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15124990483
2009 Mar 16
1
Could Asterisk be rewriting an incoming invite?
I'm not getting inbound audio from bandwidth.com. Their engineer said the
invite that they're sending me looks like this:
INVITE sip:+15129616808 at 67.198.16.18:5060;transport=udp SIP/2.0.
Record-Route: <sip:216.82.224.202;lr;ftag=VPSF506071629460>.
Record-Route: <sip:4.79.212.229;lr;ftag=VPSF506071629460>.
Via: SIP/2.0/UDP 216.82.224.202;branch=z9hG4bK6314.4f7b5d05.0.
Via: