[asterisk-users] Polycom WEB UI configuration - What needs to be put in for basic SIP registration?
Bruce B
bruceb444 at gmail.com
Fri Nov 5 14:30:50 CDT 2010
Hi Everyone,
Configuring a Polycom conference bridge IP 5000 to connect to Asterisk. For
some reason I don't see any SIP packets coming in to Asterisk at all. I
don't want to use XML or ftp etc for now and just use the Web Interface to
get it going with basic features. But the Web UI is a bit confusing with SIP
and Line tabs.
I have put this on the web interface:
SIP > Outbound Proxy:
Address = 192.168.0.2
Port = 5060
Server 1:
Address = 192.168.0.2
Port = 5060
Transport = DNSnaptr
Expires = 300
Register = 1
Line:
Display Name = 100
Address = 192.168.0.2
Authentication User ID = 100
Authentication Password = *************
Label = 100
Server 1:
Address = 192.168.0.2
Port = 5060
Transport = DNSnaptr
Expires = 300
Register = 1
I don't see any registration attempts but Snom phones on the same network
can register to Asterisk just fine.
Thanks
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101105/6807178c/attachment.htm
More information about the asterisk-users
mailing list