[Asterisk-Users] Struggling with grandstream sip to asterisk

[Stephen R. Besch]

>*CLI> NOTICE[81926]: File chan_sip.c, Line 5215 (handle_request): Registration 
>from '<sip:206 at 192.168.1.1>' failed for '192.168.1.70'
>

I've seen this in only two cases:
 
     1) when the SIP user ID, the Authenticate ID (on the GS) and the 
extension name in sip.conf are not all the same, in your case, 206. The 
user name in sip.conf doesn't matter, and in fact doesent seem to do 
anything at all with the GS.

     2) You are using a secret and the authenticate Password does not 
match the secret.  This does not appear to be relevant to your situation.

You might also check to be sure that the registration option in the GS 
is turned off, since you are hard coding the IP address.  There is a bug 
in the current GS firmware (supposed to be fixed soon) that sometimes 
messes up the registration renewal.

>    -- Username not entered
>    -- Executing Hangup("SIP/206-7ecb", "") in new stack
>
You can't use INFO with the GS. * and GS interpret the INFO standard 
differently.  As a result, the GS does multiple digit transmission.  Use 
either inband or the RFC2833 option.  INFO will not work no matter what 
you do.

Also, if you are hard coding your phone's IP addresses, then the permit 
option has no meaning.

>[206]
>username=206
>context=extensions
>qualify=yes
>incominglimit=1
>type=friend
>insecure=yes
>host=192.168.1.70
>permit=192.168.1.0/255.255.255.0
>dtmfmode=info
>canreinvite=no
>reinvite=no
>callgroup=1
>pickupgroup=1
>disallow=all
>allow=alaw
>allow=ulaw
>
>  
>