[Asterisk-Users] Re: Granstream BT100 - only partial success

[Stephen R. Besch]

George Burt wrote:
<snip>
> [grandstream1]
> host=10.0.0.26           ; we have a static but private IP address
> canreinvite=yes        ; allow RTP voice traffic to bypass Asterisk
<snip>
> IP Address:
> statically configured as:
>         IP Address:    10.0.0.26
>         Subnet Mask:   255.255.255.0
>         Default Router:   10.0.0.1
> SIP Registration:    Yes

Comments:

1) You can't ask asterisk to register your phone if you have a fixed IP 
address specified as host= in sip.conf. Either the phone sends the 
address (i.e., host=dynamic), or you enter it as an IP address. It's OK 
to be fixed at the phone and dynamic in asterisk, but that isn't 
rational - just adds net traffic. Turn off the sip registration option 
on the phone.

2) Unless I am mistaken, you are not going to be able to use re-invites 
without NAT. It will work on your calls to analog phones handled by 
Asterisk and to other IP phones on the local network. However, as soon 
as you connect to an outbound/inbound service, the reinvite will fail 
and you will lose your media stream.

3) Don't know if it will make a difference, but I always set the router 
field to 0.0.0.0.  There is no such thing as a valid router IP on a 
private network - they are not routable by design. I had quite an 
argument with Grandstream about this when I first purchased the phones. 
As a result, the firmware was modified to accept a null router entry for 
use with private IP ranges.

Sincerely,

Stephen R. Besch