[Asterisk-Users] Asterisk Newbie ...

[Eric Wieling]

No.  Asterisk only supports G723 in "pass thru" mode.  Calling voicemail
would require Asterisk to transcode from whatever format the voicemail
audio files are in (usually GSM) to G723 and Asterisk can't do that. 
Your best bet is to buy some G729 licenses from Digium and use that for
remote users.

On Mon, 2003-08-11 at 04:28, Julien wrote:
> 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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> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
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> >
> 
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