CUCM 6 and Voipgate Trunk

[Gverbist]

Hi,

I must tell you I'm new to this cucm thing. I played around with 3CX and Trixbox for a while but now I got my hands on a CUCM servers and some cisco phones. I have CUCM setup and the phones connected to the server. What I want todo now is setup the system in such a way that when I dial a 0 and then the phone number that the calls are routed to the SIP trunk. I have no clue how todo this in CUCM. Could anybody be so kind to tell me step by step what todo? I read the documentation but it still does not give the desired result. Thanks in advance for you patience.

Geert Verbist

 

[CyberX666]

Im new at CUCM too, but i think i can point you in the right direction.

First setup a partition, call it PT_SIP_TRUNK or something. Then creat a Calling calling search space or add your new PT to one you have.

Go to TRUNK and creat a SIP trunk using the PT and calling search space you created. You also need to setup SIP security under the server TAB. Use TCP 5060 for easysetup at first then you can play with TLS once its working.

Now you should just need to creat a Route Pattern e.g 0. route to SIP_TRUNK strip digits= pre-dot

Hope that helps a bit, as i said i have just started CUCM.

 

[Gverbist]

mmmm I tought it would be something like that but it's the exact settings that bother me. What to you put in you route pattern to make sure when I press 0 that it routes to the sip trunk? Should me phone be a member of the partition? stuff like that is really new to me.but thanks..

I need more answers ;-)

 

[CyberX666]

Your PT_SIP_TRUNK partition should be in a calling search space then on the phone/line put the CSS. For the route pattern just put 0. thats "zero dot" dont forget to select predot in the drop down on the same page, so when you dial 02000 eg it will strip the 0 and just send 2000, then where its says route select the route SIP TRUNK that you made under trunk.

 

[Gverbist]

Ok I got this but it still say " this call cannot be completed"

The problems is the call pattern I think. I need to form the nummer and then tell it to use the trunk.

so I need todo something like 0.003222513609

each time it says, call cannot be completed

 

[CyberX666]

Ok, try using the call analyzer in servisability to see whats going on.

 

[Gverbist]

Nice tip mate. I am so noob in this..crap. I hate it when I don't understand a thing about it. Would you be able to help me if I pay you for configuring it?

 

[Gverbist]

Cisco Unified Communications Manager Dialed Number Analyzer


DNA Analysis Output Save the Displayed Output


Cisco Unified Communications Manager Dialed Number Analyzer Results



Results Summary
Calling Party Information
Calling Party = 1000
Partition =
Device CSS =
Line CSS = SIPSPACE
AAR Group Name =
AAR CSS =
Dialed Digits = 00032476242305
Match Result = RouteThisPattern
Matched Pattern Information
Pattern = 0.XXXXXXXXXXXXX
Partition = PT_SIP_TRUNK
Time Schedule =
Called Party Number = 0032476242305
Time Zone = Greenwich Standard Time
End Device = VoipGate
Call Classification = OffNet
InterDigit Timeout = NO
Device Override = Disabled
Outside Dial Tone = NO
Call Flow
Route Pattern attern= 0.XXXXXXXXXXXXX
Positional Match List = 0032476242305
DialPlan =
Route Filter
Filter Name =
Filter Clause =
Require Forced Authorization Code = No
Authorization Level = 0
Require Client Matter Code = No
Call Classification =
PreTransform Calling Party Number = 1000
PreTransform Called Party Number = 00032476242305
Calling Party Transformations
External Phone Number Mask = NO
Calling Party Mask =
Prefix =
CallingLineId Presentation = Default
CallingName Presentation = Default
Calling Party Number = 1000
ConnectedParty Transformations
ConnectedLineId Presentation = Default
ConnectedName Presentation = Default
Called Party Transformations
Called Party Mask =
Discard Digits Instruction = PreDot
Prefix =
Called Number = 0032476242305
Device :Type= SIPTrunk
End Device Name = VoipGate
PortNumber =
Device Status = UnKnown
AAR Group Name =
AAR Calling Search Space =
AAR Prefix Digits =
Call Classification = OffNet
Calling Party Selection = Originator
CallingLineId Presentation = Default
CallerID DN =
Alternate Matches
Note: Information Not Available



IT SEEMS IT GET ROUTED BUT I GET A BUSY TONE WHEN I FORM THE NUMBER

 

[fieryhail]

Can you please explain a little more. What are your settings in CUCM for your SIP trunk? What is your Partition/CSS setup like? Are you using a SIP trunk as your gateway to the PSTN for instance? Are you using wildcards in your route pattern? What is your current route pattern, route group, line group, etc. settings? Are you using multiple SIP trunks? Thanks.