Hello, I am trying to test-call my own asterisk server to see if I can receive SIP calls properly.
I use a softphone to call the SIP address, and because twinkle doesn't support SRV records, I go via a proxy. When the call comes in, asterisk says: handle_request_invite: Sending fake auth rejection for user "martin f. krafft" <sip:[EMAIL PROTECTED]>;tag=fipzt and SIP debugging then prints: OPTIONS sip:sip05.insphone.ch SIP/2.0 Via: SIP/2.0/UDP 84.75.148.xxx:5060;branch=z9hG4bK71785803;rport From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as05fc20f4 I am not calling as username asterisk, but I think this is the proxy substituting its name for mine. Why? Is it broken? Am I misunderstanding something? How can I fix/prevent his? Also, the IP is that of my asterisk server, the one which receives the call. It goes on: To: <sip:sip05.insphone.ch> I made the call to [EMAIL PROTECTED], not the unqualified sip05.insphone.ch (which is the proxy hostname). Contact: <sip:[EMAIL PROTECTED]> Again this is not the contact address. I see this often, that with SIP, the local part of a peer address is just changes, and I think it's similar to email header rewriting. However, header rewriting is rare and somewhat frowned upon, so why is it so commonplace with SIP? -- martin | http://madduck.net/ | http://two.sentenc.es/ "die zeit für kleine politik ist vorbei. schon das nächste jahrhundert bringt den kampf um die erdherrschaft." - friedrich nietzsche spamtraps: [EMAIL PROTECTED]
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