help with transformation

[r2d22y]

Hello

Lets say a user is dialing 95080

I want to remove first digit (9) and add +461111 before 5080 so it matches my id in my Asterisk so the final number/id will be +4611115080. That number/id is then used when using my SIP-trunk.

How can I achieve this?

Regards /D_S

 

[pndscm]

In your Route Pattern, enter the pattern as 9.5080 or 9.XXXX. Then in the Route Pattern Called Party Transformations discard digits field select Pre-Dot and in the Prefix-Digits field enter 461111.
If you use the 9.XXXX you may get into an overlapping pattern condition and have a 10-15 sec delay so make this as specific as possible. If you can't make it specific enough to stop the delay, add # at the end of the number and change the discard to PreDot Trailing #.

 

[r2d22y]

Thanks for your answer but it doesnt work.

I have tested that before but I get this message "Your call can not be completed" The funny thing is I have done a trace on my Asterisk server that my SIP-trunk should direct calls to and if Im dialing the number 95080 and do as you (pndscm) described above I can see in the trace that no substitution has been made it gets 95080@ip_of_my_my_asteriskserver.

So the sip-trunk is directing calls but no substitution is made, I cant figure this out.

What am I doing wrong?


Thanks for your help!

Regards /D_S

 

[jerake75]

It might be easier for you to use a translation pattern instead of trying to manipulate the digits on a route pattern...If all extensions to your Asterisk server are consistent (i.e. they all start with 5 or 508), then you can throw in a translation pattern of 9.5XXXX, then in the Called Party Transform Mask field, enter 4611115XXXX.

Also, if I remember correctly, CCM is going to try to match translation patterns first above all else, because they're always considered urgent even though there's no urgent field in their configuration pages.

I use translation patterns all the time, to redivert calls to an extension unknown to the end user.

I hope this helps -

Tina

 

[jerake75]

Couple of other things:

There's also a prefix digits field in the Translation pattern configuration page; you can try that as well with just 461111.

Make sure the Calling Search Space you choose on the translation pattern contains the partition that contains a route pattern of 46111, or whatever RP you've set up to dial your SIP trunks, so that it can dial out the correct gateway.

TMH

 

[r2d22y]

Hello

Thanks you all for your help I have followed your advices but no tranformation is made,.... the server is about to get thrown out of my window of the 5th floor ....

Do I have to have any route group or route list when using route pattern or route transformation. As I know, with my little knowledge of CCM, RG and RL are used to priorities traffic or are those essential parts even if I dont need that?

For the moment I'm just using translation pattern which has following setup:

Pattern definition
-------------------------------
Translation pattern: 95.XXX
Partition: MyPartition
Route Filter:<None>
Calling Searc Space: MySearchSpace (has MyPartition added)
MLPP Precedence: Default


Called Party Transformation
--------------------------------
Discard Digits: PreDot
Called Party Transforma Mask: 4611115XXX
Prefix Digits:4611115

Another question is... Lets say I'm making a call from a CCM_phone to an Asterisk-phone and CCM-phone has number 5040 shouldn't then the from field in the SIP request be 5040@ip_of_CCM_server? As it is know it becomes 00@ip_of_CCM-server...or is CCM handling this transformation 00->5040?

I really cant understand that this is so hard to get to work when everything is predefined with webbinterface and fields to fill.

Your help is very appreciated!

Regards/D_S

 

[jerake75]

Alright, lets start at the beginning...can't help much with the last question, since I've never worked with SIP, but let's see if we can get your routing straightened out to your SIP trunk.

Also, forget the Translation Pattern for now, and let's go back to Route Patterns.

First, you have your SIP trunk added as a trunk in CCM, not a gateway, correct? And there's no caller ID DN set on it...? I know with gateways, a great deal of the time, any caller ID DN or transformation on the gateway will override any other pattern you have set in a TP/RP/RL. Also, check your settings on your SIP trunk in CCMAdmin - do you have a CSS assigned to your trunk containing the partition that contains the RP (95.XXX or whatever you're using) that is pointed at the SIP gateway?

Second, where are your outbound calls going - are your outbound patterns (e.g. 9.@, 9.1XXXXXXXXXX, 9.XXXXXXX, if you use explicit patterns) pointed to a separate gateway?

Third, route patterns - create a route pattern with the pattern of 95.XXX. For this example's sake, say you assign it a partition of SIPTrunkPT. Make sure you assign the SIPTrunk, if you can, in the Gateway or Route List Field. Make sure Route This Pattern is selected, and select the call classification as OffNet, since you're using outbound dial prefix (9). Question: Are these calls all within the same location, or two different ones? Why use a dial prefix if these calls are staying within the same company?

Now, when you get down the bottom of the RP configuration, enter in the following in the Called Party Transformations section:

Discard Digits: PreDot
Called Party Transforma Mask: 4611115XXX

Don't use the Prefix Digits field also, one or the other.

This should work, provided that you don't have anything else overriding the outbound digit Caller ID.

There's one more thing we can try, but give this a shot first, and let me know.

TMH

 

[r2d22y]

Thank you jerake75 its working!

I deleted all settings and followed your advice and it worked

A funny thing that I have noticed is even if I delete, in this case a route pattern, it seems like it still is in use(?)
I know that becuase I deleted one and then made a trace with Ethereal and it still direct current number after given transformationrules(?).

Do I have to do a reload/reset on other things when I have deleted or added new routepatterns?

Thanks again!

/D_S

 

[jerake75]

I'm glad it's working!

Hmmmm if you delete a route pattern, things should no longer work correctly, no reset of phones needed...what version of CCM do you have (3.X, 4.X)? I know that a few versions back there was a bug related to route patterns, where you had to reset all the phones or something like that after you deleted a route pattern (sucked big time, I ran into it).

In 4.1.3, the latest version, the routing is dynamic, so if you delete a RP/TP etc., it immediately stops working. Let me know what version you have.

TMH

 

[r2d22y]

Hello

What I can remember I'm running 4.1 I cant control for the moment becuase we are switching offices so no servers are available.

/D_S