[Asterisk-Users] SIP register/auth with Grandstream BudgeTone-100

[Larry Keyes]

Key, 

I've been playing with the Grandstreams for some weeks; one good way to see
the registration messages it to monitor the network with Ethereal. (packet
sniffer).  You'll see the SIP messages coming and going, with complete
decoding. This works pretty much as predicted when using VOCAL. (another SIP
server) 

That said, I recently asked about registration with Asterisk from the
Grandstreams, and received a reply that if the phones have a fixed IP
address they do not "register" as they are already accounted for in the
sip.conf by the user name/number assigned to the phone, and indeed the
phones work fine (at least inside the firewall).  
Other soureces have described Asterisk as having "limited or not fully
implemented SIP support"... but I have never been able to determine to what
extent SIP support is lacking. 

I was able to get the Grandstreams running only recently with Asterisk... My
sip.conf for the Grandstreams has several differences from yours: For a
grandstream phone with an assigned number of 1000 and a fixed IP address of
192.168.0.160 I have the following: 


[1000]
type=friend
username=1000
host=dynamic
reinvite=no
canreinvite=no
qualify=300
callerid="Larry's Desk" <1000>
mailbox=1000
nat=no
dtmfmode=info
disallow=all
allow=ulaw
allow=mlaw

See the message I sent to Tom Scott on 1/21 for a little more information.  

I would very much like to put together a set of instructions for the
Grandstream phones+asterisk that could expand on the scattered information
found on the wiki/web/draft manual and this list. 

And let us know what you find!

     
-- L