[Asterisk-Users] I didn't want to bother the list with this, but...

[Gregory Junker]

I simply cannot get X-Lite (Windows) or SJ (Linux) softphones (the only
ones I have tried) to register with Asterisk on the LAN (no NAT, no
routers). I have looked at every conceivable archived message regarding
401 Unauthorized, SJPhone, etc., and have looked at every relevant
article in the Wiki (and then some), and it looks to me like everything
should be fine....yet I cannot get these phones to register. All forward
and reverse addressing is working properly (and I even have _sip. SRV
entries set up in BIND). Asterisk is .3, the client is .236 (DHCP).

sip.conf:

[general]
port = 5060                     ; Port to bind to
bindaddr = 0.0.0.0              ; Address to bind SIP channel to
context = sip                   ; Default context for incoming calls

[8010]
type=friend
host=dynamic
dtmfmode=inband
username=gjunker
auth=md5
secret=xxxxxxxxxxxxxxxxxxxxxxxxxxx ; generated per instructions in the
Wiki




Asterisk sip debug output:

Sip read:
REGISTER sip:voip.shockwaveaudio.com SIP/2.0
Content-Length: 0
Contact: <sip:gjunker at 192.168.1.236:5060>
Call-ID: ED30CAFA-1DD1-11B2-B310-C0D342A71FF4 at 192.168.1.236
From: <sip:gjunker at voip.shockwaveaudio.com>;tag=2798956518
CSeq: 15 REGISTER
To: <sip:gjunker at voip.shockwaveaudio.com>
Via: SIP/2.0/UDP 192.168.1.236:5060
                                                                                
                                                                                
8 headers, 0 lines
Using latest request as basis request
Sending to 192.168.1.236 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.236:5060
From: <sip:gjunker at voip.shockwaveaudio.com>;tag=2798956518
To: <sip:gjunker at voip.shockwaveaudio.com>;tag=as1b4ad137
Call-ID: ED30CAFA-1DD1-11B2-B310-C0D342A71FF4 at 192.168.1.236
CSeq: 15 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:gjunker at 192.168.1.3>
Content-Length: 0
                                                                                
                                                                                
 to 192.168.1.236:5060
Apr  1 11:09:26 NOTICE[1163996080]: chan_sip.c:5643 handle_request:
Registration from '<sip:gjunker at voip.shockwaveaudio.com>' failed for
'192.168.1.236'
                                                                                
                                                                                
Sip read:
REGISTER sip:voip.shockwaveaudio.com SIP/2.0
Content-Length: 0
Contact: <sip:gjunker at 192.168.1.236:5060>
Call-ID: ED30CAFA-1DD1-11B2-B310-C0D342A71FF4 at 192.168.1.236
From: <sip:gjunker at voip.shockwaveaudio.com>;tag=2798956523
CSeq: 16 REGISTER
To: <sip:gjunker at voip.shockwaveaudio.com>
Via: SIP/2.0/UDP 192.168.1.236:5060
                                                                                
                                                                                
8 headers, 0 lines
Using latest request as basis request
Sending to 192.168.1.236 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.236:5060
From: <sip:gjunker at voip.shockwaveaudio.com>;tag=2798956523
To: <sip:gjunker at voip.shockwaveaudio.com>;tag=as10c1381f
Call-ID: ED30CAFA-1DD1-11B2-B310-C0D342A71FF4 at 192.168.1.236
CSeq: 16 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:gjunker at 192.168.1.3>
Content-Length: 0
                                                                                
                                                                                
 to 192.168.1.236:5060
Apr  1 11:09:26 NOTICE[1163996080]: chan_sip.c:5643 handle_request:
Registration from '<sip:gjunker at voip.shockwaveaudio.com>' failed for
'192.168.1.236'






An Ethereal trace shows the same thing as sip debug. 

I'm sure this has to be a configuration error on my part, but damned if
I can tell where or what...

TIA
Greg