[Asterisk-Users] Wrong ID going out...

[Evert Meulie]

Hi all!

I'm trying to have asterisk route all outgoing calls out via my VOIP 
provider.
exten => _NXXXXXXX,1,Dial,SIP/BYEXTENSION at VOIP seems to have them to in 
the direct direction. However, debug shows that my asterisk doesn't 
identify itself correctly:


Sip read:
SIP/2.0 100 Trying
From: "Evert"<sip:asterisk@[my IP]>;tag=as0aca53fa
To: <sip:[dialled number]@[voip IP]>
Call-ID: 0013b54e26506f7f4133cc0f59f1e561@[my IP]
CSeq: 102 INVITE
Via:SIP/2.0/UDP [my IP]:5060;branch=z9hG4bK00215868
Content-Length:0
 
 
7 headers, 0 lines
 
 
Sip read:
SIP/2.0 403 Forbidden (From header is not a Trust host or gateway)
From: "Evert"<sip:asterisk@[my IP]>;tag=as0aca53fa
To: <sip:69101701@[voip IP]>;tag=87f2a0d5-13c4-4146d5ea-1a4b30eb-3af4
Call-ID: 0013b54e26506f7f4133cc0f59f1e561@[my IP]
CSeq: 102 INVITE
Via:SIP/2.0/UDP [my IP]:5060;branch=z9hG4bK00215868
Content-Length:0

( [my IP] is my external IP, [voip IP] is the IP of the SIP server of my 
VoIP provider. [dialled number] is the number I dialled)


I don't see any sign here of the username/password being passed to my 
provider. is that ok?
IMHO I think it should identify me as [username]/[password], instead of 
'asterisk' to my VoIP provider.


What am I doing wrong...?


Regards,
   Evert