[Asterisk-Users] SIP Changes???

[Olle E. Johansson]

Lars Boegild Thomsen wrote:
> Hi Everybody
> 
> Any significant changes to CVS HEAD over the last couple of days.  I've got
> two asterisk boxes - both on public IP but one is dynamic.  The one on
> dynamic IP registers at the other one - that part is fine.
> 
> Calls going from the one with dynamic to the static one goes fine.
> 
> Call the other way results now in:
> 
> Failed to authenticate user "1101" <sip:mydk at xxx.xxx.xxx.xxx>
At which server?

> 1101 is a SIP phone authenticated at the static server.  All sip entries
> have "canreinvite=no".  Two days ago this was working fine.
> 
Yes, there's been quite a lot of changes to SIP registration and authentication.

So SIP calls from a user reigstred at the static server to an extension on the dynamic server doesn't work?
Is this the setup:
	<SIP phone 1101> -> SIP CALL -> <Static server> -> SIP CALL -> <dynamic server>

´..and the dynamic server is registred with the static server?

Please add a SIP debug of the call so we can see what happens, who refuses what call.

/O