[asterisk-users] problems with SJPhone (I feel stupid about this)

[chester c young]

have a Grandstream and SJPhone SIP phones going to asterisk.

with SJPhone (on Linux) getting.  any ideas??

SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.2.100;branch=z9hG4bKc0a802640000001045c3c2c52331d49200000678;received=24.10.123.39;rport=60754
From: <sip:user2 at ca.dummy.net>;tag=22261807771886928353
To: <sip:ca.dummy.net>;tag=as45966c6b
Call-ID: 1E42EB8C-1DD2-11B2-BDFB-B022A8C6AB96 at 192.168.2.100
CSeq: 41 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Accept: application/sdp
Content-Length: 0



 
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