have a Grandstream and SJPhone SIP phones going to asterisk.<br><br>with SJPhone (on Linux) getting. any ideas??<br><br>SIP/2.0 401 Unauthorized<br>Via: SIP/2.0/UDP 192.168.2.100;branch=z9hG4bKc0a802640000001045c3c2c52331d49200000678;received=24.10.123.39;rport=60754<br>From: <sip:user2@ca.dummy.net>;tag=22261807771886928353<br>To: <sip:ca.dummy.net>;tag=as45966c6b<br>Call-ID: 1E42EB8C-1DD2-11B2-BDFB-B022A8C6AB96@192.168.2.100<br>CSeq: 41 OPTIONS<br>User-Agent: Asterisk PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Supported: replaces<br>Accept: application/sdp<br>Content-Length: 0<br><br><br><p> 
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