[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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