Voice Over IP
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what should my route pattern look like?
I also want CDR to log the call as 91 111 222 3333 or 1 111 222 3333 or 111 222 3333
CUCM version is 8.0.2
Thanks.
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Calling Party Information
Calling Party = 1001
Partition =
Device CSS =
Line CSS = Test CSS
AAR Group Name =
AAR CSS =
Dialed Digits = 911112223333
Match Result = BlockThisPattern
Route Block Cause = Unallocated Number
Called Party Number =
Matched Pattern Information
Pattern =
Partition =
Pattern Type =
Time Zone =
Outside Dial Tone = NO
Call Flow
TranslationPattern :Pattern=
Partition =
Positional Match List =
Calling Party Number = 1001
PreTransform Calling Party Number =
PreTransform Called Party Number =
Calling Party Transformations
External Phone Number Mask = NO
Calling Party Mask =
Prefix =
CallingLineId Presentation =
CallingName Presentation =
Calling Party Number = 1001
ConnectedParty Transformations
ConnectedLineId Presentation =
ConnectedName Presentation =
Called Party Transformations
Called Party Mask =
Discard Digits Instruction =
Prefix =
Called Number =
Alternate Matches
Note: Information Not Available
Match Result = BlockThisPattern
It seems like you are calling from an IP Phone, but there is no such route pattern
911112223333, or you dialed it incorrectly:
Route Block Cause = Unallocated Number
Or the Calling Search Space assigned to the line doesn't contain the partition where the route pattern is:
Line CSS = Test CSS
Could you please verify what is the calling device?






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I don't know if this helps but the CUCM server is in europe and we are trying to make life easier for users calling the US i.e. instead of dialing 9001 222 333 4444 they can just dial the local US number. i.e. 222 333 4444...a route pattern should handle the number manipulation....discarding

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Calling Party = 1111
Partition =
Device CSS =
Line CSS = Phone VPN CSS
AAR Group Name =
AAR CSS =
Dialed Digits = 912223334444
Match Result = RouteThisPattern
Matched Pattern Information
Pattern = 9.1XXXXXXXXXX
Partition = Phone_VPN
Time Schedule = All the time
Called Party Number = 0012223334444
Time Zone = America/New_York
End Device = RL_PSTN
Call Classification = OffNet
InterDigit Timeout = NO
Device Override = Disabled
Outside Dial Tone = NO
Call Flow
TranslationPattern :Pattern=
Partition =
Positional Match List = 12223334444
Calling Party Number = 1111
PreTransform Calling Party Number =
PreTransform Called Party Number =
Calling Party Transformations
External Phone Number Mask = NO
Calling Party Mask =
Prefix =
CallingLineId Presentation =
CallingName Presentation =
Calling Party Number = 1111
ConnectedParty Transformations
ConnectedLineId Presentation =
ConnectedName Presentation =
Called Party Transformations
Called Party Mask =
Discard Digits Instruction =
Prefix =
Called Number =
Route Pattern :Pattern= 9.1XXXXXXXXXX
Positional Match List = 12223334444
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 = 1111
PreTransform Called Party Number = 912223334444
Calling Party Transformations
External Phone Number Mask = NO
Calling Party Mask =
Prefix =
CallingLineId Presentation = Allowed
CallingName Presentation = Allowed
Calling Party Number = 1111
ConnectedParty Transformations
ConnectedLineId Presentation = Default
ConnectedName Presentation = Default
Called Party Transformations
Called Party Mask =
Discard Digits Instruction = PreDot
Prefix = 00
Called Number = 0012223334444
Route List :Route List Name= RL_PSTN
RouteGroup :RouteGroup Name= RG_PSTN
PreTransform Calling Party Number = 1111
PreTransform Called Party Number = 912223334444
Calling Party Transformations
External Phone Number Mask = Default
Calling Party Mask =
Prefix =
Calling Party Number = 1111
Called Party Transformations
Called Party Mask =
Discard Digits Instructions =
Prefix =
Called Number = 912223334444
Device :Type= H323Gateway
End Device Name = 10.10.10.5
PortNumber = 0
Device Status = UnKnown
AAR Group Name =
AAR Calling Search Space =
AAR Prefix Digits =
Inbound Fast Start = Disabled
Outbound Fast Start = Disabled
Call Classification = Use System Default
Tunneled Protocol = None
Calling Party Transformations
PreTransform Calling Party Number = 1111
Calling Party Selection = Originator
Calling Party Presentation = Default
CallerID DN =
Calling Party Number = 1111
Alternate Matches
Note: Information Not Available
I asked about the number of servers so you can try the same DNA analysis from the other server. So if you navigated to http://pubIP/dna, and tested from there, try http://subIP/dna instead and determine if from the subscriber the routing is not working.






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utils network capture eth0 count 100000000000 size all file h323
Then reproduce the call until you get the annunciator message. Cancel the packet capture with CTRL-C and retrieve the file as instructed here:
https://supportforums.cisco.com/docs/DOC-11599

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----> Calling party number: '5597' ---->
----> Called party number: '0011112223333'---->
<---- Cause value: Unallocated (unassigned) number <----
You can't see the debug isdn because your router is probably not configured to match anything with destination-pattern 0011112223333 to be routed out the PRI.
Now, didn't you configured somewhere to send 912223334444 in CallManager? Where was it exactly?
The "real" test number I am using is 18882378289...shouldn't this number still match the 9.1XXXXXXXXXX# route pattern?
As far as I remember you are trying to debug if your call hits the router by means of a debug isdn q931.
This is not going to show the attempt because the router can't even route 0011112223333 (based on your last post). Once you correctly configure a dial peer to route out through a ISDN circuit, then you will be able to see the output in the debug.
Now if you dial 9 18882378289, you should also match the route pattern. Why don't you repeat the packet capture while dialing the real number and upload to cloudshark? We may discover a misconfiguration in the gateway instead.

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I assume the route pattern you suggested should match any US and Canadian number. Correct?






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http://www.cloudshark.org/captures/1327667d0960
http://www.cloudshark.org/captures/995ba1423ad8

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http://www.cloudshark.org/captures/1327667d0960 has 2 packets showing a call to 0011112223333
Maybe you want to take a DNA analysis of a call to 18882378289 and make sure its being routed out.
http://www.cloudshark.org/captures/e0bcc0b05657 -> Called party number: '0011112223333'
http://www.cloudshark.org/captures/06eb5e80d2f6 -> nothing






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Once that is done, go to your Route Pattern **918882378289, and modify it to this:
**9.18882378289, and then add a discard digit instruction of Predot.
Go to your gateway and created the following dial-peer:
dial-peer voice 99099 pots
destination-pattern 18882378289
no digit-strip
port 0/0/0 => your ISDN voice port
Now try a call while debug voip ccapi inout and debug isdn q931 are enabled with the term mon command.
**918882378289

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Remember I told you to configure a dialpeer as follows:
dial-peer voice 99099 pots
destination-pattern 18882378289
no digit-strip
port 0/0/0 => your ISDN voice port
Now I realize you did some extra manipulations of your own. No worries, the matching dialpeer should be as follows based on your last posts:
dial-peer voice 99099 pots
destination-pattern 0018882378289
no digit-strip
port 0/0/0 => your ISDN voice port
Please debug isdn q931 a call to **918882378289 once you make the changes in the IOS






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destination-pattern 001..........
route pattern should be: 9.001[2-9]XX[2-9]XXXXXX and remove the predot and prefix digits of 00
then the destination pattern on the dial-peer should be destination-pattern 001..........?
finally...I have 2 ISDN voice ports "port 0/0/0:15" & "port 0/0/1:15". does this mean I need 2 dial-peers?

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And finally, yes, you would need 2 dial-peers configured with different voice ports. That will make the system load balance between them






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debug isdn q931
debug voip ccapi inout
term mon
when you dial the call from an ip phone
route pattern:
9.1XXXXXXXXXX
discard predot
prefix 00
Dial peer on gateway:
dial-peer voice 6001 pots
destination-pattern 001T
port 0/0/0:15
forward-digits all

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Voice Over IP
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Voice over IP (VoIP) is a methodology and group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet. Other terms commonly associated with VoIP are IP telephony, Internet telephony, broadband telephony, and broadband phone service. The term specifically refers to the provisioning of communications services (voice, fax, SMS, voice-messaging) over the public Internet, rather than via the public switched telephone network (PSTN). Examples of the VoIP protocols are H.323, Media Gateway Control Protocol (MGCP), Session Initiation Protocol (SIP), H.248 (also known as Media Gateway Control (Megaco)), Real-time Transport Protocol (RTP), Real-time Transport Control Protocol (RTCP), Secure Real-time Transport Protocol (SRTP), Session Description Protocol (SDP), and Inter-Asterisk eXchange (IAX).