:
 I think ODBC is the way to go.  There is really nothing to write.

That's great.  Why haven't you written and contributed it yet?

-Tilghman

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Anyone else use Audacity for prompts?

[Brad Waite]

Why, oh why, do we have to be limited to 8kHz prompts in the first place?

Alastair Maw wrote:

On 06/10/03 08:25, Shaun Ewing wrote:

The .wav files I'm producing are of stellar quality.  However, once I
turn them into .gsm, they sound buzzy and muffled.


An example line to convert:

sox file.wav -r 8000 -c 1 file.gsm


It'll sound much better if you go:

sox file.wav -r 8000 -c 1 file.gsm resample

Of course, there's only so much you can do to make 8kHz prompts sound 
any good. Doing the original recording at 8kHz is a good start.

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Anyone else use Audacity for prompts?

[Steven Critchfield]

On Mon, 2003-10-06 at 11:08, Brad Waite wrote:
 Why, oh why, do we have to be limited to 8kHz prompts in the first
 place?

Because that is what telephony is based on. 8khz by 8 bit if on a
digital link and 7 bit if in RBS signaling.

Why are you so worried about that amount of degradation when you can't
control how much makeup and other crap is sitting in the speaker holes
on the other side deadening the sound. Then there is the weather related
noise introduced in old analog links, and then there is the whole analog
loop. If the sound quality is of such a big issue with you, go to
something other than a telephony app to use.

 Alastair Maw wrote:
 
  On 06/10/03 08:25, Shaun Ewing wrote:
  
  The .wav files I'm producing are of stellar quality.  However, once I
  turn them into .gsm, they sound buzzy and muffled.
  
  
  An example line to convert:
 
  sox file.wav -r 8000 -c 1 file.gsm
  
  
  It'll sound much better if you go:
  
  sox file.wav -r 8000 -c 1 file.gsm resample
  
  Of course, there's only so much you can do to make 8kHz prompts sound 
  any good. Doing the original recording at 8kHz is a good start.
  
 
 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED]
 http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
Steven Critchfield  [EMAIL PROTECTED]

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Anyone else use Audacity for prompts?

[Stuart Mackintosh]

I have had to work on some files recently with a similar problem.

It seems that when a file is recorded in 16 bit and converted to 8 bit,
the clarity is lost. 
I have found the following ways the most productive:

1)Record through the voicemail system then import and edit them
afterwards. As long as you use a good quality channel, this operates
well.

2) Use sox with -t oss /dev/dsp as the input and a gsm file as the
output. -r 8000 for the sample rate and -b for 8 bit.



On Mon, 2003-10-06 at 17:08, Brad Waite wrote:
 Why, oh why, do we have to be limited to 8kHz prompts in the first place?
 
 Alastair Maw wrote:
 
  On 06/10/03 08:25, Shaun Ewing wrote:
  
  The .wav files I'm producing are of stellar quality.  However, once I
  turn them into .gsm, they sound buzzy and muffled.
  
  
  An example line to convert:
 
  sox file.wav -r 8000 -c 1 file.gsm
  
  

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Anyone else use Audacity for prompts?

[Joe Dennick]

And then use standard Unix commands to move that recording to where you
want it like /var/lib/asterisk/sounds/new-recording.gsm so you can then
call it from your menus or prompts.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian West
Sent: Monday, October 06, 2003 11:28 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Anyone else use Audacity for prompts?


Why do stuff the hard way?

; used to record prompts
exten = 205,1,Wait(2)
exten = 205,2,Record(/tmp/asterisk-recording:gsm)
exten = 205,3,Wait(2)
exten = 205,4,Playback(/tmp/asterisk-recording)
exten = 205,5,Wait(2)
exten = 205,6,Hangup



On Mon, 6 Oct 2003, Stuart Mackintosh wrote:

 I have had to work on some files recently with a similar problem.

 It seems that when a file is recorded in 16 bit and converted to 8 
 bit, the clarity is lost. I have found the following ways the most 
 productive:

 1)Record through the voicemail system then import and edit them 
 afterwards. As long as you use a good quality channel, this operates 
 well.

 2) Use sox with -t oss /dev/dsp as the input and a gsm file as the 
 output. -r 8000 for the sample rate and -b for 8 bit.



 On Mon, 2003-10-06 at 17:08, Brad Waite wrote:
  Why, oh why, do we have to be limited to 8kHz prompts in the first 
  place?
 
  Alastair Maw wrote:
 
   On 06/10/03 08:25, Shaun Ewing wrote:
  
   The .wav files I'm producing are of stellar quality.  However, 
   once I turn them into .gsm, they sound buzzy and muffled.
  
  
   An example line to convert:
  
   sox file.wav -r 8000 -c 1 file.gsm
  
  

 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED] 
 http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Asterisk, X-Lite and iLBC..still..

[Joe Dennick]

Where does one find a hard-phone for $65?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Leif Madsen
Sent: Monday, October 06, 2003 11:33 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk, X-Lite and iLBC..still..


WipeOut wrote:

 Leif Madsen wrote:
 
 WipeOut wrote:

 Hi,

 Has anyone managed to get X-Lite to work with Asterisk using the 
 iLBC
 codec.. I have just tried updating the the latest version 1079 (BTW 
 this new version supports up to 10 proxy configurations, Not that I 
 can see a reason to have 10 proxies setup, I would rather have the 
 ability to transfer calls)..

 I can make a call using iLBC but the sound that I hear is just a lot
 of pop's and crunches..



 Hey... they can't give you EVERYTHING.  There has to be some reason 
 to
 buy the Pro version :)

 I think the pro version is over priced (if you are buying a single 
 copy
 anyway, volume pricing is better) when you can get a hard phone for
$65..
 
 Especially when I don't need a phone with 6 lines or conferencing or 
 10
 proxies.. I would be happy to trade it all for transfer.. :)
 
 In fact I would be happy to pay a resonable price for X-Basic, being
 X-Lite as is but with Transfer.. Xten should expand their product
range 
 to include X-Lite, X-Basic  (as above), X-Mid (include conference) and

 X-Pro (as it is) and then have tiered pricing to match each product..
 
 By thats just my Opinion.. :)

I do agree with that.  I do think the X-Pro is a bit pricey.  They'd 
probably be better served to have some sort of version at about the 
$19.95 range and $29.95 range.  I'm sure they'd see more sales that way 
(remember.. you make money in volume, not price per unit!)

Thanks,
Leif Madsen.
FWD: 18924
IAX: 1700-363-0761

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Asterisk, X-Lite and iLBC..still..

[Eric Wieling]

Two phones for $130 at
http://www.sipphone.com/tiki-index.php?page=Order%20Now

On Mon, 2003-10-06 at 11:42, Joe Dennick wrote:
 Where does one find a hard-phone for $65?
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Leif Madsen
 Sent: Monday, October 06, 2003 11:33 AM
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] Asterisk, X-Lite and iLBC..still..
 
 
 WipeOut wrote:
 
  Leif Madsen wrote:
  
  WipeOut wrote:
 
  Hi,
 
  Has anyone managed to get X-Lite to work with Asterisk using the 
  iLBC
  codec.. I have just tried updating the the latest version 1079 (BTW 
  this new version supports up to 10 proxy configurations, Not that I 
  can see a reason to have 10 proxies setup, I would rather have the 
  ability to transfer calls)..
 
  I can make a call using iLBC but the sound that I hear is just a lot
  of pop's and crunches..
 
 
 
  Hey... they can't give you EVERYTHING.  There has to be some reason 
  to
  buy the Pro version :)
 
  I think the pro version is over priced (if you are buying a single 
  copy
  anyway, volume pricing is better) when you can get a hard phone for
 $65..
  
  Especially when I don't need a phone with 6 lines or conferencing or 
  10
  proxies.. I would be happy to trade it all for transfer.. :)
  
  In fact I would be happy to pay a resonable price for X-Basic, being
  X-Lite as is but with Transfer.. Xten should expand their product
 range 
  to include X-Lite, X-Basic  (as above), X-Mid (include conference) and
 
  X-Pro (as it is) and then have tiered pricing to match each product..
  
  By thats just my Opinion.. :)
 
 I do agree with that.  I do think the X-Pro is a bit pricey.  They'd 
 probably be better served to have some sort of version at about the 
 $19.95 range and $29.95 range.  I'm sure they'd see more sales that way 
 (remember.. you make money in volume, not price per unit!)
 
 Thanks,
 Leif Madsen.
 FWD: 18924
 IAX: 1700-363-0761
 
 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED]
 http://lists.digium.com/mailman/listinfo/asterisk-users
 
 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED]
 http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
Sample configs and more: http://www.fnords.org/~eric/asterisk/

BTEL Consulting
+1-850-484-4535 x2111 (Pensacola)
+1-504-595-3916 x2111 (New Orleans)
+1-877-677-9643 x2111 (Toll Free)

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Data base structure

[Tilghman Lesher]

On Monday 06 October 2003 09:40 am, Bartosz Jozwiak wrote:
 Hello,

 Could somebody tell me where I can find information how should look
 like the structure of database for dbget, dbput and so on

Yep, see the manpage for dbopen(3).

-Tilghman

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Asterisk, X-Lite and iLBC..still..

[WipeOut]

Joe Dennick wrote:

Where does one find a hard-phone for $65?

 

Sorry, should have been $75..

But if you look at sipphone.com you can get two and it will cost you $65 
each..

Later..

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Remote control IVR

[Ívar Ragnarsson]

Hi

I work at a small company that has some IVR solutions that use Dialogic
hardware for everything.
Everything is written in C++ using MS VC++ using the Dialogic API and runs
only on Windows.
Being the rebel that I am, I would like free myself from Dialogic.
To do this without porting all our existing code to run on Linux I was
thinking of controlling the Asterisk from a Windows machine running our
existing code.  
i.e. making an API similar to the Dialogic one that would control Asterisk
over TCP/IP.

Has anyone done something like this? Or does someone have has a good idea
for implementing such a thing?

I am still familiarizing myself with Asterisk but as I understand it you can
add functionality to Asterisk through the AGI interface and by creating a
loadable application.
Which one would one should I use to solve my problem?  Does an loadable
application give you more control than an AGI script?

I would be very thankful for any advice/tips/sample code.


Best regards,
Ívar Ragnarsson
Grunnur ehf.
Iceland

ps. Does a loadable application have to be GPL licensed?
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Remote control IVR

[Tilghman Lesher]

On Monday 06 October 2003 12:13 pm, Ívar Ragnarsson wrote:
 I work at a small company that has some IVR solutions that use
 Dialogic hardware for everything.
 Everything is written in C++ using MS VC++ using the Dialogic API
 and runs only on Windows.
 Being the rebel that I am, I would like free myself from Dialogic.
 To do this without porting all our existing code to run on Linux I
 was thinking of controlling the Asterisk from a Windows machine
 running our existing code.
 i.e. making an API similar to the Dialogic one that would control
 Asterisk over TCP/IP.

 Has anyone done something like this? Or does someone have has a
 good idea for implementing such a thing?

 I am still familiarizing myself with Asterisk but as I understand
 it you can add functionality to Asterisk through the AGI interface
 and by creating a loadable application.
 Which one would one should I use to solve my problem?  Does an
 loadable application give you more control than an AGI script?

Applications and AGI are not self-starters.  They both need to be
invoked from the dial plan.  If you want to originate calls from
outside Asterisk, you need to take a look at the manager interface
which can be made to run on TCP port 5038 (it does not run by
default).

-Tilghman

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Remote control IVR

[James Sharp]

 Which one would one should I use to solve my problem?  Does an loadable
 application give you more control than an AGI script?


If you want something that runs continuously (such as a listener process
or control process), it'll have to be a loadable module.  AGI scripts only
get run when the extention they're configured for is called.



 Best regards,
   Ívar Ragnarsson
   Grunnur ehf.
   Iceland

 ps. Does a loadable application have to be GPL licensed?

Only if you plan to distribute it with asterisk, I believe.

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Remote control IVR

[Scott Stingel]

Hi Ivar-

I was in a similar position, doing mostly IVR apps, and having written lots
of Dialogic-based code.

The approach I took was to re-write some of my more popular applications in
Perl and use Asterisk's AGI interface.  However, if I were in your position
with lots of applications to convert, I might consider writing an API to
control the Digium hardware directly.  (I'm not sure how hard this would
be).  There are likely licensing issues to overcome - open source vs.
proprietary code etc, not to mention access to Digium's hardware
documentation.

Most of the asterisk code seems to be dedicated for the purpose of making
connections between ports, ie:  PBX functionality including VOIP, and
everything is driven by their dialplan which is not always ideal for IVR
applications or outbound apps.   My Perl scripts run fine for IVR app's but
most of the asterisk code is not used.  A Dialogic API emulation would be
pretty cool.

Cheers
Scott

Scott M. Stingel 
Emerging Voice Technology Inc.

Email:  [EMAIL PROTECTED]
URL:www.evtmedia.com  



 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Ívar Ragnarsson
 Sent: Monday, October 06, 2003 6:13 PM
 To: '[EMAIL PROTECTED]'
 Subject: [Asterisk-Users] Remote control IVR
 
 
 Hi
 
 I work at a small company that has some IVR solutions that 
 use Dialogic
 hardware for everything.
 Everything is written in C++ using MS VC++ using the Dialogic 
 API and runs
 only on Windows.
 Being the rebel that I am, I would like free myself from Dialogic.
 To do this without porting all our existing code to run on Linux I was
 thinking of controlling the Asterisk from a Windows machine 
 running our
 existing code.  
 i.e. making an API similar to the Dialogic one that would 
 control Asterisk
 over TCP/IP.
 
 Has anyone done something like this? Or does someone have has 
 a good idea
 for implementing such a thing?
 
 I am still familiarizing myself with Asterisk but as I 
 understand it you can
 add functionality to Asterisk through the AGI interface and 
 by creating a
 loadable application.
 Which one would one should I use to solve my problem?  Does 
 an loadable
 application give you more control than an AGI script?
 
 I would be very thankful for any advice/tips/sample code.
 
 
 Best regards,
   Ívar Ragnarsson
   Grunnur ehf.
   Iceland
 
 ps. Does a loadable application have to be GPL licensed?
 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED]
 http://lists.digium.com/mailman/listinfo/asterisk-users
 
 

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] phonecore source

[James Golovich]

On Mon, 6 Oct 2003, James Coberly wrote:

 Hi,
 
 Trying to compile gnophone and am having a bit of a time finding the 
 source for phonecore.  Anyone know of somewhere I can pull the source from?
 

I have a copy of the most recent gnophone source located at
http://asterisk.gnuinter.net/files/cvsnightly/phonecore-cvsnightly.tar.gz

James

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Asterisk, X-Lite and iLBC..still..

[Leif Madsen]

WipeOut wrote:

Joe Dennick wrote:

Where does one find a hard-phone for $65?

 

Sorry, should have been $75..

But if you look at sipphone.com you can get two and it will cost you $65 
each..

 You can get them for $65 at 
https://secure.pulver.com/pulverinnovations/order_grandstream.html

Thanks,
Leif Madsen.
FWD: 18924
IAX: 1700-363-0761
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: DB virtualization for multiple database support - Was Re: [Asterisk-Users] How to use vmdb.sql in voicemail.conf/extension.conf

[Chris Albertson]

--- Tilghman Lesher [EMAIL PROTECTED] wrote:
 On Monday 06 October 2003 01:43 am, Chris Albertson wrote:
  I think ODBC is the way to go.  There is really nothing to write.
 
 That's great.  Why haven't you written and contributed it yet?

Any code I write for free is GPL'd.  If they don't accept GPL
I don't contribute.  I'd do it if it could be done as a stand alone
application but by deffinition this project _can't_ be stand alone.
I'll let someone who is making money off Asterisks do this.  I'm
doing a GPL'd driver or simplex ham radio VOIP.  

=
Chris Albertson
  Home:   310-376-1029  [EMAIL PROTECTED]
  Cell:   310-990-7550
  Office: 310-336-5189  [EMAIL PROTECTED]
  KG6OMK

__
Do you Yahoo!?
The New Yahoo! Shopping - with improved product search
http://shopping.yahoo.com
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Start...

[Mireia.Munoz-de-jesus]

Hi all!

One easy question... I hope someone will answer me.

I've installed asterisk with the samples. Somewhere in my network I have an
H.323 Gatekeeper. What must I do to make that the gatekeeper talk with
Asterisk?

And I another little question... with the samples installed asterisk works
ok? What must I install to see how it works?

I am lost!! Please help me!

See you.

Mireia


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] chan_zap.c - echo cancelation getting in the way of dialing????

[mvickers]

It seems consistant after dialing dozens of times that the call that
doesn't go through is the one the gets the log message  No
echocancellation requested (chan_zap.c) and the Scheduleing timer
(channel.c)  in the middle of receiving the DTMF tones.

I'm now using the T400P card last week very simular problems the the T100P
(although I think I was actually loosing digits with the T100P card)


***
Example of failure Below  Echo cancelation in middle of DTMF tones not all
digits received.
***

Oct  6 10:55:12 DEBUG[27664]: File chan_zap.c, Line 3428 (zt_read): DTMF digit: 6 on 
Zap/18-1
Oct  6 10:55:13 DEBUG[27664]: File chan_zap.c, Line 3428 (zt_read): DTMF digit: 9 on 
Zap/18-1
Oct  6 10:55:13 DEBUG[27664]: File chan_zap.c, Line 1056 (zt_enable_ec): No 
echocancellation requested
Oct  6 10:55:16 DEBUG[27664]: File channel.c, Line 953 (ast_settimeout): Scheduling 
timer at 160 sample intervals
Oct  6 10:55:16 DEBUG[27664]: File chan_zap.c, Line 3428 (zt_read): DTMF digit: 7 on 
Zap/18-1
Oct  6 10:55:21 DEBUG[27664]: File channel.c, Line 953 (ast_settimeout): Scheduling 
timer at 0 sample intervals
Oct  6 10:55:21 DEBUG[27664]: File channel.c, Line 953 (ast_settimeout): Scheduling 
timer at 0 sample intervals
Oct  6 10:55:21 DEBUG[27664]: File chan_zap.c, Line 1629 (zt_hangup): Hangup: channel: 
18 index = 0, normal = 36, callwait = -1, thirdcall = -1
Oct  6 10:55:21 DEBUG[27664]: File chan_zap.c, Line 1996 (zt_setoption): Set option 
TDD MODE, value: OFF(0) on Zap/18-1
Oct  6 10:55:21 DEBUG[27664]: File chan_zap.c, Line 1028 (update_conf): Updated 
conferencing on 18, with 0 conference users
Oct  6 10:56:15 DEBUG[5126]: File chan_sip.c, Line 657 (create_addr): Setting NAT on 
RTP to 0
Oct  6 10:56:15 DEBUG[5126]: File chan_sip.c, Line 548 (__sip_ack): Stopping 
retransmission on '[EMAIL PROTECTED]'
of Request 102: Found



Working call


Oct  6 10:59:55 DEBUG[28688]: File chan_zap.c, Line 3428 (zt_read): DTMF digit: 6 on 
Zap/19-1
Oct  6 10:59:56 DEBUG[28688]: File chan_zap.c, Line 3428 (zt_read): DTMF digit: 9 on 
Zap/19-1
Oct  6 10:59:56 DEBUG[28688]: File chan_zap.c, Line 3428 (zt_read): DTMF digit: 7 on 
Zap/19-1
Oct  6 10:59:56 DEBUG[28688]: File chan_zap.c, Line 3428 (zt_read): DTMF digit: 1 on 
Zap/19-1
Oct  6 10:59:56 DEBUG[28688]: File chan_zap.c, Line 1056 (zt_enable_ec): No 
echocancellation requested
Oct  6 10:59:56 DEBUG[28688]: File chan_sip.c, Line 657 (create_addr): Setting NAT on 
RTP to 0
Oct  6 10:59:56 DEBUG[28688]: File chan_sip.c, Line 857 (sip_call): Outgoing Call for 
mvickers
Oct  6 10:59:56 DEBUG[5126]: File chan_sip.c, Line 568 (__sip_semi_ack): (Provisional) 
Stopping retransmission (but retaining packet) on
'[EMAIL PROTECTED]' Request 102: Found
Oct  6 10:59:57 DEBUG[5126]: File chan_sip.c, Line 568 (__sip_semi_ack): (Provisional) 
Stopping retransmission (but retaining packet) on
'[EMAIL PROTECTED]' Request 102: Found
Oct  6 10:59:57 DEBUG[28688]: File chan_zap.c, Line 3612 (zt_indicate):
Received AST_CONTROL_PROGRESS on Zap/19-1


***
A call that got all the digits but still fialed:



Oct  6 11:08:41 DEBUG[40976]: File chan_zap.c, Line 3428 (zt_read): DTMF digit: 6 on 
Zap/6-1
Oct  6 11:08:42 DEBUG[40976]: File chan_zap.c, Line 3428 (zt_read): DTMF digit: 9 on 
Zap/6-1
Oct  6 11:08:42 DEBUG[40976]: File chan_zap.c, Line 3428 (zt_read): DTMF digit: 7 on 
Zap/6-1
Oct  6 11:08:42 DEBUG[40976]: File chan_zap.c, Line 1056 (zt_enable_ec): No 
echocancellation requested
Oct  6 11:08:45 DEBUG[40976]: File channel.c, Line 953 (ast_settimeout): Scheduling 
timer at 160 sample intervals
Oct  6 11:08:45 DEBUG[40976]: File chan_zap.c, Line 3428 (zt_read): DTMF digit: 1 on 
Zap/6-1
Oct  6 11:08:50 DEBUG[40976]: File channel.c, Line 953 (ast_settimeout): Scheduling 
timer at 0 sample intervals
Oct  6 11:08:50 DEBUG[40976]: File channel.c, Line 953 (ast_settimeout): Scheduling 
timer at 0 sample intervals
Oct  6 11:08:50 DEBUG[40976]: File chan_zap.c, Line 1629 (zt_hangup): Hangup: channel: 
6 index = 0, normal = 24, callwait = -1, thirdcall = -1
Oct  6 11:08:50 DEBUG[40976]: File chan_zap.c, Line 1996 (zt_setoption): Set option 
TDD MODE, value: OFF(0) on Zap/6-1
Oct  6 11:08:50 DEBUG[40976]: File chan_zap.c, Line 1028 (update_conf): Updated 
conferencing on 6, with 0 conference users




Mark Vickers, RealNetworks Inc.  Desk: (206) 674-2391  Fax: (206)674-3588
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Remote control IVR

[Steven Critchfield]

On Mon, 2003-10-06 at 12:13, Ívar Ragnarsson wrote:
 Hi
 
 I work at a small company that has some IVR solutions that use
 Dialogic
 hardware for everything.
 Everything is written in C++ using MS VC++ using the Dialogic API and
 runs
 only on Windows.
 Being the rebel that I am, I would like free myself from Dialogic.
 To do this without porting all our existing code to run on Linux I was
 thinking of controlling the Asterisk from a Windows machine running
 our
 existing code.  
 i.e. making an API similar to the Dialogic one that would control
 Asterisk
 over TCP/IP.
 
 Has anyone done something like this? Or does someone have has a good
 idea
 for implementing such a thing?

It would be ugly but doable to link up with the IAX library and then be
just a VoIP endpoint. Then asterisk is just your gateway. Your problem
will be in licensing. You could try and sidestep that by using SIP or
h.323 especially if you can get access into the MS protocols. This way
you haven't left the comfort of your MS world, and you haven't had to
make interfaces do things they shouldn't.

 I am still familiarizing myself with Asterisk but as I understand it
 you can
 add functionality to Asterisk through the AGI interface and by
 creating a
 loadable application.
 Which one would one should I use to solve my problem?  Does an
 loadable
 application give you more control than an AGI script?

Yes, and no. A loadable app is going to have less overhead involved and
runs in the same memory space as asterisk itself. AGI on the other hand
is run separate and if it crashes, it doesn't take asterisk down and the
port is still available to answer the next call coming in. AGI can run
the applications that are loadable too.  

 ps. Does a loadable application have to be GPL licensed?

If they are distributed then yes. Only when it is distributed does the
license really matter.

-- 
Steven Critchfield  [EMAIL PROTECTED]

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Re: Remote control IVR

[George Pajari]

However, if I were in your
position
with lots of applications to convert, I might consider writing an API to
control the Digium hardware directly.  (I'm not sure how hard this would
be).  There are likely licensing issues to overcome - open source vs.
proprietary code etc
Just to clarify the GPL, there are no licensing issues if one combines 
open source with proprietary code for one's own use. The GPL only 
becomes a concern when you distribute the combination.

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Modem and Fax over VoIP

[Steven Critchfield]

On Mon, 2003-10-06 at 10:12, Eduardo Goncalves wrote:
 Hello,
 
   I have the fowling scenario:
 
 fxs[asterisk1]-iax-[asterisk2]e1em---PSTN
   
   I want to know the steps to transmit fax from a machine connected to the fxs 
 to a fax machine on the PSTN. The same for dial-up's.
   Is it possible only with a/ulaw ? 
   What configs I need in asterisk1?

If asterisk2 is your only access to the PSTN, then it doesn't make a lot
of sense to do fax over VoIP. Put a couple of modems on asterisk2 with
matching FXS ports and learn to use Hylafax. You will get better
performance out of the faxing and potentially lower bills. Also it is
more flexible. Not to mention that the bandwidth over VoIP to make data
quality calls is around 80k of VoIP to get flakey 9.6k. Do you need to
waste 10x bandwidth for sub par functionality?  
-- 
Steven Critchfield  [EMAIL PROTECTED]

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: DB virtualization for multiple database support - Was Re: [Asterisk-Users] How to use vmdb.sql in voicemail.conf/extension.conf

[Tilghman Lesher]

On Monday 06 October 2003 12:57 pm, Chris Albertson wrote:
 --- Tilghman Lesher [EMAIL PROTECTED] wrote:
  On Monday 06 October 2003 01:43 am, Chris Albertson wrote:
   I think ODBC is the way to go.  There is really nothing to
   write.
 
  That's great.  Why haven't you written and contributed it yet?

 Any code I write for free is GPL'd.  If they don't accept GPL
 I don't contribute.  I'd do it if it could be done as a stand alone
 application but by deffinition this project _can't_ be stand alone.
 I'll let someone who is making money off Asterisks do this.  I'm
 doing a GPL'd driver or simplex ham radio VOIP.

cdr_mysql_addon in asterisk-addons is also GPL.  Why not
contribute your GPLed cdr_odbc through the addons?

-Tilghman

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] suggested hardware especially sound cards

[Armand A. Verstappen]

Hi,

On Fri, 2003-10-03 at 15:52, mattf wrote:
 I've seen various suggestions thrown around for hardware when people ask,
 but can we all agree on some basic hardware recommendations for a few basic
 setups(and post them on a website) to make it easier for new people to avoid
 some of the hardware/software pitfalls when they are setting up their first
 systems.
snip
 I think we should have these setups listed:
 - home user with 1-2 telco lines and 2-5 phones
 - small office with 4-8 telco lines and 8-16 phones
 - small office with a fractional E1/T1 and 12-24 phones
 - medium office with full E1/T1 and 24-48 phones
 - medium office with 2-4 E1/T1s and 48-100 phones
 - large office with 4-16 E1/T1s and 100-500 phones
 - multi-location corporate offices with 16-64 E1/T1s distributed and
 500-2500 phones
 - ACD heavy office suggestions
 - IVR or Conference heavy suggestions

You can add a section this to the wiki
(http://www.voip-info.org/wiki-Asterisk), and fill out the suggestions
you have information for, then invite others to complete the others. All
you need to do is register, which is free.

wkr,

-- 
Envida http://www.envida.net/
Armand A. Verstappen   Graadt van Roggenweg 328
[EMAIL PROTECTED]   3531 AH Utrecht
tel: +31 (0)30 298 2255Postbus 19127
fax: +31 (0)30 298 21113501 DC Utrecht


signature.asc
Description: This is a digitally signed message part

Re: [Asterisk-Users] Modem and Fax over VoIP

[Eduardo Goncalves]

On Mon, 06 Oct 2003 13:43:21 -0500
Steven Critchfield [EMAIL PROTECTED] wrote:

 On Mon, 2003-10-06 at 10:12, Eduardo Goncalves wrote:
  Hello,
  
  I have the fowling scenario:
  
  fxs[asterisk1]-iax-[asterisk2]e1em---PSTN

 
 If asterisk2 is your only access to the PSTN, then it doesn't make a lot
 of sense to do fax over VoIP. Put a couple of modems on asterisk2 with
 matching FXS ports and learn to use Hylafax. 

I can't. Because asterisk1 is at a remote site. And the only access to PSTN at 
this remote site is trough asterisk2.

 performance out of the faxing and potentially lower bills. Also it is
 more flexible. Not to mention that the bandwidth over VoIP to make data
 quality calls is around 80k of VoIP to get flakey 9.6k. Do you need to
 waste 10x bandwidth for sub par functionality?  

Is it possible to config asterisk1 to change to g.711 only when a fax 
transmission is detected?

Thanks
Eduardo
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Modem and Fax over VoIP

[Steven Critchfield]

On Mon, 2003-10-06 at 13:55, Eduardo Goncalves wrote:
 On Mon, 06 Oct 2003 13:43:21 -0500
 Steven Critchfield [EMAIL PROTECTED] wrote:
 
  On Mon, 2003-10-06 at 10:12, Eduardo Goncalves wrote:
   Hello,
   
 I have the fowling scenario:
   
   fxs[asterisk1]-iax-[asterisk2]e1em---PSTN
 
  
  If asterisk2 is your only access to the PSTN, then it doesn't make a lot
  of sense to do fax over VoIP. Put a couple of modems on asterisk2 with
  matching FXS ports and learn to use Hylafax. 
 
   I can't. Because asterisk1 is at a remote site. And the only access
 to PSTN at this remote site is trough asterisk2.

unless you are using standalone fax units for the scanning, then you
should be able to. The scenario I laid out above was for specifically
this case. Drop a cheap flatbed scanner in to capture the image, then
send it to hylafax at asterisk2 location to hop a local modem to a FXS
port and off to the net. You can then receive incoming faxes via email
or similar and then choose what to print. 

  performance out of the faxing and potentially lower bills. Also it is
  more flexible. Not to mention that the bandwidth over VoIP to make data
  quality calls is around 80k of VoIP to get flakey 9.6k. Do you need to
  waste 10x bandwidth for sub par functionality?  
 
   Is it possible to config asterisk1 to change to g.711 only when a fax
 transmission is detected?

If it is on a specific non shared port, then you can set it up in a
context that uses a different username and password to connect, and in
that definition disallow all and allow ulaw. Just know that this method
is spotty, and may not work because of problems in the fax protocol and
timing needs. search archive for very good explanations of this problem.
-- 
Steven Critchfield  [EMAIL PROTECTED]

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Let's TALK ABOUT IT!!!

[Armand A. Verstappen]

On Sat, 2003-10-04 at 18:53, Rich Adamson wrote:
 Why not add an Article to the www.voip-info.org site, and those that are
 interested with helping can list their FWD, IAXTEL, or other access number,
 probable hours of availability, any special focus skills, size of their
 current * environment, etc?
 
 I'm game.

Sounds good. Wouldn't it be possible to login to the queue of an *
server providing this servers from my extension through my *
installation? That way calls could be routed to available volunteers.

wkr,

-- 
Envida http://www.envida.net/
Armand A. Verstappen   Graadt van Roggenweg 328
[EMAIL PROTECTED]   3531 AH Utrecht
tel: +31 (0)30 298 2255Postbus 19127
fax: +31 (0)30 298 21113501 DC Utrecht


signature.asc
Description: This is a digitally signed message part

RE: [Asterisk-Users] IVR Questions?

[Joe Dennick]

OK, I've been playing with it and I must be missing something.  Here's a
script that I've written:

#!/usr/bin/perl

use Asterisk::AGI;

$AGI = new Asterisk::AGI;

my $repid =
$AGI-get_data('sai-enter-rep_id', 5, 5);

$AGI-say_digits($repid);

exit;

The script is called as part of Extension 501 (extensions.conf):

exten = 501,1,Wait(1)
exten =
501,2,EAGI(/home/joed/asterisk-perl-0.08/examples/sai-get-repid.agi)
exten = 501,3,Wait(1)
exten = 501,4,Hangup

And the CLI Output looks like:

-- Executing Wait(SIP/7002-7751, 1) in new stack
-- Executing EAGI(SIP/7002-7751,
/home/joed/asterisk-perl-0.08/examples/sai-get-repid.agi) in new stack
-- Launched AGI Script
/home/joed/asterisk-perl-0.08/examples/sai-get-repid.agi
-- Playing 'sai-enter-rep_id'
-- AGI Script
/home/joed/asterisk-perl-0.08/examples/sai-get-repid.agi completed,
returning 0
-- Executing Wait(SIP/7002-7751, 1) in new stack
-- Executing Hangup(SIP/7002-7751, ) in new stack

So, I know that is sees the script, and the stream file for playback,
but I never hear the playback (Prompt to enter code), and I don't get
any error messages anywhere.

Does anyone have an idea of what I'm missing?

Thank you for your assistance!


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of duncan
Sent: Monday, October 06, 2003 7:53 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] IVR Questions?



I failed to mention in my original post that I've looked at perl 
through
AGI, but haven't yet found a function that allows me to capture digits
to 
a variable that I can then manipulate.  I probably should also mention 
that I'm not a programmer-type, although I can usually muddle through 
simple scripts for smaller uses like this one.  Can you suggest which 
function I would use?

using the asterisk-perl module from http://asterisk.gnuinter.net/

you would use the get_data function:

my $captured_dtmfs = 
$AGI-get_data('/var/lib/asterisk/sounds/whatever','5','4');

from the show agi get data help

show agi get data
Usage: GET DATA file to be streamed [timeout] [max digits]
  Stream the given file, and recieve DTMF data. Returns the
digits 
recieved from the channel at the other end.


hope this helps


duncan

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Modem and Fax over VoIP

[Andrew Joakimsen]

Fax with G711 works fine. Modem will be slow, but if you really need to
use it slown them down to 28.8 or 33.6

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Eduardo Goncalves
 Sent: Monday, October 06, 2003 2:56 PM
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] Modem and Fax over VoIP
 
 On Mon, 06 Oct 2003 13:43:21 -0500
 Steven Critchfield [EMAIL PROTECTED] wrote:
 
  On Mon, 2003-10-06 at 10:12, Eduardo Goncalves wrote:
   Hello,
  
 I have the fowling scenario:
  
   fxs[asterisk1]-iax-[asterisk2]e1em---PSTN
 
 
  If asterisk2 is your only access to the PSTN, then it doesn't make a
lot
  of sense to do fax over VoIP. Put a couple of modems on asterisk2
with
  matching FXS ports and learn to use Hylafax.
 
   I can't. Because asterisk1 is at a remote site. And the only
access
 to PSTN at this remote site is trough asterisk2.
 
  performance out of the faxing and potentially lower bills. Also it
is
  more flexible. Not to mention that the bandwidth over VoIP to make
data
  quality calls is around 80k of VoIP to get flakey 9.6k. Do you need
to
  waste 10x bandwidth for sub par functionality?
 
   Is it possible to config asterisk1 to change to g.711 only when
a
 fax transmission is detected?
 
 Thanks
 Eduardo
 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED]
 http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] X100P too quiet

[Ed Dack]

I've got * up and running everything seems to work ok except for when you
dial out using the X100P card.

Everything sounds great this end but the person you call complains that they
can't hear you very well (Very Whispered). 

Is their any way to turn up the volume.  I've fiddled with the gain settings
in the zapata.conf file but to no avail.

Any help would be much appreciated!

-Ed 

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Re: Modem and Fax over VoIP

[George Pajari]

Eduardo asks:
 I want to know the steps to transmit fax from a machine
 connected to the fxs to a fax machine on the PSTN.
It was suggested:
 Put a couple of modems on asterisk2 with
 matching FXS ports and learn to use Hylafax.
Not to take anything away from Hylafax, but our commercial fax server 
software products run on Linux and play well with Asterisk and other 
VoIP solutions. We also provide support.

Please contact me out-of-band if we can be of assistance.

g.

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Modem and Fax over VoIP

[Steven Critchfield]

On Mon, 2003-10-06 at 15:12, Andrew Joakimsen wrote:
 Fax with G711 works fine. Modem will be slow, but if you really need
 to
 use it slown them down to 28.8 or 33.6

This depends on if you have consistent latency and otherwise no jitter.

On my 12 hop link with the office over a cable modem, it is iffy to get
a modem connection up. The connection occasionally claimed 14400, but
never was over 1200 in actual output. So again, this is chancy at best
and in the end will result in longer calls between fax machines. A cheap
modem, and a $10 scanner will solve the problem and keep you faxes going
out fast.

  Steven Critchfield [EMAIL PROTECTED] wrote:
  
   On Mon, 2003-10-06 at 10:12, Eduardo Goncalves wrote:
Hello,
   
I have the fowling scenario:
   
fxs[asterisk1]-iax-[asterisk2]e1em---PSTN
  
  
   If asterisk2 is your only access to the PSTN, then it doesn't make a
 lot
   of sense to do fax over VoIP. Put a couple of modems on asterisk2
 with
   matching FXS ports and learn to use Hylafax.
  
  I can't. Because asterisk1 is at a remote site. And the only
 access
  to PSTN at this remote site is trough asterisk2.
  
   performance out of the faxing and potentially lower bills. Also it
 is
   more flexible. Not to mention that the bandwidth over VoIP to make
 data
   quality calls is around 80k of VoIP to get flakey 9.6k. Do you need
 to
   waste 10x bandwidth for sub par functionality?
  
  Is it possible to config asterisk1 to change to g.711 only when
 a
  fax transmission is detected?
  
  Thanks
  Eduardo
  ___
  Asterisk-Users mailing list
  [EMAIL PROTECTED]
  http://lists.digium.com/mailman/listinfo/asterisk-users
 
 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED]
 http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
Steven Critchfield  [EMAIL PROTECTED]

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] runing asterisk and apache

[Peter Brown]

DAve,

JUst wondering whether you can disclose the number of users you have on
your system and what CPU memory and disks you have. I'm looking to do
multiple functions on a single boxen too.


Peter

At 16:25 6/10/2003 +0200, you wrote:
On Mon, 2003-10-06 at 15:57, listas iPfone wrote:
 Hi All,
 
 I´m thinking in install apache in my asterisk machine to host a litle site.

Asterisk + Openvpn + Apache + MySql + Postfix + Amavisd + Sophie + local
mirror of Mandrake Cooker.

This is Linux at work not M$ at play. 
 
-- 
Dave Cotton [EMAIL PROTECTED]

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] callerid name modification (or adding)

[john lawler]

Is there any way to take an incoming callerid string and remove the 
given name part of it and replace it w/ something arbitrary, or add to 
a blank name string (possibly by looking up the number in a database)?

Thanks,

John Lawler

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] direct-inward-dialing (DID)

[john lawler]

I know that Asterisk supports DID, but does anyone have documentation on 
how to write the configuration for it?

I'll be trying to setup a hybrid system where some incoming numbers will 
be DID enabled and others won't, so I'll need to be able to sort between 
the two, i.e. directly connect the DID dialed numbers and route the 
others to an autoattendant for extension dialing.

Thanks,

John Lawler

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] callerid name modification (or adding)

[Steven Critchfield]

On Mon, 2003-10-06 at 15:45, john lawler wrote:
 Is there any way to take an incoming callerid string and remove the 
 given name part of it and replace it w/ something arbitrary, or add to 
 a blank name string (possibly by looking up the number in a database)?

Be glad I'm ridding my angst in other ways than lazy users these days.

show application setCidName

 -= Info about application 'SetCIDName' =- 

[Synopsis]:
  Set CallerID Name

[Description]:
  SetCIDName(cname[|a]): Set Caller*ID Name on a call to a new
value, while preserving the original Caller*ID number.  This is
useful for providing additional information to the called
party. Sets ANI as well if a flag is used.  Always returns 0


-- 
Steven Critchfield  [EMAIL PROTECTED]

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] direct-inward-dialing (DID)

[Steven Critchfield]

On Mon, 2003-10-06 at 15:45, john lawler wrote:
 I know that Asterisk supports DID, but does anyone have documentation on 
 how to write the configuration for it?
 
 I'll be trying to setup a hybrid system where some incoming numbers will 
 be DID enabled and others won't, so I'll need to be able to sort between 
 the two, i.e. directly connect the DID dialed numbers and route the 
 others to an autoattendant for extension dialing.

Dude, get with the reading and googling. You sent this question less
than 1 minute after your last question according to your computer.

google using 
DID documentation site:lists.digium.com
or
http://www.google.com/search?hl=enie=UTF-8oe=UTF-8q=DID+documentation+site%3Alists.digium.combtnG=Google+Search

5 lines down you will find 
http://lists.digium.com/pipermail/asterisk-users/2003-January/007077.html


-- 
Steven Critchfield  [EMAIL PROTECTED]

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Echo Cancellation

[Stephen R. Besch]

I have been struggling with echo cancellation for the last few days.  It 
seems to me that it would be useful to start up a technical discussion 
of the issue so that we don't have to solve the problem empirically.  My 
system is SIP (Grandstream) = Asterisk = Adtran TSU600 =FXO 
=POTS.  From what I can tell from testing on my system the echo has 
the following characteristics:

1) It varies over time.  It is worst at the start of a call and after 
periods of silence.  From this I conclude that the echo canceller is 
adaptive, adjusting delay and amplitude based on real time signal analysis.

2) The echo is only heard on the SIP phones, not on the far end of the 
POTS connection.  In spite of seeming somewhat mysterious at first, I 
concluded that this is actually the expected behavior.  Note: Voice is 
digitized at SIP phone, transmitted (with delay) to the FXO card and 
converted to analog.  The analog signal is placed on the 2-wire POTS 
line, where it loops back to the analog receiver in the FXO card and is 
digitized and sent back to the SIP phone as an echo. The POTS end hears 
no echo because there is no pathway for the echoed packets to get back 
to the POTS.

From this, it seems to me that the ideal place to deal with the echo 
would be at the point of conversion (from A to D and back), since there 
would be virtually no delay at this point and it would be simpler to 
determine the correct amplitude correction.  If this is the case, then 
it explains why adjusting receiver and transmitter gains would aid in 
cancelling echo, with the caveat that it should only work in those cases 
where the Rx ant Tx gains that are being adjusted are those which are 
right at the A/D and D/A converters.  In other words, adjustment of 
these values in Zaptel.conf should have no effect if I am using an 
outboard channel bank.

What I would like is for the real experts to jump into this discussion, 
lay out the real theory and technical details of the echo cancellation 
used in asterisk so that we can all make more intelligent attacks on the 
echo problem.  How about it?

Stephen R. Besch

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] IVR Questions?

[Paul Crick]

Try putting an Answer() in your extensions.conf before you call the AGI
code - a common gotcha I think?

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] callerid name modification (or adding)

[Robert Hajime Lanning]

quote who=Steven Critchfield
 On Mon, 2003-10-06 at 15:45, john lawler wrote:
 Is there any way to take an incoming callerid string and remove the
given name part of it and replace it w/ something arbitrary, or add
to a blank name string (possibly by looking up the number in a
database)?

 Be glad I'm ridding my angst in other ways than lazy users these days.

 show application setCidName

  -= Info about application 'SetCIDName' =-

 [Synopsis]:
   Set CallerID Name

 [Description]:
   SetCIDName(cname[|a]): Set Caller*ID Name on a call to a new
 value, while preserving the original Caller*ID number.  This is
 useful for providing additional information to the called
 party. Sets ANI as well if a flag is used.  Always returns 0

*CLI show application LookupCIDName
  -= Info about application 'LookupCIDName' =-

[Synopsis]:
  Look up CallerID Name from local database

[Description]:
  LookupCIDName: Looks up the Caller*ID number on the active
channel in the Asterisk database (family 'cidname') and sets the
Caller*ID name.  Does nothing if no Caller*ID was received on the
channel.  This is useful if you do not subscribe to Caller*ID
name delivery, or if you want to change the names on some incoming calls. 
Always returns 0.

-- 
END OF LINE

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] direct-inward-dialing (DID)

[Paul Crick]

It's pretty easy, in your extensions.conf.

If your DIDs are in a range, you could set up some pattern matching to take
a block of incoming DIDs and map to extension numbers then dial or hand off
to the dial'n'voicemail macro thing. If your DIDs are non-contiguous, you'll
have to set up a separate entry for each one.

Example: 456-7000 is your main number and you have 7001 to 7099 as DIDs:

exten = 7000,1,Goto(AutoAttendant|s|1)
exten = _7XXX,1,Macro(yourdialmacro|${EXTEN})

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] getting inbound caller-id from sip remote-party-id field

[Steve Dolloff]

I am looking for examples or instructions on how to route calls to
voicemailmain based on remote-party-id.

I have the following entry in my extensions.conf file:

exten = 200,1,Voicemailmain(${CALLERIDNUM})

I am routing calls to * via SER and sending Remote-Party-ID in the SIP
headers.  I am trying to find out how to map the SIP Remote-Party-ID
field to caller-id so that I can use it to map to the correct extension
for voicemail retrieval.

Thanks,

Stephen
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Conferencing Calls on Cisco 7940

[Babak Pasdar]

Hello,

I am trying to conference two or more calls on a Cisco 7940 phone.  When I have one 
inbound call and one outbound (I initiate the second call by pressing conference) I 
get the join button at the bottom of the screen and I can conference.

When I initiate both calls or I receive both calls I dont get the join button.  As a 
side question what would represent the hook flash on a Cisco 7940 or is this 
capability not possible.

Thanks

Babak 

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Snom100 H.323 sample config

[Tilghman Lesher]

I'm trying to get a Snom100 configured with H.323.  Right now, the
phone is not even connecting to the Asterisk server, so there's
obviously a problem with the snom config.  Does anybody have a
sample working configuration with the snom phone, using H.323?

I've checked the archives, but everybody seems to be using SIP with
the Snom phone, not H.323.

-Tilghman

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] IVR Questions?

[Joe Dennick]

That makes a lot of sense, but...it still doesn't work.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Crick
Sent: Monday, October 06, 2003 4:18 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] IVR Questions?


Try putting an Answer() in your extensions.conf before you call the AGI
code - a common gotcha I think?

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Digium TDM400P and Analog DID Trunks

[George Pajari]

Has anyone any experience using the TDP400P to connect to analog DID 
(Direct Inward Dialling) trunks?

Analog DID trunks are the opposite to non-DID analog lines and require 
an FXS port (rather then the FXO used for non-DID analog lines).

Any hints or pointers much appreciated. A google search did not reveal 
anything.

g.

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Grandstream 102

[John Vozza]

I haven't checked in a few months but while the info below is correct the
102 limits the PC Lan port to 10mb even if using a 100mb NIC card.

Can anyone else confirm or deny this?

John
-
NetRom Internet Services973-208-1339 voice
[EMAIL PROTECTED]   973-208-0942 fax
http://www.netrom.com
-


On Sun, 5 Oct 2003, Michael T Farnworth wrote:

 Typically you run a cable into the phone, then a cable out of the phone
 into the computer, it appears to just be a bridge.  It also works fine to
 run the cable into the phone and then a cable into another phone and then
 into a computer.

 Michael

 On Sun, 5 Oct 2003, Nicolas Gudino wrote:

  Sorry about this off-topic question... I want to know if the second ethernet port 
  on the Grandstream 102 phone works as a bridge to connect from there to a PC. Do
  I need two ethernet jacks to connect a phone and a PC, or this phone let me 
  connect both with only one? Thanks in advance!
 
  Nicolas Gudino
 


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Web Voicemail Permissions

[Tilghman Lesher]

On Monday 06 October 2003 05:13 pm, Carlton J. O'Riley wrote:
 Are there any plans to incorporate the running of Asterisk as a
 non-root user into the current CVS?  There is nothing in Asterisk
 that requires root access as far as I know and this would solve the
 vmail.cgi script permissions problem.

Here's a reason why it might need to run as root:
bash# ls -l /dev/zap/ctl
crw-r--r--1 root root 196,   0 Oct  6 13:15 /dev/zap/ctl

-Tilghman

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] IVR Questions?

[Paul Crick]

 That makes a lot of sense, but...it still doesn't work.
DOH! :-(

Hmm.. how are you connecting to the box? Zaptel device? SIP connection? I
wonder what audio format's being used?

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] IVR Questions?

[Richard Lyman]

simply add...

..
my $AGI = new Asterisk::AGI;
my %input = $AGI-ReadParse();  ##  this line
..

Joe Dennick wrote:
 
 That makes a lot of sense, but...it still doesn't work.
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Paul Crick
 Sent: Monday, October 06, 2003 4:18 PM
 To: [EMAIL PROTECTED]
 Subject: RE: [Asterisk-Users] IVR Questions?
 
 Try putting an Answer() in your extensions.conf before you call the AGI
 code - a common gotcha I think?
 
 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED]
 http://lists.digium.com/mailman/listinfo/asterisk-users
 
 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED]
 http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] MP3s in /var/lib/asterisk/mohmp3 causes Asterisk crash

[Leif Madsen]

Hi All,

I have just compiled the newest version of mpg123 on a RedHat 9.0 system 
(mpg321 has not been installed) and I am using the newest CVS version of 
asterisk.  Whenever I place any mp3 files in the 
/var/lib/asterisk/mohmp3/ directory, Asterisk crashes a horrible fiery 
death.

If mp3s exist in that directory, then I can't even start Asterisk.  If I 
start it without files then copy them over, then it crashes.  If I use 
the mp3player() app then it crashes.  Using mpg123 -t filename.mp3 seems 
to be fine.

If you have any idea's on things I can test for, it'd be greatly 
appreciated.

Thanks in advance,

+---+
|Leif Madsen - http://www.hacklocalhost.com |
+---+
|@| leif at hacklocalhost dot com   |
|  FWD| 18924   |
|  IAX| 1700-363-0761   |
|  SMS| sms at hacklocalhost dot com|
|  ICQ| 3445119 |
|iptel| 8972-1969   |
|sipph| 1-747-386-1618  |
+---+
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] SIP X100P Echo Problems

[Brian Schrock]

Like most others on this list I also have some really annoying echo whenever
a call goes out to the PSTN from a SIP phone...

SNOM/Budgettone - Asterisk - X100P - PSTN

I have tried every echo canceler in the makefile and turned on and off
aggressive suppressor etc. etc. etc. tried 32,16,128, and 256 bridgetaps and
I can get it reduced to only a few seconds on the intro of the call and
after silence, as well as a really annoying beep every so often, and some
audio artifacts.

I am using KT400 VIA based motheroards MSI KT4VL and x100P and the new
tdm400p. Calls within the pbx sound great something to really be proud of
(congrats to all of you developers), but going out to the PSTN is extremely
annoying.

Who do I pay and how much to get rid of this extrordinarily annoying echo
from sip - pstn calls? I am not kidding and I also hope others on this list
who are making money on asterisk would chip in to help out.

This problem is killing me.

Brian J. Schrock
Anistone Technologies, LLC
6926 Avery Rd.
Dublin, OH 43017
Phone: 614-798-9106


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] SIP X100P Echo Problems

[Mark Spencer]

I have a new echo can I'm working on, let me see if I can get it fixed
tonight and in CVS.

Mark

On Mon, 6 Oct 2003, Brian Schrock wrote:

 Like most others on this list I also have some really annoying echo whenever
 a call goes out to the PSTN from a SIP phone...

 SNOM/Budgettone - Asterisk - X100P - PSTN

 I have tried every echo canceler in the makefile and turned on and off
 aggressive suppressor etc. etc. etc. tried 32,16,128, and 256 bridgetaps and
 I can get it reduced to only a few seconds on the intro of the call and
 after silence, as well as a really annoying beep every so often, and some
 audio artifacts.

 I am using KT400 VIA based motheroards MSI KT4VL and x100P and the new
 tdm400p. Calls within the pbx sound great something to really be proud of
 (congrats to all of you developers), but going out to the PSTN is extremely
 annoying.

 Who do I pay and how much to get rid of this extrordinarily annoying echo
 from sip - pstn calls? I am not kidding and I also hope others on this list
 who are making money on asterisk would chip in to help out.

 This problem is killing me.

 Brian J. Schrock
 Anistone Technologies, LLC
 6926 Avery Rd.
 Dublin, OH 43017
 Phone: 614-798-9106


 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED]
 http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Message Waiting on Cisco 7960

[Juan J. Sierralta P.]

On Fri, 2003-10-03 at 14:53, Babak Pasdar wrote:
 This issue was resolved by adding the @context in the voicemail.conf
 file for the extension to the mailbox=XXX command.
 
 [EMAIL PROTECTED]
 
 Thanks so much for your help.

Is there anything special I need to configure on the Cisco phone to get
MWI ?
Because a I have a 7960 hanging from asterisk and I have followed all
the suggestions here and I have not MWI on the phone.
Here are my confs:

--- extensions.conf -
[demo]
exten = 8991,1,Dial(SIP/8991,20)
exten = 8991,2,Voicemail2([EMAIL PROTECTED])
exten = 8991,102,Voicemail2([EMAIL PROTECTED])
exten = 8991,103,Hangup

 voicemail.conf -

[demo]
8991 = 8991,Juanjo,[EMAIL PROTECTED]

--- sip.conf 

[8991]
type=friend
username=8991
secret=
nat=no  ; This phone may be natted
host=dynamic
canreinvite=no  ; Cisco poops on reinvite sometimes
qualify=500 ; Qualify peer is no more than 200ms
away
;defaultip=192.168.0.15
context=demo
[EMAIL PROTECTED]

Any sugestions will be appreciated.

-- 
Juanjo sin .sig

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] MP3s in /var/lib/asterisk/mohmp3 causes Asterisk crash

[Leif Madsen]

Leif Madsen wrote:

Hi All,

I have just compiled the newest version of mpg123 on a RedHat 9.0 system 
(mpg321 has not been installed) and I am using the newest CVS version of 
asterisk.  Whenever I place any mp3 files in the 
/var/lib/asterisk/mohmp3/ directory, Asterisk crashes a horrible fiery 
death.

If mp3s exist in that directory, then I can't even start Asterisk.  If I 
start it without files then copy them over, then it crashes.  If I use 
the mp3player() app then it crashes.  Using mpg123 -t filename.mp3 seems 
to be fine.

If you have any idea's on things I can test for, it'd be greatly 
appreciated.
So yah.. apparently the 0.59s version compiled with linux-mmx doesn't 
play too nicely with whatever Asterisk is calling it as.  Downgrading 
to version 0.59r seemed to fix the problem.

Nothing to see here, please move along.

Thanks,

+---+
|Leif Madsen - http://www.hacklocalhost.com |
+---+
|@| leif at hacklocalhost dot com   |
|  FWD| 18924   |
|  IAX| 1700-363-0761   |
|  SMS| sms at hacklocalhost dot com|
|  ICQ| 3445119 |
|iptel| 8972-1969   |
|sipph| 1-747-386-1618  |
+---+
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Conferencing Calls on Cisco 7940

[Brian West]

Works fine on my 7960 with 5.3 firmware.

bkw

On Mon, 6 Oct 2003, Babak Pasdar wrote:



 Hello,

 I am trying to conference two or more calls on a Cisco 7940 phone.  When I have one 
 inbound call and one outbound (I initiate the second call by pressing conference) I 
 get the join button at the bottom of the screen and I can conference.

 When I initiate both calls or I receive both calls I dont get the join button.  As a 
 side question what would represent the hook flash on a Cisco 7940 or is this 
 capability not possible.

 Thanks

 Babak

 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED]
 http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] ISDN Dialout

[Jay Tyndall]

Hi,

I am having some trouble with ISDN Dialout. Using a Netjet-s PCI Card.

When in Minicom, the only way I can dialout is if i issue ATS18=1 First.
Otherwise I get a BUSY message.  So thats fine.
But when I dialout from asterisk, I get an immediate hangup, so my guess is 
that asterisk is not issuing ATS18=1 to the ttyI device.

Here are my configs, any input would be greatly appriciated.



extensions.conf
exten = 4000,1,Dial,Modem/g3:0422xx|60|r


modems.conf
; net jet suff
context=default
msn=0269xxL*
icomingmsn=0269xxL*
driver = i4l
group=2
stripmsd=1
mode=immediate
device = /dev/ttyI0
msn=0269xx
group=3
device = /dev/ttyI1
mode = immediate
type = autodetect
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Message Waiting on Cisco 7960

[Chad Sawyer]

 --- sip.conf 
 
 [8991]
 type=friend
 username=8991
 secret=
 nat=no  ; This phone may be natted
 host=dynamic
 canreinvite=no  ; Cisco poops on reinvite sometimes
 qualify=500 ; Qualify peer is no more than 200ms
 away
 ;defaultip=192.168.0.15
 context=demo
 [EMAIL PROTECTED]  -- use mailbox=8991
 

use 
mailbox=500

instead of [EMAIL PROTECTED]

where 500 is your voicemail box in voicemail.conf

I don't think you have to specify a context

Chad
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] IVR Questions?

[Joe Dennick]

I added the line suggested below, and now I hear the prompt for input,
but then nothing.  The CLI says its playing the prompt, and nothing
more.  When I finally end the call (hang up the phone)(BTW, I'm using
X-Ten's soft-phone for testing) Asterisk crashes and has to be
restarted.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Richard
Lyman
Sent: Monday, October 06, 2003 6:09 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] IVR Questions?


simply add...

..
my $AGI = new Asterisk::AGI;
my %input = $AGI-ReadParse();  ##  this line
..

Joe Dennick wrote:
 
 That makes a lot of sense, but...it still doesn't work.
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Paul Crick
 Sent: Monday, October 06, 2003 4:18 PM
 To: [EMAIL PROTECTED]
 Subject: RE: [Asterisk-Users] IVR Questions?
 
 Try putting an Answer() in your extensions.conf before you call the 
 AGI code - a common gotcha I think?
 
 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED] 
 http://lists.digium.com/mailman/listinfo/asterisk-users
 
 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED] 
 http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] how many production systems are there?

[Garry Adkins]

 The biggest feature we hope to offer, which we're going to call 
 Wireless InterCon, allows customers who opt for the service to expose 
 a local PSTN line for sharing with other members of the club.  Because 
 we operate across many ILEC exchanges and two LATAs, we have the ability 
 to route calls privately and then out the PSTN as a local call to 
 exchanges that may be far away.  The principle of locality of 
 reference extends past the CO, and so many people are excited about 
 being able to talk to others in nearby towns for whom the call is now a 
 toll charge.

Isn't this what Worldcom is in trouble for?  (one of many...)

(running lines across lata as a local call)?

-G

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Message Waiting on Cisco 7960

[Brian West]

 use
 mailbox=500

 instead of [EMAIL PROTECTED]

[EMAIL PROTECTED]

since he doesn't have his stuff in the default context

bkw
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Re: DB virtualization for multiple database support - Was Re: [Asterisk-Users] How to use vmdb.sql in voicemail.conf/extension.conf

[Garry Adkins]

Not familiar with it...  You have a URL?


- Original Message - 
From: Matteo Brancaleoni [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, October 05, 2003 4:52 PM
Subject: [Asterisk-Users] Re: DB virtualization for multiple database
support - Was Re: [Asterisk-Users] How to use vmdb.sql in
voicemail.conf/extension.conf


 Like what PEARS (php libs) do for db backends?

 Matteo.

 Garry Adkins wrote:
 
 I am trying a scenerio where the * will take the email and mailbox
  number from the Mysql database for sendming mail to a voicemail user. I
  have seen vmdb.sql file but is not able to determine its use.
 
You can't anymore MySQL was ripped from Asterisk because the client
libs
are GPL.
 
  I would be more than happy to help write a DB Virtualization function
  for *.
 
  I *love* the way it works in Java, but that's not a real possibility.
  It wouldn't need to be as complicated as JDBC but it's a nice model.
 
  We could however:
  1)  Abstract out the schema from the database calls
  2)  Develop a pluggable driver interface to translate to whatever DB
  you're using.
 
  This way...  You want MySQL, you develop a translation driver that
  maps * db calls to MySql.  (fairly trivial)
  Same for Postgres  (I'd suggest making this the default, as no GPL
  issues for mark, etc.)
  Same for Oracle, etc.


 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED]
 http://lists.digium.com/mailman/listinfo/asterisk-users




___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Message Waiting on Cisco 7960

[Kevin]

Use in your sip.conf:
[EMAIL PROTECTED]

You need to use a the @context with voicemail2

Kevin,

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Juan J.
Sierralta P.
Sent: Monday, October 06, 2003 6:58 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Message Waiting on Cisco 7960


On Fri, 2003-10-03 at 14:53, Babak Pasdar wrote:
 This issue was resolved by adding the @context in the voicemail.conf
 file for the extension to the mailbox=XXX command.

 [EMAIL PROTECTED]

 Thanks so much for your help.

Is there anything special I need to configure on the Cisco phone
to get MWI ?
Because a I have a 7960 hanging from asterisk and I have
followed all the suggestions here and I have not MWI on the phone.
Here are my confs:

--- extensions.conf -
[demo]
exten = 8991,1,Dial(SIP/8991,20)
exten = 8991,2,Voicemail2([EMAIL PROTECTED])
exten = 8991,102,Voicemail2([EMAIL PROTECTED])
exten = 8991,103,Hangup

 voicemail.conf -

[demo]
8991 = 8991,Juanjo,[EMAIL PROTECTED]

--- sip.conf 

[8991]
type=friend
username=8991
secret=
nat=no  ; This phone may be natted
host=dynamic
canreinvite=no  ; Cisco poops on reinvite sometimes
qualify=500 ; Qualify peer is no more than 200ms
away
;defaultip=192.168.0.15
context=demo
[EMAIL PROTECTED]

Any sugestions will be appreciated.

--
Juanjo sin .sig

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users



___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Help with questions for initial Asterisk wizard (GUI)

[Leif Madsen]

Hey all,

I am in the middle of creating a new user wizard which will generate all 
the .conf's the new Asterisk user will require to get themselves up and 
running in Asterisk without having to touch a single configuration file. 
 This is what I have come up with as a rough draft.  It is far from 
complete, so I'm asking people to submit things that should be added, 
changed, removed etc.. etc...  so please help me come up with a good 
logic for the questions so that I may start the work on the actual wizard!

Thanks in advance for all your help!

Hardware
Are you using Digium Hardware? (yes/no)
+ TE410P
+ How many?
+ TDM400P
+ How many?
+ How many modules (each card)
+ T400P
+ How many?
+ T100P
+ How many?
+ E100P
+ How many?
+ X100P
+ How many?
Other hardware
+ Do you have a soundcard? (yes/no)
+ ALSA or OSS?
Do you have any SIP devices? (yes/no)
+ SIP phone
+ Softphone
Extension Ranges
+ Start and Stop range
+ Would like you to enable voicemail on any of these extensions? (yes/no)
+ all of them (yes/no)
+ which ones?
+ What should the default voicemail password be?

+ Default formats for writing voicemail
+ GSM, wav49, WAV
+ Email notification? (yes/no)
+ Who should the email appear to come from?
+ Should we attach it to the email?
+ Would you like to specify a maximum message length? (yes/no)
+ How long?
+ Would you like to specify a maximum greeting length? (yes/no)
+ How long?
+ Which country are you in? (for indication)
+ United States
+ Australia
+ France
+ Netherlands
+ United Kingdom
+ Which language? (for zapata.conf)
Would you like to activate any of these extensions now? (yes/no)
	+ List extensions with CONFIG | EDIT | DELETE | ADD links
		+ Who is going to use this extension? (name)
		+ What is the email address of the person as this extension? (for 
email notification)
		+

For each channel of the hardware the user has
+ Which signalling for this channel?
kewl start
loop start
ground start
+ Enable three way calling?
+ Enable transfer?
+ Enable call waiting?
+ Enable busy detection?
+ Use CallerID?
+ rxgain
+ txgain
+ Immediate? (yes/no)
+ CallerID String
+ Name
+ Number
+ Enable mailbox indication?
+ Mailbox number(s) to be associated with this channel
+ Context
Would you like to setup registration for FWD, IAXtel, SIPPhone or iptel?
+
--
+--+
|Leif Madsen - http://www.hacklocalhost.com|
+--+
|@| leif at hacklocalhost dot com  |
|  SMS| sms at hacklocalhost dot com   |
|  FWD| 18924  IAX| 1700-363-0761  |
|iptel| 8972-1969sipph| 1-747-386-1618 |
+--+
--
+--+
|Leif Madsen - http://www.hacklocalhost.com|
+--+
|@| leif at hacklocalhost dot com  |
|  SMS| sms at hacklocalhost dot com   |
|  FWD| 18924  IAX| 1700-363-0761  |
|iptel| 8972-1969sipph| 1-747-386-1618 |
+--+
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Grandstream 102

[Dave Cotton]

On Tue, 2003-10-07 at 00:44, John Vozza wrote:
 I haven't checked in a few months but while the info below is correct the
 102 limits the PC Lan port to 10mb even if using a 100mb NIC card.
 
 Can anyone else confirm or deny this?
 
Yes my system indicates a 10mb connection.
-- 
Dave Cotton [EMAIL PROTECTED]

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Start...

[Manoj K Gupta]

If u are using the latest CVS then i suppose that u don;t need to do
anything except configuring  h323.conf for the ip address of the h323 gK.Or
u can also use oh323 channe driver available from www.inaccessnetwork.com .

The default sample works fine for test purposes.

Rgds
Manoj K Gupta

- Original Message - 
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, October 06, 2003 11:46 PM
Subject: [Asterisk-Users] Start...


 Hi all!

 One easy question... I hope someone will answer me.

 I've installed asterisk with the samples. Somewhere in my network I have
an
 H.323 Gatekeeper. What must I do to make that the gatekeeper talk with
 Asterisk?

 And I another little question... with the samples installed asterisk works
 ok? What must I install to see how it works?

 I am lost!! Please help me!

 See you.

 Mireia


 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED]
 http://lists.digium.com/mailman/listinfo/asterisk-users



___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users