On 4/13/05, Leif Madsen - Certified Asterisk Consultant
<[EMAIL PROTECTED]> wrote:
> On 4/12/05, Xu Wang <[EMAIL PROTECTED]> wrote:
> > Our Asterisk works fine with 'real' IP. But when we change the domain to a
> > virtual IP, the audio stream probably goes to the 'real' IP. There is no
> > sound coming back. Asterisk log shows that it does not hang up.
> >
> > Do you know what might be wrong?
> 
> This sounds like the bug currently being worked on in CVS. Please test
> the patch and submit feedback to the bug tracker.
> 
> To quote the bug description:
> 
> Currently if we have Asterisk SIP channel driver binding to all
> interfaces, and eth0 has many subnets attached to it (a primary
> 10.1.200.1, and then alias interfaces eth0:1 with 10.1.201.1, eth0:2
> with 10.1.202.1, eth0:3 with 10.1.203.1..
> 
> If an INVITE is sent to Asterisk on 10.1.202.1 (eth0:2) the response
> is always returned to 10.1.200.1. We need it to come back to
> 10.1.202.1.
> 
> Here is a direct link to the bug and much more information.
> 
> http://bugs.digium.com/bug_view_page.php?bug_id=0002358


Or if you can't/don't want to do that you could always try setting the
ip address to bind to in sip.conf and see if that fixes the problem. 
Sounds like it might.

Chris
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