SIP Woes

Hello all,

I’ve been trying to play with asterisk (after a two month break) and am
having some problems getting my SIP connection to a third party provider
to work. In the asterisk console I notice:

I believe that’s some sort of SIP routing issue related to ReInvite’s ?

  • Is there a workaround for this? In the attempt that someone may be
    able to shed some light on the matter, I’ve uploaded my current
    configuration to:

I’ve also uploaded the output of ‘sip debug’. The interesting bit in
that (to me at least) is the message:

Is it so simple that I’ve missed something out in my outgoing bit on my
dialplan ? Anyway, the complete log can be found here:


Ohh. And:

If anyone would be so kind as to shed some insight into the matter it’d
be greatly appreciated!,

There does not appear to be any files at the link you posted, I get a 404. Are you having problems registering, making, or receiving calls? Post your related sip.conf, extensions.conf settings, and information about your network setup (IE: is the asterisk box behind NAT?).