[Asterisk-Users] Net2Phone SIP

[Mark Thompson]

Just to say that I've now managed to get this going by pretending to be
an ATA 186.
change the User-Agent string in chan_sip.c from "Asterisk" to "Cisco ATA
186" and the Net2Phone Sip service works with *

Would it be possible to pick this up from sip.conf in a future release?

Regards
Mark



-----Original Message-----
From: Mark Thompson 
Sent: 02 June 2003 21:11
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] Net2Phone SIP



I've been trying to use net2phone's sip service at sip.net2phone.com
with * but keep getting 
SIP/2.0 401 Unauthorized. Do you know if this should be possible?

So far:
I can use an ata186 to connected directly to n2p through
sip.net2phone.com without any special settings. I can connect from * to
iconnecthere, but, whatever I try from * to n2p produces "SIP/2.0 401
Unauthorized" 
(Can forward the full * sip log and ata186 log if it would help)

Many thanks

Mark

--------------------------

INVITE sip:18002479297 at sip.net2phone.com SIP/2.0
Via: SIP/2.0/UDP 10.0.0.55:5060;branch=z9hG4bK7d701e63
From: "112" <sip:2763072144 at 10.0.0.55>;tag=as16e11aa8
To: <sip:18002479297 at sip.net2phone.com>
Contact: <sip:2763072144 at 10.0.0.55>
Call-ID: 7d2fd41f7e897e54712371ae0f782262 at 10.0.0.55
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Content-Type: application/sdp
Content-Length: 149

v=0
o=root 9698 9698 IN IP4 10.0.0.55
s=session
c=IN IP4 10.0.0.55
t=0 0
m=audio 19622 RTP/AVP 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
 (no NAT) to 66.33.146.12:5060
    -- Called 18002479297 at n2p
Sip read:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.0.55:5060;branch=z9hG4bK7d701e63
From: "112" <sip:2763072144 at 10.0.0.55>;tag=as16e11aa8
To: <sip:18002479297 at sip.net2phone.com>;tag=3ed78add-12013
Call-ID: 7d2fd41f7e897e54712371ae0f782262 at 10.0.0.55
CSeq: 102 INVITE
Contact: "net2phone" <sip:66.33.146.12:5060>
User-Agent: Asterisk PBX
Content-Length: 0 _______________________________________________
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