IP phones using Asterisk and a VoIP trunk


#1

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.

Regards,
Adrian


#2

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


#3

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