[cisco-voip] incoming dial peers

Nick Matthews matthnick at gmail.com
Fri Oct 16 17:39:41 EDT 2009


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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