[Asterisk-Users] Controlling SIP mobile extensions.

2004-06-02 Thread XISCOAIR
Hi everybody,

I'm trying to develop a web application for controlling if SIP users 
are registered in * or not, and show it in a web.

My problem is that I don't now if it's possible to do a Shell Script to 
control this:
1. Connect to console.
2. Execute command.
3. Obtain users registered.
4. Update a BdD.

This is possible? There are any best way to implement this?

Thanks a lot.


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Re: [Asterisk-Users] Asterisk goes international :-)

2004-05-03 Thread XISCOAIR
Hi,

I'm from Spain and I have developed in Perl 2 scripts to say_number and 
say_digits in 3 diferents language (spanish, german and english). The 
problem is that I don't know how to adapt it to C in order to 
complement say.c

If somebody can help me, I will be very pleasured.

Thanks a lot.

- Mensaje Original -
De: Olle E. Johansson [EMAIL PROTECTED]
Fecha: Jueves, Abril 29, 2004 8:49 am
Asunto: Re: [Asterisk-Users] Asterisk goes international :-)

 Altus Snyman wrote:
 
  So what do I have to do to add South-Africa to this list?
  
 If you are saying numbers in a different way than english or if 
 you are thinking
 about another language, check the latest version of say.c in CVS 
 head and
 see if you can construct the syntax needed for your language.
 
 When you have a patch, open a bug report under 
 Internationalization in
 http://bugs.digium.com and add the patch there.
 
 I don't know how Asian languages work, but it would certainly be 
 interestingto see patches for those.
 
 Also, we need a larger group that works with the next generation 
 architecture.
 /O
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[Asterisk-Users] Choosing a VoIP Protocol.

2004-01-15 Thread XISCOAIR
Hello everybody,

I have the next scenario, working now.

ISDN Network AGW IP 1(*)IP NetworkGW IP 2(*)ISDN 
Network B

Now we are looking the avalibility of sending call's from 1 to 2, using 
an VoIP Protocol supported by *. But we have the next questions, before 
began to do it.

1. There are any VoIP protocol that allow us to send the source number 
from ISDN Network A and the destination number (in ISDN Network B), 
during the session establishment of the call.
2. It's possible that * take the destination number and dial it 
automatically without any user accion to the ISDN Network B.

Note, that we have E400P installed in IP GW2.

Thanks a lot for you help.

Bye bye


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[Asterisk-Users] Limit of conference rooms.

2003-11-04 Thread XISCOAIR
Hi, I'm trying to configure a system with dynamic's conference-room 
creation.

Anybody knows whats the limit of conference-room supported by 
asterisk?? and the rang of this conference-room (number)??

Thks a lot.




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[Asterisk-Users] Problems with MeetMe SIP / Mobile

2003-10-15 Thread XISCOAIR
Hi everybody,

I'm trying to use MeetMe(2000|p) in order to enter in a conference 
room. But when a mobile or a SIP call press '#' to go out from it, 
everybody goes out. Instead when a analg press it, all works fine.

Anyone else have this problem?? Can anybody help me please??

Thanks a lot.

Regards.


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[Asterisk-Users] Problems with MeetMe.

2003-10-13 Thread XISCOAIR
Good afternoon,
 
I'm trying to use MeetMe in an AGI script written in Perl, as follows:
 
print EXEC MeetMe 2000|p \n;
$res = checkresult();
 
The problem that I have is that when a user press '#' in order to exit 
from the conference, everybody goes out. This is randomized because 
sometimes doesn't happened.
 
My current version of asterisk is:
 
Asterisk CVS-05/22/03-11:14:50 built by [EMAIL PROTECTED] on a i686 running 
Linux
 
Does somebody knows the problem? It's a version problem of *???
 
I would be very pleasured if somebody can help me or told me if I'm 
mistaked about this functionality of MeetMe.
 
Thanks a lot.


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[Asterisk-Users] Problems with meetme.

2003-10-13 Thread XISCOAIR
Good afternoon,

I'm trying to use MeetMe in an AGI script written in Perl, as follows:

print EXEC MeetMe 2000|p \n;
$res = checkresult();

The problem that I have is that when a user press '#' in order to exit 
from the conference, everybody goes out. This is randomized because 
sometimes doesn't happened.

My current version of asterisk is:

Asterisk CVS-05/22/03-11:14:50 built by [EMAIL PROTECTED] on a i686 running 
Linux

Does somebody knows the problem? It's a version problem of *???

I would be very pleasured if somebody can help me or told me if I'm 
mistaked about this functionality of MeetMe.

Thanks a lot.
 



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