This is not really the best way to handle unavailable redirection to VM
in asterisk. Also, for SIP only it's not a good idea to use the 'r'
parameter as this causes asterisk to generate a ringing signal even if
the endpoint is not responding. Please check the asterisk mailing list
or check the
voip-info.org wiki on asterisk sip.conf.
Dave
-----Original Message-----
From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On
Behalf Of Ahmed Boreau
Sent: 18 November 2004 14:23
To: yair(a)hakak.com; blairs(a)isc.upenn.edu
Cc: serusers(a)fox.iptel.org
Subject: Re: [Serusers] Ser + Asterisk
Thanks for you help.
below are what I did in * extensions.conf
[globals]
SERADDRESS=@IP ser server:5060
[serserver]
exten => _3XXX,1,Dial(SIP/${EXTEN}@${SERADDRESS},20,r)
exten => _3XXX,2,WaitMusicOnHold,15
exten => _3XXX,3,VoicemailMain(u${EXTEN:2})
In sip.conf, I just add
[general]
autocreatepeer=yes
Phones are ringing but I could not get voicemail.
Thanks in advance
At 13:36 18/11/2004, Yair Hakak wrote:
hello,
set autocreatepeer=yes in sip.conf and you should be fine.
obviously you need to make sure that things are set up properly in
extensions.conf to connect calls properly.
On Thu, 18 Nov 2004 13:32:26 +0000, Ahmed Boreau <ahmed.boreau(a)esmt.sn>
wrote:
> Hi,
>
> I need help. I'm actually trying to set up ser+asterisk which are
actually
> working separately.
> By now, I want to let asterisk receiving ser calls.
>
> I add these commands into ser.cfg
>
> if(method=="INVITE"){
> if (uri=~"^sip:1[0-9]{10}@*") {
> log(1,"Forwarding to Asterisk\n");
> rewritehostport("10.0.0.13: 5061");
> t_relay();
> break;
> }
> }
>
> What could I need to do into sip.conf at * side.
>
> Thanks in advance
>
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>
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>
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