Possibly if I could take a look at their GUI and custom contexts. That could be quite a bit of work....
On Tue, Feb 10, 2009 at 10:26 AM, Mike Hammett <[email protected]> wrote: > Do you know enough about Trixbox to tell me where they need to fix their > misconfiguration, or is it a Trixbox design flaw? > > > ----- > Mike Hammett > Intelligent Computing Solutions > http://www.ics-il.com > > > > -------------------------------------------------- > From: "Steve Totaro" <[email protected]> > Sent: Tuesday, February 10, 2009 8:58 AM > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <[email protected]> > Subject: Re: [asterisk-users] Asterisk - Trixbox > >> Yes, they should fix this on their side, otherwise DID routing will >> not work. If you don't need it, you just need to create a DID entry >> for any/all or any/any, I cannot remember which it is right now, but >> it should be apparent when you look at it. >> >> The s extension is only used when no DID or extension is received. >> >> Thanks, >> Steve Totaro >> >> On Tue, Feb 10, 2009 at 9:49 AM, Mike Hammett <[email protected]> >> wrote: >>> They must have changed something after I complained because it no longer >>> references the incorrect phone number. I did disable >>> >>> However, it still wants to send everything to the s extension. >>> Everything I >>> have worked with before has sent calls the the DID's extension (a call to >>> 8887776666 goes to exten => 8887776666,1,blah). Is this something they >>> can >>> change in Trixbox? >>> >>> http://pastebin.com/fa8b4f4e I highlighted the lines that contain the s >>> extension. >>> >>> >>> ----- >>> Mike Hammett >>> Intelligent Computing Solutions >>> http://www.ics-il.com >>> >>> >>> >>> -------------------------------------------------- >>> From: "Steve Totaro" <[email protected]> >>> Sent: Tuesday, February 10, 2009 7:29 AM >>> To: "Asterisk Users Mailing List - Non-Commercial Discussion" >>> <[email protected]> >>> Subject: Re: [asterisk-users] Asterisk - Trixbox >>> >>>> Mike, >>>> >>>> Please explain the problem more clearly and post a pastebin that shows >>>> the problem and only the problem, not a huge SIP dump. >>>> >>>> If you could point out the line numbers where you suspect an issue. >>>> >>>> Thanks, >>>> Steve >>>> >>>> On Mon, Feb 9, 2009 at 10:56 PM, Mike Hammett >>>> <[email protected]> >>>> wrote: >>>>> Can anyone help me determine where the problem lies and how to fix it? >>>>> >>>>> >>>>> ----- >>>>> Mike Hammett >>>>> Intelligent Computing Solutions >>>>> http://www.ics-il.com >>>>> >> >> _______________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
