[asterisk-users] SIP proxy screwing up peer addresses.

[martin f krafft]

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:xxxxxxxxxxxxxxxx at sip05.insphone.ch>;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:asterisk at 84.75.148.xxx>;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 madduck at madduck.net, not the unqualified
sip05.insphone.ch (which is the proxy hostname).

  Contact: <sip:asterisk at 84.75.148.xxx>

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: madduck.bogus at madduck.net
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