Hello all..
In an effort to find a reasonably priced analog VOIP-TA here in Japan, I
came accross one
made by NTT for just over $100 (most are over $400). I have been able to
call out fine, but incoming
always gives me a "404 Not Found".
I don't think the different IPs are related (this is just because
Asterisk is in the middle)
Is there anyway to get the same Via and Record-Route headers?
From provider:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Call-ID: [EMAIL PROTECTED]
Content-Length: 143
Content-Type: application/sdp
From: <sip:[EMAIL PROTECTED]>;tag=3353277539
Cseq: 1 INVITE
To: <sip:[EMAIL PROTECTED]>
Via: SIP/2.0/UDP
218.40.158.36:5060;branch=z9hG4bKf9633b2366eec368695129eb7d378f5e
Via: SIP/2.0/UDP
218.40.158.55:5060;branch=z9hG4bK920207c5494c9f6e9562fbe2304f98f0
Contact: <sip:218.40.158.55:5060;transport=udp>
Max-Forwards: 69
Allow: INVITE,ACK,PRACK,BYE,CANCEL,REGISTER,UPDATE
Date: Thu, 06 Apr 2006 01:58:59 GMT
Supported: timer , 100rel
Min-SE: 300
Session-Expires: 300;refresher=uac
Record-Route: <sip:218.40.158.36:5060;maddr=218.40.158.36;lr>
Compare with Asterisk:
Reliably Transmitting (NAT) to 221.43.146.48:5061:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 203.83.240.142:5060;branch=z9hG4bK541c8170
From: "090xxxxxxxx" <sip:[EMAIL PROTECTED]>;tag=as46f05419
To: <sip:[EMAIL PROTECTED]>
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 06 Apr 2006 03:24:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 312
<-- SIP read from 221.43.146.48:5061:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 203.83.240.142:5060;branch=z9hG4bK041c678b
Via: SIP/2.0/UDP 203.83.240.142:5060;branch=z9hG4bK041c678b
From: "090xxxxxxxx" <sip:[EMAIL PROTECTED]>;tag=as20fd027a
To: <sip:[EMAIL PROTECTED]>;tag=2150617465
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Content-Length: 0
Any help would be greatly appreciated...and will benifit the Asterisk
User's of Japan!
Jason
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