IP phones using Asterisk and a VoIP trunk

We have an asterisk box running, with a few Netcomm V85 VoIP phones, and a SIP trunk to a VoIP service provider.

We have a dedicated 512/512 line for our Asterisk box and a 1500/256 line for our normal network. However the IP phones have DCHP enabled and are aquiring an IP from our DHCP server that sets the Default Gateway as the 1500/256 line.

My question is really regarding the routing of SIP with an asterisk box. I’m assuming I will need to set the IP’s of all the VoIP phones statically, so I can assign the correct default gateway (the 512/512 line), because at present when we make a call it would be routing out via the 1500/256 line. Or does the call route via the Asterisk box and use it’s default gateway?

Any light that can be shed would be greatly appreciated.


You don’t need to set the phones’ addresses statically. Just set canreinvite=no in sip.conf and all calls will go through Asterisk.

Thanks heaps for that.

Having a look into it now, and studying the bible (Asterisk The Future of Telephony)

Thanks for the pointer in the right direction