Re: [asterisk-users] Authenticate by IP?
Saludos Carlos, Como vas a recibir las llamadas via SIP, puedes especificar el IP del host que te enviara las llamadas, por ej. Este es un bloque que tengo definido en el SIP.conf de uno de mis servers para enrutar las llamadas internacionales y a telefonos moviles utilizando un proveedor de terminacion. [oficina] type=peer context=from_office ; Esto va a mi 'extensions.conf' host=200.88.42.29; Este es el ip publico en la oficina (estatico) nat=no canreinvite=no qualify=yes disallow=all allow=g729 allow=ulaw Creo que eso contesta tu pregunta. -- Jose P. Espinal slackware-es.com Carlos Chavez wrote: I have a customer that needs an Asterisk server to sell minutes for cell phones in Mexico. I do not see a problem with that since he will get the calls by SIP and then use GSM adapters to get the calls into the GSM network. My problem is that his customers only want to be identified by IP and not by a username and password. Is there a way to authenticate just by using an IP address? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Authenticate by IP?
Carlos, No solo para enviar llamadas, sino tambien para recibir (de hecho, ese bloque que puse ahi lo uso para recibir, no para enviar). Te posteo un ejemplo del ejemplo que trae asterisk de sip.conf ;[sip_proxy] ; For incoming calls only. Example: FWD (Free World Dialup) ; We match on IP address of the proxy for incoming calls ; since we can not match on username (caller id) ;type=peer ;context=from-fwd ;host=fwd.pulver.com Como ves, para validar por IP se usa en el campo 'host' el ip o dominio de quien nos enviara la llamada :) -- Jose P. Espinal slackware-es.com Carlos Chavez wrote: On Mon, 2007-10-22 at 15:35 -0400, Rurouni Alucard wrote: Saludos Carlos, Como vas a recibir las llamadas via SIP, puedes especificar el IP del host que te enviara las llamadas, por ej. Este es un bloque que tengo definido en el SIP.conf de uno de mis servers para enrutar las llamadas internacionales y a telefonos moviles utilizando un proveedor de terminacion. [oficina] type=peer context=from_office ; Esto va a mi 'extensions.conf' host=200.88.42.29; Este es el ip publico en la oficina (estatico) nat=no canreinvite=no qualify=yes disallow=all allow=g729 allow=ulaw Creo que eso contesta tu pregunta. Hola José. Gracias por tu contestación. Lo que me estas especificando el para hacer llamadas de salida (PEER). Yo necesito autentificar a un usuario de entrada, voy a intentar haciendo algo parecido solo cambiando a type=user para ver si así funciona. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] H.323
Hi there, I have use the H.323 module that comes with asterisk-addons and i consider it (so far) VERY stable for my needs. Im talking about 10,000 minutes at month , + or - , and never had a crash or something bad about it. Personally, i recommend it, -- J. P. rakh at slackware-es dot com bilal ghayyad wrote: Hi List; Did any one tried the H.323 module? How much it is stable and work fine? Regards, ITS IP Telephony and Contact Center Engineer Eng. Bilal Ghayad Mobile: 00965 9849460 Ready for the edge of your seat? Check out tonight's top picks on Yahoo! TV. http://tv.yahoo.com/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial out from AGI
Hi Roy,
Look I dont know why u specify 'zap/1-1', but i do things like this on
my agi scripts a lot of times:
...
$stdin= fopen('php://stdin', 'r');
$stdout = fopen('php://stdout', 'w');
$stdlog = fopen('/tmp/outPUT.log', 'a');
...
fwrite($stdout,EXEC DIAL \Zap/g2/18092203555\ \n);
fflush($stdout);
// to see result //
$msg = fgets($stdin);
fwrite($stdlog,$msg. \n);
Try it out if u want...
J. Espinal,
Roy Kidder wrote:
I'm writing an AGI script and want it to dial a number on a channel
connected to the PSTN. It would look something like this (pseudo-code
follows):
if ($a){
dial(8005551212);
}else{
dial(866555);
}
The part I can't seem to get right is the dial function. I tried to
mimic the dial plan like so
sub dial($number){
print Dial(\Zap/1-1\, \Zap/g2/$number\)\n;
}
but I get the error
handle_exec: Could not find application (Dial(Zap/1-1,Zap/g2/866555)
Anyone have any suggestions?
Thanks,
Roy
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[asterisk-users] LinkSys SPA 2002 ATA hardphone UNREACHABLE...!!!
When using Grandstreamg Handytone ATA everything works fine incoming/ougoing but when using Linksys SPA 2002 ATA 'sip show peers' marks those extensions as UNREACHABLE and can't receive calls, but they can call out. Any Idea about possible reasons ?... Rurouni, ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Music On Hold not working with new 1.2.7.1 install
Did you check your mpg123 version ?, asterisk needs a specific version in order to work... - Original Message - From: Richard Reina To: asterisk-users@lists.digium.com Sent: Wednesday, June 07, 2006 6:02 AM Subject: [Asterisk-Users] Music On Hold not working with new 1.2.7.1 install I have followed the instructions provided at:http://voip-info.org/tiki-index.php?page=asterisk+config+musiconhold.confincluding installing asterisk-addons-1.2. I have left musiconhold.conf as is, calm-river et al are fine for now.However, no sound is heard and I get this message from the CLI when accessing MOH:-- Started music on hold, class 'default', on channel 'Zap/19-1'-- Stoped music on hold on Zap/19-1This happens whether it's a parked call or whether I access MOH directly via:exten = 800,1,Answerexten = 800,2,MusicOnHold()Any help would be greatly appreciated.Thank you very much.Richard __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
