Hello 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? thanks! steven
Xu Wang wrote:> Hello > > 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?Did you look at rtp.conf? -- Always do right. This will gratify some people and astonish the rest. Mark Twain
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On Tuesday 12 April 2005 03:30 pm, Xu Wang 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.By virtual IP, you mean that you have several IP all attached to virtual network interfaces on the same machine, correct? i.e. they are named similar to: eth0, eth0:0, eth0:1, eth0:2, ...? -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.1 (GNU/Linux) iD8DBQFCXFIcgYKvkeyp3F4RAj/YAJ9j0Q+C7IfGtwbwBfoEmcXntpUAcwCfdR0T XlAq3K/8j7jb6djpAA1cbrw=1HDw -----END PGP SIGNATURE-----
Leif Madsen - Certified Asterisk Consultant
2005-Apr-13 07:48 UTC
[Asterisk-Users] binding Asterisk to virtual IP
On 4/12/05, Xu Wang <xwang@cascotec.com> 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 Thanks, Leif Madsen http://www.leifmadsen.com