[Asterisk-Users] SIP Retransmission Patch

[Stephen Davies]

On Sat, 29 Mar 2003, Luke Howard wrote:

> 
> This seems to fix incoming calls but outgoing calls terminate 
> immediately, at least for me, with a
> 
> 481 "Call Leg/Transaction Does Not Exist"
> 
> from the SIP phone. Here's the SIP debug output (NB: IP addresses
> have been changed). It *used* to work.

Hi,

You need to post earlier on in the dialogue - something already went wrong
before that I'd say.

There is a general missing link with outgoing calls from * which is that
it doesn't handle the Record-Route.  Perhaps iconnecthere has started to
use it?

Steve