<div>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.</div>
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<div>dial-peer voice 1000 voip<br> description incoming Call<br> translation-profile incoming aa<br> preference 1<br> session protocol sipv2<br> session target sip-server<br> incoming called-number 16784663444<br> dtmf-relay rtp-nte<br>
no vad<br>!</div>
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<div>dial-peer voice 100 voip<br> description 1-5xx extension to PUBLISHER<br> preference 1<br> session target ipv4:10.1.80.6<br> incoming called-number [1-5]..<br> voice-class h323 50<br> dtmf-relay h245-alphanumeric<br>
codec g711ulaw<br> no vad<br><br></div>
<div class="gmail_quote">On Fri, Oct 16, 2009 at 5:39 PM, Nick Matthews <span dir="ltr"><<a href="mailto:matthnick@gmail.com">matthnick@gmail.com</a>></span> wrote:<br>
<blockquote style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">Translation patterns work on SIP calls, yes.<br><br>to match any number - /.*/ /190/<br>
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<div class="h5"><br>On Fri, Oct 16, 2009 at 3:14 PM, Dane Newman <<a href="mailto:dane.newman@gmail.com">dane.newman@gmail.com</a>> wrote:<br>> What is a translation pattern to match any number?<br>><br>> On Fri, Oct 16, 2009 at 2:13 AM, Dane Newman <<a href="mailto:dane.newman@gmail.com">dane.newman@gmail.com</a>> wrote:<br>
>><br>>> Hello Nick<br>>><br>>> Thanks for your help<br>>><br>>> I created the the following config<br>>><br>>> voice translation-rule 1<br>>> rule 1 /*^/ /190/<br>>> voice translation-profile aa<br>
>> translate called 1<br>>><br>>><br>>> dial-peer voice 1000 pots<br>>> description incoming Call<br>>> preference 1<br>>> incoming called-number 6784663444<br>>> progress_ind setup enable 3<br>
>> progress_ind progress enable 8<br>>> dtmf-relay rtp-nte<br>>> no vad<br>>> translation-profile incoming aa<br>>><br>>><br>>> This should match the incoming called number 6784663444 and translate it<br>
>> to my auto attendant hunt pilot of 190 correct?<br>>><br>>> I had read below that Digit translation rules are not supported for<br>>> inbound session initiation protocol (SIP) calls.<br>>><br>
>><br>>> <a href="http://www.cisco.com/en/US/docs/ios/12_2/voice/configuration/guide/vvfpeers.html" target="_blank">http://www.cisco.com/en/US/docs/ios/12_2/voice/configuration/guide/vvfpeers.html</a><br>>><br>
>> I guess this has been updated in the latest code to work right?<br>>><br>>><br>>> On Thu, Oct 15, 2009 at 6:07 PM, Nick Matthews <<a href="mailto:matthnick@gmail.com">matthnick@gmail.com</a>><br>
>> wrote:<br>>>><br>>>> This may help:<br>>>><br>>>> <a href="http://www.cisco.com/en/US/partner/tech/tk652/tk90/technologies_tech_note09186a008010fed1.shtml" target="_blank">http://www.cisco.com/en/US/partner/tech/tk652/tk90/technologies_tech_note09186a008010fed1.shtml</a><br>
>>><br>>>> On your first example you don't have a codec set. It will be g729r8<br>>>> by default. As well, no DTMF-relay so you will not have that.<br>>>><br>>>> You probably want to add something like this:<br>
>>><br>>>> codec g711ulaw<br>>>> no vad<br>>>> dtmf-relay rtp-nte<br>>>><br>>>><br>>>> To 'plar' the number you would use a translation pattern.<br>>>><br>
>>><br>>>> Regarding your other problem - there isn't enough information. Your<br>>>> dial peer looks like a standard dial peer, but without a better idea<br>>>> of the call flow it's hard to tell. There's nothing specific in that<br>
>>> dial peer that would make it an incoming dial peer, but rather an<br>>>> outgoing dial peer.<br>>>><br>>>> -nick<br>>>><br>>>> On Thu, Oct 15, 2009 at 3:46 PM, Dane Newman <<a href="mailto:dane.newman@gmail.com">dane.newman@gmail.com</a>><br>
>>> wrote:<br>>>> > Hello All,<br>>>> ><br>>>> > I have a question about how to exactly configure incoming dial<br>>>> > peers. I<br>>>> > have a sip trunk to an ITSP for my incoming telco connection. The<br>
>>> > number is<br>>>> > 6784663444. There are four channels in the trunk but only one number.<br>>>> > Am I<br>>>> > matching the called number properly? How can I plar it to the unity<br>
>>> > connection auto attendant of hunt pilot 190?<br>>>> ><br>>>> > dial-peer voice 1000 voip<br>>>> > description incoming Call<br>>>> > preference 1<br>>>> > incoming called-number 6784663444<br>
>>> > progress_ind setup enable 3<br>>>> > progress_ind progress enable 8<br>>>> > !<br>>>> ><br>>>> > dial-peer voice 100 voip<br>>>> > description 190 AA<br>
>>> > preference 1<br>>>> > destination-pattern 190<br>>>> > progress_ind setup enable 3<br>>>> > progress_ind progress enable 8<br>>>> > voice-class h323 50<br>
>>> > session target ipv4:10.1.80.6<br>>>> > dtmf-relay h245-alphanumeric<br>>>> > codec g711ulaw<br>>>> > no vad<br>>>> ><br>>>> ><br>>>> ><br>
>>> > Also....<br>>>> ><br>>>> ><br>>>> > If I have another gateway across a WAN sending calls to extentions<br>>>> > 100-500<br>>>> > Will these incoming dial peers match the incoming calls and route them<br>
>>> > to my<br>>>> > cucm 7 properly?<br>>>> ><br>>>> > dial-peer voice 100 voip<br>>>> > description 1-5xx extension to PUBLISHER<br>>>> > preference 1<br>
>>> > incoming called-number [1-5]..<br>>>> > progress_ind setup enable 3<br>>>> > progress_ind progress enable 8<br>>>> > voice-class h323 50<br>>>> > session target ipv4:10.1.80.6<br>
>>> > dtmf-relay h245-alphanumeric<br>>>> > codec g711ulaw<br>>>> > no vad<br>>>> ><br>>>> ><br>>>> > _______________________________________________<br>
>>> > cisco-voip mailing list<br>>>> > <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br>>>> > <a href="https://puck.nether.net/mailman/listinfo/cisco-voip" target="_blank">https://puck.nether.net/mailman/listinfo/cisco-voip</a><br>
>>> ><br>>>> ><br>>><br>><br>><br></div></div></blockquote></div><br>