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Disabling ISDN PRI Anonymous Call Rejection

Answered Question
Aug 3rd, 2009
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Hello,


I have a AS5350 with an ISDN PRI which handles incoming voice calls which get converted to SIP calls. I'm running 12.4(15) T8. I've noticed that inbound callers who have privacy enabled on their lines for their caller ID get a "number not available" announcement when they dial into the 5350. But when they prefix their call with a *82 (enable caller ID per call), they are able to get through. Turning on q931 debug, I can see that the call reaches the 5350 and is rejected with a ISDN Cause Code of '1'.


This seems to be a symptom of "anonymous caller ID blocking" being enabled. The question is, how to turn this off on the 5350 or be able to set the presentation and screening bits to the appropriate values to allow calls through.


Thank you.


- Bill


Correct Answer by paolo bevilacqua about 8 years 2 weeks ago

Please configure DP 100 as I've indicated above.

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paolo bevilacqua Mon, 08/03/2009 - 07:36
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On the 5350, do you have something like ?


dial-peer voice xxx pots

incoming called-number .

direct-inward-dial


bcepananet Mon, 08/03/2009 - 07:53
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Something similar, inbound/outbound pots dialpeer is as follows:


dial-peer voice 100 pots

numbering-type unknown

destination-pattern .+

direct-inward-dial

port 3/0:D


I also have a few specific inbound numbers used for T.37 fax:


dial-peer voice 900 pots

service fax-inbound

incoming called-number xxxxxxxxxx

direct-inward-dial

port 3/0:D



Correct Answer
paolo bevilacqua Mon, 08/03/2009 - 08:14
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Please configure DP 100 as I've indicated above.

bcepananet Mon, 08/03/2009 - 09:00
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I assume "incoming called-number" will not impact any outbound calls matching this dial peer since "incoming called-number" has a higher matching precedence than a "destination-pattern" dial peer. Right?


Does an "incoming called-number" dial peer handle this particular issue of "presentation/screening" differently from a "destination-pattern" dial peer? Is there some reference that you know of that I can review?


I will try this during a maintenance window tonight since this is a production system.


Thanks much Paolo!

paolo bevilacqua Mon, 08/03/2009 - 09:04
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Respectively, yes, and no.


You don't need that much caution because the change brings the GW into standard configuration, thing that it doesn't have now.


bcepananet Tue, 08/04/2009 - 03:29
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That was it Paolo! I made the change this morning and it worked. It's a little disconcerting that a subtle change like this regarding the dial peers actually has this kind of subtle behavior change. And it's not documented well. It was very difficult to track down because very few people would report it, very infrequently and when it did happen, it would seem like a far end problem.


Thanks very much for your assistance!


-Bill


paolo bevilacqua Tue, 08/04/2009 - 03:39
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One have to understand how the router works. After you do, most things start making sense.


Thanks for the nice rating and good luck.

bcepananet Tue, 08/04/2009 - 08:22
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Hmm, too quick to conclude.


I had shutdown my original dial-peer and replaced it with the one you suggested which fixed the privacy issue. But it broke my outbound (SIP to PSTN) faxing, however it did not break my outbound (SIP to PSTN) voice calls for some reason.


Unfortunately, I have to revert back.

Not sure if I can keep both dial peers in service.


-Bill


paolo bevilacqua Tue, 08/04/2009 - 08:43
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You must have been changing something else.

incoming called-number on a pots DP cannot influence faxing.


bcepananet Tue, 08/04/2009 - 08:50
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Nope, nothing else was changed.


We send faxes out to PSTN fax machines in various offices all over the country by using a fax server that initiates a SIP/T.38 fax over IP to the Cisco which converts the T.38 protocol to G3 Fax out the ISDN PRI.


That's how pots dial peers are connected to my faxing.


That's why I find it very strange that my outbound fax calls seem to all fail, while my outbound voice calls (again SIP to PSTN) seem to work fine.


I suspect there must be some other dial peer that's taking precedence. But I find that very hard to believe since I don't have specific specific outbound PSTN digit matching.


-Bill



paolo bevilacqua Tue, 08/04/2009 - 08:55
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Take "debug voip dialpeer" then.

bcepananet Tue, 08/04/2009 - 09:23
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Yes it will require some more detailed analysis - thanks.


-Bill


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