George Burt wrote: <snip>
<snip>[grandstream1] host=10.0.0.26 ; we have a static but private IP address canreinvite=yes ; allow RTP voice traffic to bypass Asterisk
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
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