To my knowledge, this is a limitation in asterisk. It only considers the IP address of the remote, all other values are ignored for type=peer. You have to sort it out in the dialplan.
On Fri, Sep 7, 2012 at 11:27 AM, Dean Yorke <[email protected]> wrote: > Hi All, > > Just wondering if someone can help me clarify and understand something > > I have an Voip Provider with whom I have 4 DID's > > My asterisk box is behind a router with no ports open. > > I can send and receive calls fine. > > The issue is that regardless of the DID registered, all calls from the same > Provider seem to come in on the first trunk. That is, in the CLI it shows > the DID@from-pstn and then it drops to SIP/DID > > Any idea's on what is happening? What can I do to try and correct this. > > Thanks > > > --------------------------------------------------------------------- > To unsubscribe, e-mail: [email protected] > For additional commands, e-mail: [email protected] > --------------------------------------------------------------------- To unsubscribe, e-mail: [email protected] For additional commands, e-mail: [email protected]
