Yeah, if only it were that simple, as we migrate to VoIP which numbers go where will constantly change. That was the nice part about routing through the main servers, they "know" which numbers are currently connected everything else goes out to the gateways with the fallback rules.
I think that best bet is to have aliases for the DIDs that are yet to be assigned in SIPX and just be careful that when we assign a new DID to SIPX it has either an alias or a user. On Tue, Sep 28, 2010 at 9:26 AM, Michael Picher <[email protected]> wrote: > Well, you would end up with a couple different dial plans right? > > Route outbound calls to the audiocodes gateways... route calls to other > sipXecs IP connected sites with unmanaged gateways to those systems... > > clear as mud? > > On Tue, Sep 28, 2010 at 11:18 AM, Kyle Haefner > <[email protected]<mailto:[email protected]>> wrote: > Mike, > > > True, I thought about that, do you know if Audiocodes gateways also > act as sip proxies for IP to IP calls? If not, the down side I think > is that calls destined for IP phones on the main campus from the > remote campus will take up two trunks in the gateway and won't have > that nice G722 codec. > > > Kyle > > > On Tue, Sep 28, 2010 at 9:00 AM, Michael Picher > <[email protected]<mailto:[email protected]>> wrote: >> I'm not sure I fully understand the question but let me answer what I think >> you are asking... >> >> From a gateway perspective you could just individually add gatways to each >> of the sipXecs systems... you don't necessarily need to go back through a >> single system just because it has the gateway 'on it'. >> >> But the remote system has no knowledge of another system's permissions / >> groups. >> >> Mike >> >> On Mon, Sep 27, 2010 at 3:44 PM, Kyle Haefner >> <[email protected]<mailto:[email protected]><mailto:[email protected]<mailto:[email protected]>>> >> wrote: >> Another wrench! >> >> I enabled a default permission on my 5 digit dial rule. It worked as >> Josh said it would, now sipx sends a 407 back to the gateways then the >> Nortel switch drops the call. What I found out though is remote >> offices could not call five digit numbers out through the gateway (I >> have sit-to-site dialing between the main office and remote offices, >> that permission blocks the calls from remote offices that are bound >> for non-sipx 5 digit extensions) is there a way to remote sites >> permissions? >> >> Kyle >> >> On Sun, Sep 26, 2010 at 4:10 AM, Michael Picher >> <[email protected]<mailto:[email protected]><mailto:[email protected]<mailto:[email protected]>>> >> wrote: >>> Jason, >>> >>> What you could do is create a permission labeled 'Call3000' (or whatever) >>> and then make a dial plan entry that looks for 3xxx and requires permission >>> Call3000. Then users that don't have that won't be able to call those >>> extensions. >>> >>> Not exactly sure what this would do to inbound calling however ;-) You >>> could route all your inbound calls to aliases... >>> >>> Mike >>> >>> On Fri, Sep 24, 2010 at 8:06 PM, >>> [email protected]<mailto:[email protected]><mailto:[email protected]<mailto:[email protected]>><mailto:[email protected]<mailto:[email protected]><mailto:[email protected]<mailto:[email protected]>>> >>> >>> <[email protected]<mailto:[email protected]><mailto:[email protected]<mailto:[email protected]>><mailto:[email protected]<mailto:[email protected]><mailto:[email protected]<mailto:[email protected]>>>> >>> wrote: >>> His Nikolay >>> >>> I am trying to use yr method to block internal extension dialing, that is >>> 2xxx cannot dial 3xxx. Same thing, I create a dial plan that translate 3xxx >>> to something else when 2xxx extension press 3xxx number. Then send the >>> translated number to the unmanaged gateway. >>> >>> I have try, but seem that the dial plan does not process my number >>> >>> Is it possible to do it this way? >>> >>> Thanks >>> Jason >>> >>> >>> >>> >>> >>> >>> >>> On Sep 24, 2010, at 16:52, "Nikolay Kondratyev" >>> <[email protected]<mailto:[email protected]><mailto:[email protected]<mailto:[email protected]>><mailto:[email protected]<mailto:[email protected]><mailto:[email protected]<mailto:[email protected]>>>> >>> wrote: >>> >>>> The following worked for me: >>>> >>>> I have 4 digit extensions 3xxx and 2xxx. >>>> I want to route all 2xxx unknown numbers to a separate AA. >>>> >>>> I created new AA and dial rule for it, so that this new AA has number 3333. >>>> Then I created the folowing custom rule: >>>> Match prefix 2 and 3 digits, dial 3333, no gateway. >>>> The rule is the very last in the dial plan. >>>> This rule intercepts all calls to 2xxx, so that registered extensions are >>>> not available any more. >>>> >>>> But... Some time ago (a year or more) somebody in the list offered the >>>> following workaround (and it works for me now): >>>> I created an unmanaged gateway, named, say, Myself, with sipx own ip >>>> address. >>>> I modified my rule, so that 2xxx numbers are translated to 3333 and routed >>>> to gateway Myself. >>>> (In this way the rule goes into fallbackrules.xml instead of >>>> mappingruels.xml. Mappingrules is processed first.) >>>> >>>> Now the call is routed to the phone (or vm) if a user exists, and to new AA >>>> when not. >>>> >>>> I'm not sure if any side effect may appear... When call tranferring or >>>> pickup or something else... >>>> Additional tests need to be performed... >>>> >>>> Rgds, >>>> Nikolay. >>>> >>>> >>>> >>>>> -----Original Message----- >>>>> From: >>>>> [email protected]<mailto:[email protected]><mailto:[email protected]<mailto:[email protected]>><mailto:[email protected]<mailto:[email protected]><mailto:[email protected]<mailto:[email protected]>>> >>>>> [mailto:[email protected]<mailto:[email protected]><mailto:[email protected]<mailto:[email protected]>><mailto:[email protected]<mailto:[email protected]><mailto:[email protected]<mailto:[email protected]>>>] >>>>> On Behalf Of >>>>> Kyle Haefner >>>>> Sent: Thursday, September 23, 2010 11:44 PM >>>>> To: sipx-users >>>>> Subject: [sipx-users] Non-route-able numbers >>>>> >>>>> Hi, >>>>> >>>>> Is there a way to setup the dial plan such that any extension >>>>> of a certain length is answered even if there is no >>>>> corresponding user/attendant/conference? >>>>> >>>>> The problem is I have a block of numbers that have been >>>>> ported from our DMS100 switch, not all these numbers are >>>>> assigned to users yet i.e. I have not created users with >>>>> those extensions yet. What I would like is to have these >>>>> numbers sent to an auto-attendant that simply plays a message >>>>> saying that the call cannot be completed as dialed. I know I >>>>> could do this with a phantom user and aliases, but that >>>>> quickly becomes unmanageable. How do I set-up a catch-all >>>>> for inbound calls? >>>>> >>>>> Thanks! >>>>> >>>>> Kyle >>>>> _______________________________________________ >>>>> sipx-users mailing list >>>>> [email protected]<mailto:[email protected]><mailto:[email protected]<mailto:[email protected]>><mailto:[email protected]<mailto:[email protected]><mailto:[email protected]<mailto:[email protected]>>> >>>>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>>> >>>> _______________________________________________ >>>> sipx-users mailing list >>>> [email protected]<mailto:[email protected]><mailto:[email protected]<mailto:[email protected]>><mailto:[email protected]<mailto:[email protected]><mailto:[email protected]<mailto:[email protected]>>> >>>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>> _______________________________________________ >>> sipx-users mailing list >>> [email protected]<mailto:[email protected]><mailto:[email protected]<mailto:[email protected]>><mailto:[email protected]<mailto:[email protected]><mailto:[email protected]<mailto:[email protected]>>> >>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>> >>> >>> >>> -- >>> There are 10 kinds of people in this world, those who understand binary and >>> those who don't. >>> >>> [email protected]<mailto:[email protected]><mailto:[email protected]<mailto:[email protected]>><mailto:[email protected]<mailto:[email protected]><mailto:[email protected]<mailto:[email protected]>>> >>> blog: http://www.sipxecs.info >>> call: >>> sip:[email protected]<mailto:sip%[email protected]><mailto:sip%[email protected]<mailto:sip%[email protected]>><mailto:sip%[email protected]<mailto:sip%[email protected]><mailto:sip%[email protected]<mailto:sip%[email protected]>>> >>> >> _______________________________________________ >> sipx-users mailing list >> [email protected]<mailto:[email protected]><mailto:[email protected]<mailto:[email protected]>> >> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >> >> >> >> -- >> There are 10 kinds of people in this world, those who understand binary and >> those who don't. >> >> [email protected]<mailto:[email protected]><mailto:[email protected]<mailto:[email protected]>> >> blog: http://www.sipxecs.info >> call: >> sip:[email protected]<mailto:sip%[email protected]><mailto:sip%[email protected]<mailto:sip%[email protected]>> >> > _______________________________________________ > sipx-users mailing list > [email protected]<mailto:[email protected]> > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > > > > -- > There are 10 kinds of people in this world, those who understand binary and > those who don't. > > [email protected]<mailto:[email protected]> > blog: http://www.sipxecs.info > call: sip:[email protected]<mailto:sip%[email protected]> > _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/
