[cisco-voip] incoming dial peers

[Dane Newman]

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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