MLS Inbound calls Route From CUCM to Asterisk extension. Using only routing 1 DID to Asterisk Extension

[YANWILLIE4925]

Would appreciated that anyone could give me some help on this.
I had a MLS service hook to Cisco 2900 router and connected to CUCM. 
I'm trying to create a lab and referring one of the DID numbers from the MLS services connected to CUCM and route it to one of the extension in the asterisk.
I've been to do a sip trunk link between asterisk and cucm. Intercom works between both side's extension.
I can even do a outbound route from asterisk extension to my mobile phone numbers.
But inbound doens't works.
I've configured other pabx before and every is a breeze.
Such as panasonic pabx system, i only need to configure DID to our DIL/DID section and locked a DID number to an extension which will automatically dial to that extension. And best of it is that the DID can be point to an extension on the opposite pbx's extension. As for cucm, everything seems so rigid and legacy...

[YANWILLIE4925]

I guess that the great o mighty cisco products, with it's forums under the official site of cisco could have such a dormant state community dealing with pleas and woes of users. I barely find it bustling with eager forumers to take on questions despite we have erm... ccna,ccnp and also ccie haunting these forums. I guess it does not make a difference switching to Hwahuei products that are generally cheaper options but had better community responsive fellas to dealt with your request.
Sometimes i just find it upsetting to find my questions thrown down the rubbish chute at a corner not even looked on by anyone in the community.

[YANWILLIE4925]

I guess that the great o mighty cisco products, with it's forums under the official site of cisco could have such a dormant state community dealing with pleas and woes of users. I barely find it bustling with eager forumers to take on questions despite we have erm... ccna,ccnp and also ccie haunting these forums. I guess it does not make a difference switching to Hwahuei products that are generally cheaper options but had better community responsive fellas to dealt with your request.
Sometimes i just find it upsetting to find my questions thrown down the rubbish chute at a corner not even looked on by anyone in the community.

[TONY SMITH]

So your inbound call comes from the SIP trunk into CUCM, and you want to route it on to Asterisk, is that correct?   And you have a SIP trunk between CUCM and Asterisk?   If so, and making a guess about your CUCM configuration you probably want a route pattern matching the dialled number, in a partition that's included in the incoming trunk CSS.  Point the route pattern to your Asterisk trunk.   The route pattern, matching one specific DID, will take precedence over less specific configuration matching the DID range as a whole.

Difficult to be more specific without knowing your exact architecture and numbering plan.

[YANWILLIE4925]

Thx a great deal for the explaination.
Yes , it's SIP trunk with CUCM and i wanted to route 1 DID number to Asterisk Extension Number. It's a MLS to be precise though the outgoing & incoming trunk services.
For the SIP trunk connection between CUCM and Asterisk has been done. I had Created a SIP trunk, SIP Profile Except for Calling Search Space.
For Calling Search Place, i did not create one for Asterisk, instead i used the existing CSS of the CUCM.
I'm not sure what am i missing. Should i configure a translation pattern for this individual DID to Asterisk ? It seems that the translation pattern is for routing incoming calls to the device pool. But it's set with ranges, Which i'm unsure if an individual DID can be set to route to the Trunk of the Asterisk..
Let me try again and update u the results...

[TONY SMITH · Accepted solution]

"Should i configure a translation pattern for this individual DID to Asterisk ? It seems that the translation pattern is for routing incoming calls to the device pool. But it's set with ranges, Which i'm unsure if an individual DID can be set to route to the Trunk of the Asterisk.."

It might be helpful to review the difference between a Route Pattern and a Translation Pattern.  A translation pattern matches the dialled number, changes any or all of called number, calling number, CSS and maybe some other parameters as well, CUCM then processes the call based on those changes.  You could consider the translation pattern as an intermediate step within CUCM.

In contrast when a Route Pattern is matched it directs the call to a Route List or direct to a Gateway, in other words it makes the decision to route the call out from CUCM.

Hence I was suggesting a Route Pattern.  The pattern should match the single DID number exactly as presented by your MLS service, and optionally you could make any changes needed to make that number acceptable to your Asterisk installation.

We'd need to know more about your dial plan and your existing CUCM configuration to be more specific.

[YANWILLIE4925]

Here's the configuration.

Service Provider MLS TRUNK(Multi Line SIP)
Asterisk SIP Trunk to CUCM

SIP Profile for MLS TRUNK to Router 2900

Asterisk SIP Profile to CUCM

CUCM Route Plan

Route Pattern Configuration for Asterisk

CUCM Translation Pattern

Translation Pattern Configuration to CUCM Internal Extension Example 1.

Translation Pattern Configuration to CUCM Internal Example Example 2

[TONY SMITH]

Your screen prints come up too small for everything to be readable.  However irrespective some things can be seen,  for example it looks as if extensions on CUCM are four digit form 3XXX,  corresponding to the last four digits of the DID.   I assume your second example is a typo and should also be 3XXX not 3XXXX as shown.

So lets say you want to route only DID 032253123 to Asterisk

Create a Route Pattern with  032253123 in the same partition as your Translation 03225.3XXX so that it can be seen by the trunk and will be a more explicit match.   Point to the same gateway or route list that your 1XXX Route Pattern uses.  If you need to use a different extension number in Asterisk then you can either convert at that end, or use a Called Party Transform Mask in the Route Pattern for example XXX to reduce to last three digits.

Hope this helps.

[YANWILLIE4925]

thx for the effort to reply and see those terrible image i uploaded... I took quite alot of time snipping and pasting them into one piece hoping it would get more details on the setup of my cucm.. Unfortunately, it turn out to be worse. Everything shrunk to be unreadable.
Extensions of the CUCM are actually 5 digits. There are 2 sets of partition though, we had 2 sites in different country. With one site utilizing Extension 2XXXX while the other 1XXXX. 
I believed that CUCM might have refer inbound traffics from the Device_pool or the CSS_pool that the DN associates  .
I did what u mentioned. I did a route pattern specifically with 6032253123 *Our isp needs to have a 6 infront for it to reach out or dial in* to Asterisk. So as i did the routing, i noticed that supposingly extension 33123 should ring if outside dials 6032253123. It stopped ringing but only to be forwarded to a service not valid voice from our Service provider auto answer msg. So i guess it did really worked that route pattern indeed directed the call to 6032253123 away from extension 33123.
However, i checked the CDR reports of the Asterisk. Logs showed that calls calling into 6032253123 did not divert to any particular extension. It shows that incoming number, cucm responds to 6032253123 but it went to a " , " log... Which means it does not know where is asterisk. Since i already did the asterisk inbound route, i guess the problem lies with the cucm. The route pattern i route it to a Asterisk trunk i created. Not sure did i do it right.
I tried changing the calling party information with prefix. But seems that i'm wrong.
This CUCM really driving me nuts..

[TONY SMITH]

You can check the CUCM routing using Dialled Number Analyzer (x.x.x.x/dna where "x.x.x.x" is ip address of CUCM).  Check the incoming PSTN trunk to see that it correctly matches your incoming DID.

Assuming it's correctly matching and pointing to the Asterisk trunk, I'd suggest taking a Wireshark trace, or CUCM traces and look at the SIP decodes for the CUCM<->Asterisk leg.

I'm not clear what you mean by .. "I believed that CUCM might have refer inbound traffics from the Device_pool or the CSS_pool that the DN associates".   Unless you have a very unconventional setup inbound routing won't be affected by Device Pool, or the CSS of the destination DN.  Inbound routing depends on the CSS of the Trunk, and the Partitions of the destination patterns.

[YANWILLIE4925]

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@TONY SMITH wrote:

You can check the CUCM routing using Dialled Number Analyzer (x.x.x.x/dna where "x.x.x.x" is ip address of CUCM).  Check the incoming PSTN trunk to see that it correctly matches your incoming DID.

Assuming it's correctly matching and pointing to the Asterisk trunk, I'd suggest taking a Wireshark trace, or CUCM traces and look at the SIP decodes for the CUCM<->Asterisk leg.

I'm not clear what you mean by .. "I believed that CUCM might have refer inbound traffics from the Device_pool or the CSS_pool that the DN associates".   Unless you have a very unconventional setup inbound routing won't be affected by Device Pool, or the CSS of the destination DN.  Inbound routing depends on the CSS of the Trunk, and the Partitions of the destination patterns.

Thanks for the guide and advise.. I'm not very well verse with cucm since this was my first cucm interaction with intergration of an open source pbx system. But most of the direction u provided worked really well.
The routing pattern actually was the one i should be looking at. 
I crossed checked the log from Asterisk side since it's producing a "s" result when i intend to call into Asterisk inbound was a result of entering CID number into the Asterisk Trunk properties. Hence why the inbound call failed.CUCM did redirected the inbound calls to Asterisk. Just that Asterisk was given a wrong information on it's trunk hence why the call got terminated before it reached the extension.

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