Hi all!
I'm trying to have asterisk route all outgoing calls out via my VOIP provider.
exten => _NXXXXXXX,1,Dial,SIP/[EMAIL PROTECTED] 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:[EMAIL PROTECTED] IP]>;tag=as0aca53fa To: <sip:[dialled [EMAIL PROTECTED] IP]> Call-ID: [EMAIL PROTECTED] 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:[EMAIL PROTECTED] IP]>;tag=as0aca53fa To: <sip:[EMAIL PROTECTED] IP]>;tag=87f2a0d5-13c4-4146d5ea-1a4b30eb-3af4 Call-ID: [EMAIL PROTECTED] 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
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