Gregory Junker
2004-Apr-01 09:20 UTC
[Asterisk-Users] I didn't want to bother the list with this, but...
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@192.168.1.236:5060>
Call-ID: ED30CAFA-1DD1-11B2-B310-C0D342A71FF4@192.168.1.236
From: <sip:gjunker@voip.shockwaveaudio.com>;tag=2798956518
CSeq: 15 REGISTER
To: <sip:gjunker@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@voip.shockwaveaudio.com>;tag=2798956518
To: <sip:gjunker@voip.shockwaveaudio.com>;tag=as1b4ad137
Call-ID: ED30CAFA-1DD1-11B2-B310-C0D342A71FF4@192.168.1.236
CSeq: 15 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:gjunker@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@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@192.168.1.236:5060>
Call-ID: ED30CAFA-1DD1-11B2-B310-C0D342A71FF4@192.168.1.236
From: <sip:gjunker@voip.shockwaveaudio.com>;tag=2798956523
CSeq: 16 REGISTER
To: <sip:gjunker@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@voip.shockwaveaudio.com>;tag=2798956523
To: <sip:gjunker@voip.shockwaveaudio.com>;tag=as10c1381f
Call-ID: ED30CAFA-1DD1-11B2-B310-C0D342A71FF4@192.168.1.236
CSeq: 16 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:gjunker@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@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
Rainer Jochem
2004-Apr-01 09:26 UTC
[Asterisk-Users] I didn't want to bother the list with this, but...
Hi,> [8010] > type=friend > host=dynamic > dtmfmode=inband > username=gjunker > auth=md5 > secret=xxxxxxxxxxxxxxxxxxxxxxxxxxx ; generated per instructions in the > Wiki >This auth-style is news to me. Where did you find it in the wiki? I only know the usage of md5secret=xxxxxxxxxxxxxxxxxxxxxxxxx without any auth=md5 stuff which is working here. Greetings, Rainer -- http://graphics.cs.uni-sb.de/VoIP/ -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20040401/b55e4e7f/attachment.pgp