It doesn't seem to work. The problems seem in RTP relay. SIP signals such as Register and INVITE are fine with virtual IP, but audio stream seems not able to feed back to callers. Is there a way to monitor where RTP goes to?
thanks! steven -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Tuesday, April 12, 2005 4:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] binding Asterisk to virtual IP right. So it probably requires to set externid. -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Daniel Bruce Lynes Sent: Tuesday, April 12, 2005 3:56 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [Asterisk-Users] binding Asterisk to virtual IP -----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----- _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
