[Asterisk-Users] Asterisk Newbie ...

[Julien]

It works now ...sorry but it was my linux box ... I had Sip express router
installed on this machine :-\
So my ip phones loged on S.E.R and not on asterisk ;)

My voice mail works fine :)))))
Just a last question, if i configure G723 in my ATA, i can't call the
voicemail for exemple. I've seen that messages were in GSM format. Is there
a way to be able to acces to the voice mail in G723 (for remote users) and
in G711 for local users ?

Thanks a lot all ;)
Julien.

----- Original Message ----- 
From: "Julien" <fabia at free.fr>
To: <asterisk-users at lists.digium.com>
Sent: Sunday, August 10, 2003 4:41 PM
Subject: Re: [Asterisk-Users] Asterisk Newbie ...


> With this configuration, the 1943, 1945 are available , it's ok
> but the 2999 is not available... In sjphone 404 error, on the ata busy
tone
> ...
>
> Julien.
>
> ----- Original Message ----- 
> From: "Andy Powell" <andy at beagles-den.demon.co.uk>
> To: <asterisk-users at lists.digium.com>
> Sent: Sunday, August 10, 2003 4:10 PM
> Subject: Re: [Asterisk-Users] Asterisk Newbie ...
>
>
> Fabia,
>
> The only numbers you should be able to dial from that config are
>
> 1945
> 1943
> 2999
>
> and nothing else...
>
> The entry under bogon-calls (isn't it bogus calls?) should read
>
> exten => s,1,Congestion
>
> rather that using the _. ...
>
> HTH
>
> Andy
>
> *********** REPLY SEPARATOR  ***********
>
> On 10/08/2003 at 15:13 Fabia wrote:
>
> >Hi ;)
> >
> >I'm a french newbie and i installed asterisk 1 day ago.
> >I've got an ATA186 and a computer with Sjphone installed.
> >
> >If i want to call the sjphone from the ata or call the ata from de
sjphone
> >everything is ok.
> >My problem is ,that i can't call the voicemail or any other phone number
> >..as 600 for exemple from the ata or the jphone.
> >I don't know why but i looked after a long time..
> >
> >here a copy of my extension.conf , sip.conf and voicemail.conf.
> >
> >Thanks for your help.
> >Julien.
> >
> >Extension.conf
> >
> >[general]
> >
> >static=yes
> >writeprotect=yes
> >
> >[bogon-calls]
> >exten => _.,1,Congestion
> >[from-sip]
> >exten => 1943,1,Dial(SIP/1943,5)
> >exten => 1943,2,Voicemail(u1943)
> >exten => 1943,102,Voicemail(b1943)
> >exten => 1943,103,Hangup
> >
> >exten => 1945,1,Dial(SIP/1945,6)
> >exten => 1945,2,Voicemail(u1945)
> >exten => 1945,102,Voicemail(b1945)
> >exten => 1945,103,Hangup
> >
> >exten => 2999,1,VoicemailMain(${CALLERIDNUM})
> >
> >
> >-----------------------------
> >sip.conf
> >
> >[general]
> >
> >port = 5060
> >bindaddr = 0.0.0.0
> >allow=all
> >context = bogon-calls
> >
> >[1943]
> >
> >type=friend
> >username=1943
> >secret=1943
> >host=dynamic
> >context=from-sip
> >mailbox=1943
> >
> >[1945]
> >
> >type=friend
> >username=1945
> >secret=1945
> >host=dynamic
> >context=from-sip
> >mailbox=1945
> >-----------------------
> >voicemail.conf
> >
> >[general]
> >
> >format=wav
> >
> >[local]
> >
> >1943 => 1943,Essai 1,xxx at yy.com
> >1945 => 1945,Essai2,rrr at ttt.bil
> >
> >_______________________________________________
> >Asterisk-Users mailing list
> >Asterisk-Users at lists.digium.com
> >http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
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