[cisco-voip] incoming dial peers

[Dane Newman]

So please correct me if I am wrong If I have the dial peer below  If it is
matching the dial peer 1000 it shoul traslate the number and send it to
190.. When I am calling in sadly I am just getting a fast busy signal.

dial-peer voice 1000 voip
 description incoming Call
 translation-profile incoming aa
 preference 1
 session protocol sipv2
 session target sip-server
 incoming called-number 16784663444
 dtmf-relay rtp-nte
 no vad
!

dial-peer voice 100 voip
 description 1-5xx extension to PUBLISHER
 preference 1
 session target ipv4:10.1.80.6
 incoming called-number [1-5]..
 voice-class h323 50
 dtmf-relay h245-alphanumeric
 codec g711ulaw
 no vad

On Fri, Oct 16, 2009 at 5:39 PM, Nick Matthews <matthnick at gmail.com> wrote:

> Translation patterns work on SIP calls, yes.
>
> to match any number -  /.*/   /190/
>
> On Fri, Oct 16, 2009 at 3:14 PM, Dane Newman <dane.newman at gmail.com>
> wrote:
> > What is a translation pattern to match any number?
> >
> > On Fri, Oct 16, 2009 at 2:13 AM, Dane Newman <dane.newman at gmail.com>
> wrote:
> >>
> >> Hello Nick
> >>
> >> Thanks for your help
> >>
> >> I created the the following config
> >>
> >> voice translation-rule 1
> >> rule 1 /*^/ /190/
> >> voice translation-profile aa
> >> translate called 1
> >>
> >>
> >> dial-peer voice 1000 pots
> >> description incoming Call
> >> preference 1
> >> incoming called-number 6784663444
> >> progress_ind setup enable 3
> >> progress_ind progress enable 8
> >> dtmf-relay rtp-nte
> >> no vad
> >> translation-profile incoming aa
> >>
> >>
> >> This should match the incoming called number 6784663444 and translate it
> >> to my auto attendant hunt pilot of 190 correct?
> >>
> >> I had read below that Digit translation rules are not supported for
> >> inbound session initiation protocol (SIP) calls.
> >>
> >>
> >>
> http://www.cisco.com/en/US/docs/ios/12_2/voice/configuration/guide/vvfpeers.html
> >>
> >> I guess this has been updated in the latest code to work right?
> >>
> >>
> >> On Thu, Oct 15, 2009 at 6:07 PM, Nick Matthews <matthnick at gmail.com>
> >> wrote:
> >>>
> >>> This may help:
> >>>
> >>>
> http://www.cisco.com/en/US/partner/tech/tk652/tk90/technologies_tech_note09186a008010fed1.shtml
> >>>
> >>> On your first example you don't have a codec set.  It will be g729r8
> >>> by default.  As well, no DTMF-relay so you will not have that.
> >>>
> >>> You probably want to add something like this:
> >>>
> >>> codec g711ulaw
> >>> no vad
> >>> dtmf-relay rtp-nte
> >>>
> >>>
> >>> To 'plar' the number you would use a translation pattern.
> >>>
> >>>
> >>> Regarding your other problem - there isn't enough information.  Your
> >>> dial peer looks like a standard dial peer, but without a better idea
> >>> of the call flow it's hard to tell.  There's nothing specific in that
> >>> dial peer that would make it an incoming dial peer, but rather an
> >>> outgoing dial peer.
> >>>
> >>> -nick
> >>>
> >>> On Thu, Oct 15, 2009 at 3:46 PM, Dane Newman <dane.newman at gmail.com>
> >>> wrote:
> >>> > Hello All,
> >>> >
> >>> > I have a question about how to exactly configure incoming dial
> >>> > peers.  I
> >>> > have a sip trunk to an ITSP for my incoming telco connection.  The
> >>> > number is
> >>> > 6784663444.  There are four channels in the trunk but only one
> number.
> >>> > Am I
> >>> > matching the called number properly?  How can I plar it to the unity
> >>> > connection auto attendant of hunt pilot 190?
> >>> >
> >>> > dial-peer voice 1000 voip
> >>> > description incoming Call
> >>> > preference 1
> >>> > incoming called-number 6784663444
> >>> > progress_ind setup enable 3
> >>> > progress_ind progress enable 8
> >>> > !
> >>> >
> >>> > dial-peer voice 100 voip
> >>> > description 190 AA
> >>> > preference 1
> >>> > destination-pattern 190
> >>> > progress_ind setup enable 3
> >>> > progress_ind progress enable 8
> >>> > voice-class h323 50
> >>> > session target ipv4:10.1.80.6
> >>> > dtmf-relay h245-alphanumeric
> >>> > codec g711ulaw
> >>> > no vad
> >>> >
> >>> >
> >>> >
> >>> > Also....
> >>> >
> >>> >
> >>> > If I have another gateway across a WAN sending calls to extentions
> >>> > 100-500
> >>> > Will these incoming dial peers match the incoming calls and route
> them
> >>> > to my
> >>> > cucm 7 properly?
> >>> >
> >>> > dial-peer voice 100 voip
> >>> > description 1-5xx extension to PUBLISHER
> >>> > preference 1
> >>> > incoming called-number [1-5]..
> >>> > progress_ind setup enable 3
> >>> > progress_ind progress enable 8
> >>> > voice-class h323 50
> >>> > session target ipv4:10.1.80.6
> >>> > dtmf-relay h245-alphanumeric
> >>> > codec g711ulaw
> >>> > no vad
> >>> >
> >>> >
> >>> > _______________________________________________
> >>> > cisco-voip mailing list
> >>> > cisco-voip at puck.nether.net
> >>> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >>> >
> >>> >
> >>
> >
> >
>
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