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:[EMAIL PROTECTED]>;tag=22261807771886928353
To: <sip:ca.dummy.net>;tag=as45966c6b
Call-ID: [EMAIL PROTECTED]
CSeq: 41 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Accept: application/sdp
Content-Length: 0
---------------------------------
Check out the all-new Yahoo! Mail beta - Fire up a more powerful email and get
things done faster.
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
