[Asterisk-Users] missing name part in to field of SIP header
Trond Andersen
trond.andersen at tandberg.net
Fri Nov 11 08:48:42 MST 2005
Hi everyone.
I have a small problem with my Asterisk setup?!?
I am trying to connect to another endpoint through my asterisk server.
The packet going in is just like i want it, but the packet going out of
asterisk at to the other endpoint is missing a part in the header?
it looks like this:
To: <sip:x.x.x.x>;tag=.....
where is the phone2@ part in my SIP URI??
I want it to look like:
To: <sip:phone2 at x.x.x.x>;tag=.....
I have my own very simple dialplan using:
exten => s,2,Dial(${ARG2},20,Cf) where ARG2 is SIP/phone2
The reason i need this is to have several conferences going on at the
same time at the same ip-address.
Any ideas ?
Trond
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