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John Todd

2003-Jun-02 13:00 UTC

[Asterisk-Users] Net2Phone SIP

Hints to debugging this issue:

Just for your own use, I'd suggest using the "tethereal -V port
5060"
incantation of ethereal to see a more detailed packet-level debug 
"from the wire".

Look at the REGISTER packets (if N2P indeed does use REGISTERs) since 
that may be less complex, and may give you the same errors.

Compare a packet trace from your ATA-186 with that of Asterisk and 
see where the differences are.

JT

>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@sip.net2phone.com SIP/2.0
>Via: SIP/2.0/UDP 10.0.0.55:5060;branch=z9hG4bK7d701e63
>From: "112" <sip:2763072144@10.0.0.55>;tag=as16e11aa8
>To: <sip:18002479297@sip.net2phone.com>
>Contact: <sip:2763072144@10.0.0.55>
>Call-ID: 7d2fd41f7e897e54712371ae0f782262@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@n2p
>Sip read:
>SIP/2.0 401 Unauthorized
>Via: SIP/2.0/UDP 10.0.0.55:5060;branch=z9hG4bK7d701e63
>From: "112" <sip:2763072144@10.0.0.55>;tag=as16e11aa8
>To: <sip:18002479297@sip.net2phone.com>;tag=3ed78add-12013
>Call-ID: 7d2fd41f7e897e54712371ae0f782262@10.0.0.55
>CSeq: 102 INVITE
>Contact: "net2phone" <sip:66.33.146.12:5060>
>User-Agent: Asterisk PBX
>Content-Length: 0
>_______________________________________________
>Asterisk-Users mailing list
>Asterisk-Users@lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-users

Mark Thompson

2003-Jun-02 13:11 UTC

head link

[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@sip.net2phone.com SIP/2.0
Via: SIP/2.0/UDP 10.0.0.55:5060;branch=z9hG4bK7d701e63
From: "112" <sip:2763072144@10.0.0.55>;tag=as16e11aa8
To: <sip:18002479297@sip.net2phone.com>
Contact: <sip:2763072144@10.0.0.55>
Call-ID: 7d2fd41f7e897e54712371ae0f782262@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@n2p
Sip read:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.0.55:5060;branch=z9hG4bK7d701e63
From: "112" <sip:2763072144@10.0.0.55>;tag=as16e11aa8
To: <sip:18002479297@sip.net2phone.com>;tag=3ed78add-12013
Call-ID: 7d2fd41f7e897e54712371ae0f782262@10.0.0.55
CSeq: 102 INVITE
Contact: "net2phone" <sip:66.33.146.12:5060>
User-Agent: Asterisk PBX
Content-Length: 0

Stephen Davies

2003-Jun-02 14:53 UTC

head link

[Asterisk-Users] Net2Phone SIP

On Mon, 2 Jun 2003, Mark Thompson wrote:
> 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)
It is normal for n2p to send back an Unauthorised if you send an
unauthenticated INVITE.  Asterisk should re-send the INVITE but this
time authenticated.  For that to work, the entry in sip.conf needs a
username and secret.

Steve

Brian Capouch

2003-Jun-02 15:10 UTC

head link

[Asterisk-Users] Net2Phone SIP

Stephen Davies wrote:>> On Mon, 2 Jun 2003, Mark Thompson wrote:
> 
> 
>>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)
> 
> 
> It is normal for n2p to send back an Unauthorised if you send an
> unauthenticated INVITE.  Asterisk should re-send the INVITE but this
> time authenticated.  For that to work, the entry in sip.conf needs a
> username and secret.
> 
Do you (or does anyone) have an example sip.conf entry for net2phone to 
use as a template?

I played around with that extensively a month or two ago without 
success.  Perhaps now things are going to work.

Thx.

B.

Mark Thompson

2003-Jul-09 06:41 UTC

head link

[Asterisk-Users] Net2Phone SIP

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@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@sip.net2phone.com SIP/2.0
Via: SIP/2.0/UDP 10.0.0.55:5060;branch=z9hG4bK7d701e63
From: "112" <sip:2763072144@10.0.0.55>;tag=as16e11aa8
To: <sip:18002479297@sip.net2phone.com>
Contact: <sip:2763072144@10.0.0.55>
Call-ID: 7d2fd41f7e897e54712371ae0f782262@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@n2p
Sip read:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.0.55:5060;branch=z9hG4bK7d701e63
From: "112" <sip:2763072144@10.0.0.55>;tag=as16e11aa8
To: <sip:18002479297@sip.net2phone.com>;tag=3ed78add-12013
Call-ID: 7d2fd41f7e897e54712371ae0f782262@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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