Adding a Route Pattern to a Line Group?

Hi
We have an analog device which is patched into our Voice Gateway and has a dial peer using extension 444.
I have setup a route pattern within CUCM which points at the gateway for 444.
However I can't seem to add this Route pattern to a line group to be part of a hunt?
The line group seems to be just internal extensions but I need the route pattern of 444 to be in the same line group as two internal extensions?
Thanks

Hi
Route Patterns are used for destinations that are 'off-ssytem' i.e. over trunks and so on.
Line Groups can only contain local lines on the system. This pretty much means only SCCP controlled lines, as CUCM is fully aware of the line state at all times.
If you want a port to be in a line group, you'll need to register it to CUCM as an SCCP gateway.
Aaron

Similar Messages

Matching Route Patterns with standard Local Route group and Specific Route Group

Hi
I have a customer with CUCM 8.6 with few branches
couple of branches in UK and few in Europe and middle east.
I configured route patterns with Standard local route group, but using their own Voice gateway, everything was working fine until adding the recent branch with matching pattern
UK has a mobile pattern with 9.07XXXXXXXXX (11 digits)
One Branch has a mobile with 9.07XXXXXXXX (10 digits)
When branch call 907X..(10digit) number there was a delay and I ticked the Urgent priority to process it quicker, but later realized the UK branch cannot dial 907x.. (11Digit) mobile.
I created Route List for branch and added the 10 digit pattern to that but still the UK cannot call 11 digit. so i believe when you call out it will check the pattern first and the Route-List and Route-Group and gateway play a part.
Is there a way to get 07 -10digit call out quickly also allowing the 07 -11digit pattern as well ( without changing the T302 timer)
Really appreciate your support
thanks
shameer

Yes, they key to managing overlapping centralized dial plans is to be really good with patterns, partitions, and CSSs. You can have 3 different 9.0[2-9]XX-[2-9]XX-XXXX patterns and assign them a different partition, and then assign that to the branch CSS. This will only work if each Branch has a different CSS.
For example:
9.0[2-9]XX-[2-9]XX-XXXX @ Egypt-PT ->Routes to Local route group of Egypt.
9.0[2-9]XX-[2-9]XX-XXX @ UK-PT -> Routes to Local route group of UK
9.0[2-9]XX-[2-9]XX-XXX @ Germany-PT -> Routes to Local Route group of Germany.
//PT = partition//
Then have Egypt-CSS that contains 9.0[2-9]XX-[2-9]XX-XXXX @ Egypt-PT.
UK-CSS contains 9.0[2-9]XX-[2-9]XX-XXX @ UK-PT
Germany- CSS contains 9.0[2-9]XX-[2-9]XX-XXX @ UK-PT
The other patterns will be invisible to your sites because they are in a different partition that is not in their CSS. 2 overlapping patterns in the same PT will cause you to wait for the inter-digit timeout unless you press #.
Thanks,
Frank

Port not available to be added to route group

CUCM 9.1.2
I have a 2911 (MGCP) with an FXO card. The FXO port is configured and shows as registered but will not show up in the ports list in my route groups so that I can add it. Every other port from my other MGCP gateways shows up fine. Resetting the port and having it re-register did not help. Any ideas?

If the port is assigned to a route pattern, it wont be an option available under Route groups and vice versa. I would check depending records on the ports to see where its assigned and change it.

Does the route list gets reset when adding route pattern in version 9 of CUCM?

Hi All,
I created a new route pattern in call manager version 9 and then associated it to an existing route list. But when I was trying to save the route pattern. I was prompted with the message saying " Any updates to this Route Pattern automatically resets the associated gateway or Route List"
I was not able to see this message on CUCM version 8.6 however on CUCM version 9 it is there.
Does that mean that the route list/gateway will be reset and the existing calls will be dropped?
Or is it a bug?
A response will be greatly appreciated.
Thanks

Hi Farhad,
It does pop up when we modify the Route Pattern in CUCM 8.6 too.
If you asociate the GW/Trunk directly to Route Pattern, yes, it will reset them and calls will be interrupted. To avoid that, we need to associate the Route List to Route Patterns.
//Suresh
Please rate all the useful posts.

How do I add a route pattern to CUCM 7.1

I am currently using CUCM7.1 and need to add the route pattern 9911 to dial out to emergency dispatch. I do not want the capability of dialing 9 for an outside line for all users, just when I a calling 911 emergency. We recently changed to dialing a # for an outside line due to excessive 911 hangup calls. I tried adding 9911 to the route pattern list but I am missing something. I received the message stating could not complete call as dialed. Thank you, Cindy

Are you sure the phone has a CSS that allows you to dial 9911??
HTH
java
If this helps, please rate
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Design RG in Route Pattern

I would like to set that all calls to a specific destination, were commuted to a Gateway A, and if all lines are busy, these calls are switched by Gateway B.
I have this configuration:
Actual RP:
Route Pattern* : 9199.XXXXXX      Partition: Outgoing Call      Gateway or Route List: 10.201.30.250
New Configuration (In mind....)
Route Pattern* : 919953.XXXX      Partition: Outgoing Call      Gateway or Route List: Gateway A
Route Pattern* : 919953.XXXX      Partition: Outgoing Call      Gateway or Route List: Gateway B
IP Address Gateway A: 10.201.30.250
IP Address Gateway B: 172.17.20.10
I had thought to create a Route Group List with the two Rotuers (GW 10.201.30.250 and 172.17.20.10), but the actual configuration of the Route Patterns in the Call Manager, is using the Gateways in the configuration of Router Pattern (in the configuration RGL are not set).
How I can configure this requeriments whitout RGL?
Whit this configuration:
Route Pattern* : 919953.XXXX      Partition: Outgoing Call      Gateway or Route List: Gateway A
Route Pattern* : 919953.XXXX      Partition: Outgoing Call      Gateway or Route List: Gateway B
If the lines in the GatewayA are busy the call is switched to the Gateway B? Or this is achieved only by setting RGL?
****Cisco Call Manager version: 4.2
Thanks

Hello,
Q/ New Configuration (In mind....)
Route Pattern* : 919953.XXXX      Partition: Outgoing Call      Gateway or Route List: Gateway A
Route Pattern* : 919953.XXXX      Partition: Outgoing Call      Gateway or Route List: Gateway B
A/ You can not have two identical route patterns on the same partition.
Q/ I had thought to create a Route Group List with the two Rotuers (GW 10.201.30.250 and 172.17.20.10), but the actual configuration of the Route Patterns in the Call Manager, is using the Gateways in the configuration of Router Pattern (in the configuration RGL are not set).
A/ If you want redundancy in CallManager, then you need route group and route list. If you can not add your routers to a route group is because they're already associated with route pattern(s). First, dissociate them from all route patterns, then create the route group and finally the route list. Now you can associate your route pattern(s) to the new route list. Final configuration should look similar to:
Route Group Name: My_Route_Group
                             Current members: 10.201.30.250
                                                        172.17.20.10
Route List: My_Route_List
                              Selected Groups: My_Route_Group
Route Pattern: 9199.XXXXXX      Partition: Outgoing Call      Gateway or Route List: My_Route_List
Route Pattern: whatever            Partition: Outgoing Call      Gateway or Route List: My_Route_List
etc
Q/ If the lines in the GatewayA are busy the call is switched to the Gateway B? Or this is achieved only by setting RGL?
A/ If your gateways are H.323, there's a way to achieve redundancy without involving CallManager. This is done with redundant dialpeers. I do not recommend it because that will not achieve the highest possible redundancy for your calls. For example, if gateway A is down, calls would never be routed to router B. However, if you configure a route list then CallManager will be able to route your calls to gateway B without problem.
Hope it helps, please rate if it does.
Good luck!.
Kind regards,
- Adrian.

Device Weight: Route Pattern Vs IP Phone

Hi, we are starting the planning for removing two CO class GTD5 switches that route 28,000 DID numbers for the County. The switches are +16 years old and maintenance is expensive. Our Callmanager cluster is designed to replace the existing phone infrastructure (2 CO switches, +60 PBXs, and +20 key systems). Here's a quick overview of our Cluster:
- Publisher only running database services (MCS7845H2)
- Two TFTP servers only running TFTP service and MOH streaming (multicast) (MCS7835H)
- Two subscribers configured as 1-to-1 backup (MCS7845H2)
- CCM version 4.1(3)sr3b
We are planning to add the additional six MCS7845H2 subscribers as we need the capacity.
The first step in our migration is to move all the DID's to new T1/PRI's on a set of six Communication Media Modules (CMM) spread out over four Cat6513 switches.
Essentially our CCM cluster will be acting as a Tandem switch until we get all the trunks moved off the GTD5 CO switches to the CMMs.
Unfornutately all of the 28,000 DIDs are pretty much shot gunned all over the County. So, we will have +20,000 route patterns in the beginning. Over time we will also be converting sites to IP Tel and removing the route patterns as we migrate.
My question: Does a route pattern for one directory number carry the same device weight as an IP phone with the directory number assigned to a line?
We are thinking it does and if it does, then we need to scale up our CCM cluster for 30,000 devices before we start the migration.
Thanks in advance for any advice.
Tom.

Thanks Greg for taking the time to reply. Our cluster design is following the "Cisco Callmanager Best Practices" book. The Publisher and TFTP servers are not running the Callmanager service. This allows 8 subscribers running the Callmanager service in the Cluster.
Our research is trying to understand the cost of a route pattern in terms of the dialing forest and the impact on the subscriber's memory and CPU.
We own two complete prefixes plus another 8,000 DNs from a third prefix. We fortunately do not have an overlapping dialing plan. Each directory number will either be assigned to an IP phone or it will belong to phone on a PBX.
All directory numbers for IP phones belong to the same partition and we also followed the Best Practices book for our dialing plan and use the line/device CSS design.
All +80 remote sites connect back to our GTD5 and the GTD5 routes all the numbers to the remote sites.
We are first migrating all the DID services to our new PRI's handled by the Callmanager cluster. We when start the migration it will be a simple process of a route pattern such as [4-5]XXXX to route the 874-xxxx and 875-xxxx numbers to the GTD5. Then as we disconnect the tie-lines from each remote PBX and re-connect it to the CMM for Callmanager to route, we will need to add all the specific route patterns to route the numbers for the site.
It would be ideal if we did not have to retire the GTD5 switch. We would follow our 5 to 7 year plan to migrate the entire County to IP Tel and leave the GTD5 in place routing the numbers to the remote PBXs. However, we have been directed by management to decommission the GTD5 switch within 12 months.
So we are trying to understand the impact to the subscribers when we begin adding 1,000's of route patterns. We are planning to consolidate as many of the route patterns as possible to reduce the number of route patterns. However, we inherited a design that we refer to as "Number-lose-ability", where individual numbers are routed and not blocks of numbers. Over the years of adds, moves, and changes the numbers have been scattered to all the sites. We have very few sites with consecutive numbers.
Another question that we are trying to answer: what is the cost of a route pattern such as 5555x compared to 10 individual route patterns for the same number range. Again, in terms of memory and CPU on the subscriber doing the digit analysis. We are asking this question because we may have 6 of the 10 numbers going to the same PBX, but the other 4 numbers each going to a different site. To consolidate route patterns we would add the 5555X pattern and the four individual route patterns. What we do not know is how the 5555x is added to the dialing forest. Is it expanded to 10 patterns or just one expression.
Thanks again for any help,

Failed to create the CTI Route Point and corresponding Line on Cisco Unified CM.

Dear All
Kindly I got that message "Failed to create the CTI Route Point and corresponding Line on Cisco Unified CM." When I am trying to create Trigger on the UCCX for an Application,
Bye the way I create Call Control Group and it gives me OK, and I create it from Subsystem>Cisco Unified CM Telephony>Call Control Group
and Type CTI Ports: 4 | Inbound | Starting DN : 9000 "Nothing in my route pattern report have created CTI numbers"
So Appreciate your help to avoid that errors raised when I am trying to create new prompts
Thanks
Ahmed Samir Yosuef
Integration Manager

Let me know the deployment model and Node details

Route Pattern issue

I am running Cisco Unified Communications Manager Version 8.6 and I am trying to add a route pattern that I can not get to work. I am copying an existing route pattern that works and only changing the prefix. i.e. 9.331XXXX copied and changed to 9.551XXXX. Both are in the same route list and group and I have verified that the new prefix is a local prefix. When dialing the 551 number it is telling me you need to dial a 1 first. Dialing 9.1551XXXX gets the message that says "your call can not be completed as dialed. please consult your directory.......". If you dial it as 9-1-area code-551XXXX it still tells you that you need to dial a 1 before dialing the number.
Not sure what I am missing but any suggestions would be much appreciated.
John

The only message that comes from CUCM is the "your call cannot be completed as dialed..." all of the other messages come from your telco, you would need to ask them exactly what digits they're expecting to accept the call as valid.
Did you change the RP to be 9.1551XXXX or did you simply tried to dial with 9.551XXXX configured?????
I'm guessing the latter as you got the "your call cannot be completed as dialed..." message from CUCM

Route pattern to SIP trunk problem

Hello, I have a 2801 router that has been configured with CME and a working SIP connection to my local ISP.
Tested with calls via CME so I know for sure that the SIP config and dial plan is fine on this gateway.
Next I wanted to try out CUCM so I set up a CUCM 8.6 box that is connected to the 2801 router to use as it's SIP gateway.
The only change I made to the gateway router config was to alter the "ip option 150" address so that the phones go to CUCM for their configs etc (which they do with no problems).
Then I set up a SIP trunk in CUCM along with a route pattern which is to use the SIP trunk within the Gateway/Route list option.
But when I make a call that matches this route pattern all I get is the intermittent beep message from the phone. I cannot route calls succesfully through it.
I have checked network connectivity and all is fine. The IP address I specfied in CUCM for the SIP trunk is simply one of the interfaces on the 2801 router and it is definitley reachable.
I also activated "debug ccsip all" on the 2801 gateway router but nothing appears. So it seems like the calls are not even reaching the 2801 gateway ?
Is the problem possibly a conflit between CME on the gateway router and my CUCM ?
Do I need to disable CME somehow on the gateway first ? Or am I not doing something correct in the CUCM config ?
Thank you kindly for any suggestions.
ps. I have attached a couple of screenshots of my config.

Hello, thanks for helping.
I activated "debug voice ccapi inout" as well as "debug ccsip all" on the gateway but nothing showed up.
Therefore I deduce the call is not even making it to across the SIP trunk into the gateway router ?
As I am a newbie trying this out for the first time, it is guranteed to be something really simple.
I have included my running config from the gateway router below..
One addition I made was to add an incoming dial peer. That is "dial peer 5, description CUCM SIP trunk".
I set it up with a destination patter 2... to match my phone config on CUCM which have numbering in the 2000 range.
Sorry, I got RTMT up and running but could not get any meaningful results from it. I need to learn up on that.
I did however run a 'dialed number analysis' from CUCM direct and have attached the result. It seems the dialled number "99" is matching the route pattern OK.
So why is it not then moving down the SIP trunk to my gateway and getting picked up by the incoming dial peer ?
Thanks if you guys can offer any more help.
version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname Router
boot-start-marker
boot system flash:c2801-ipvoicek9-mz.151-2.T0a.bin
boot-end-marker
no aaa new-model
clock timezone nzst 13 0
dot11 syslog
ip source-route
ip dhcp pool DATA_SCOPE
   network 192.168.200.0 255.255.255.0
   default-router 192.168.200.1
   dns-server 8.8.8.8
ip dhcp pool VOICE_SCOPE
   network 192.168.100.0 255.255.255.0
   default-router 192.168.100.1
   option 150 ip 192.168.2.115
ip dhcp pool MGMT_SCOPE
   network 192.168.1.0 255.255.255.0
   default-router 192.168.1.99
ip cef
ip name-server 4.2.2.2
no ipv6 cef
multilink bundle-name authenticated
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g729r8
codec preference 3 g711ulaw
codec preference 4 ilbc
voice translation-rule 1
rule 1 /^9/ //
voice translation-profile Strip9ToGetOut
translate called 1
voice-card 0
crypto pki token default removal timeout 0
crypto pki trustpoint TP-self-signed-2995340181
enrollment selfsigned
subject-name cn=IOS-Self-Signed-Certificate-2995340181
revocation-check none
crypto pki certificate chain TP-self-signed-2995340181
certificate self-signed 01
3082023E 308201A7 A0030201 02020101 300D0609 2A864886 F70D0101 04050030
31312F30 2D060355 04031326 494F532D 53656C66 2D536967 6E65642D 43657274
69666963 6174652D 32393935 33343031 3831301E 170D3733 30363034 31393534
32305A17 0D323030 31303130 30303030 305A3031 312F302D 06035504 03132649
4F532D53 656C662D 5369676E 65642D43 65727469 66696361 74652D32 39393533
34303138 3130819F 300D0609 2A864886 F70D0101 01050003 818D0030 81890281
8100C34D C8ECBB53 E01373A3 2E286B78 2D23042B 1C8588B1 A7861899 BA1C6860
AE1D7868 2A59E3BC 54D0A457 8FFDE27F C09104E5 C7A429F3 74CD9DA8 4A980366
675CC27C CDB94838 821CC05F 2C0AC2BC D882C132 6CAA1FA6 6DA740E4 562428B1
12B741F1 A50C9246 4CC35EDA DEE1D038 3883BB35 A91ABF8B 483E4160 F5FA4B5A
9A570203 010001A3 66306430 0F060355 1D130101 FF040530 030101FF 30110603
551D1104 0A300882 06526F75 74657230 1F060355 1D230418 30168014 72119640
F3396E1F E4168086 D31D8619 0D8337FF 301D0603 551D0E04 16041472 119640F3
396E1FE4 168086D3 1D86190D 8337FF30 0D06092A 864886F7 0D010104 05000381
81003B5A 29DE3A1E C5AB6092 E8D90650 C80752FC 0AAC93FD C5DE3D69 071B08FA
D4013232 81CA07E7 15F90190 6A3AD6A0 1D05F0F2 13479568 888332A5 F81E2681
7DA44095 4D11CFB7 CA79579A 8D95DE54 7B00173C E2C50573 A310C8C9 1487FEFC
CE35B66E 9EF94CFA 8D6D6DCD ADC78132 2709F198 6DF2F0FA D80CC088 D0C4C7D1 080B
      quit
license udi pid CISCO2801 sn FTX0947W07M
username xxx privilege 15 password 0 xxx
interface FastEthernet0/0
ip address 192.168.3.50 255.255.255.0
duplex auto
speed auto
interface FastEthernet0/1
no ip address
duplex auto
speed auto
interface FastEthernet0/1.2
encapsulation dot1Q 2
ip address 192.168.2.1 255.255.255.0
interface FastEthernet0/1.99
encapsulation dot1Q 99
ip address 192.168.1.99 255.255.255.0
interface FastEthernet0/1.100
description voice_VLAN
encapsulation dot1Q 100
ip address 192.168.100.1 255.255.255.0
interface FastEthernet0/1.200
description data_VLAN
encapsulation dot1Q 200
ip address 192.168.200.1 255.255.255.0
ip forward-protocol nd
ip http server
ip http authentication local
ip http secure-server
ip route 0.0.0.0 0.0.0.0 192.168.3.1
logging esm config
tftp-server flash:/phone/7940-7960/P00307020200.bin alias P00307020200.bin
tftp-server flash:/phone/7940-7960/P00307020200.loads alias P00307020200.loads
tftp-server flash:/phone/7940-7960/P00307020200.sb2 alias P00307020200.sb2
tftp-server flash:/phone/7940-7960/P00307020200.sbn alias P00307020200.sbn
control-plane
mgcp fax t38 ecm
dial-peer voice 1 voip
description local_7_Digit_Calling
translation-profile outgoing Strip9ToGetOut
destination-pattern 9[2-9]......
session protocol sipv2
session target ipv4:203.184.16.2
voice-class codec 1
dial-peer voice 2 voip
description international_calling
translation-profile outgoing Strip9ToGetOut
destination-pattern 900T
session protocol sipv2
session target ipv4:203.184.16.2
voice-class codec 1
dial-peer voice 3 voip
description national_calling
translation-profile outgoing Strip9ToGetOut
destination-pattern 90[34679].......
session protocol sipv2
session target ipv4:203.184.16.2
voice-class codec 1
dial-peer voice 4 voip
translation-profile outgoing Strip9ToGetOut
destination-pattern 90[34679].......
dial-peer voice 5 voip
description CUCM SIP trunk
destination-pattern 2...
session protocol sipv2
session target ipv4:192.168.2.115
voice-class codec 1
sip-ua
authentication username xxxxxxxxxx password xxxxxxxx
060
telephony-service
max-ephones 10
max-dn 20
ip source-address 192.168.1.99 port 2000
load 7960-7940 P00307020200
max-conferences 4 gain -6
transfer-system full-consult
create cnf-files version-stamp Jan 01 2002 00:00:00
ephone-dn 1 dual-line
number 1000
name Lydia Francis
ephone-dn 2 dual-line
number 1001
name Leah Francis
ephone-dn 3 dual-line
number 1002
n
ephone-dn 4 dual-line
number 1003
ephone 1
mac-address C80A.A970.01DE
type CIPC
button 2:2
ephone 2
mac-address 000C.3070.8705
button 1:1 2:15
ephone 3
mac-address 000C.8546.5954
button 1:3 2:15
line con 0
logging synchronous
line aux 0
line vty 0 4
privilege level 15
login local
transport input telnet ssh
scheduler allocate 20000 1000
ntp server 195.43.74.123
end

Route partition on phone line

Hello,
We have cucm version 8 what is the reason for the option route partition under line. The reason I ask this if I have no css for line and device but I have assing a partition under router partition under line nothing happens cannot call no where until i assing a css under line or device. So what the point for route partition. Even if I assign the same partition under route partition to another dn partition I cannot dial that dn in the same partition. I feel the route partition is useless or is it?
Thanks,

Jaime is spot on (+5 J.). It would be a really good idea if you would take a couple of hours to go through the SRND. It is kinda heavy so you may need to iterate through it a couple of times. In regards to Route Partitions (Partitions) and Calling Search Spaces (CSS) try to think of a partition as a collection of patterns and a CSS as a collection of partitions.
The partition of "<none>" is special. It basically says that this pattern (whatever the pattern is: Directory Number, Translation, Route Pattern, etc.) is part of every CSS. Think of "<none>" as default for the time being.
The CSS of "<none>" is exactly what it says "none". As in there is no CSS. As in the only partition in the collection the "<none>" or default partition. This is why you can put a phone number on two phones, not assign a partition or CSS, and make calls. This is not a proper design, even if the calls work.
When you put your Directory Number (DN) in a partition (e.g. MyPhones_PT) you have removed the pattern from the <none> partition. Since the <none> CSS does not visibility of MyPhones_PT, you can't call the phone. This is operating as expected.
There are other rules around patterns/partitions/CSS relationships. No sense in digging into that here. However, this relationship is far from useless. It is core to the whole digit analysis process.
HTH
-Bill (@ucguerrilla)

Route Pattern Discard Digit

Hello,
I have a all route pattern with a discard digit "9" and the national calling starting from "2" and GSM from "9",
i hear a outside dialtone whenever i press a second digit "9" or digit "2", is it possible to get the outside dial on pressing first digit whcih is discard digit "9"
I think it is not possible but i want to confirm from youll experts,

There are no hints, you really need to check EVERYTHING. DNs, meetme, call park, anything that has a pattern/DN configured. Use the route plan report.
And Suresh is falling in one of the most common mistakes when this happens, "it must be something with 9..."/"it must be a RP/TP", sometimes it might be, many other it won't.
You need to check for ANYTHING that has wildcards that might match 9, as well as anything that start with 9. And not limit yourself to just RPs and TPs.
I can tell you back when I was in TAC, as soon as I asked them if they had configured anything else other than a RP/TP with 9, they often said "Oh yeah, I added a DN/CTI/meet-me, etc with 9, could that be the issue??"
HTH
java
if this helps, please rate
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Old AC hunt groups vs. post CCM4.X line groups

This is old gripe I have with CCM 4.X and I want to know what others think about it.
From very early versions CCM has had the AC hunt groups that had the neat feature of being able to specify AC usernames and line #s as hunt group members as well as DNs as hunt group members. You could mix and match for robust scenarios.
Starting with version 4.0 line group/hunt list/hunt pilot was introduced. Also the "busy trigger" for DNs was introduced.
The new LG/HL/HP model introduced a much better feature set for hunting. It also observes the "busy trigger" such that if a person is on a single phone call and his/her DN is configured with a busy trigger of 2, the call will still hunt to that user.
Problem 1: The old AC hunt group does not observe the busy trigger. So if I have a DN with a busy trigger of 6 for a switchboard person, the AC based hunt group will consider that line busy and hunt past it with only 1 call on the line.
Problem 2:The new model of LG/HL/HP DOES observes the busy trigger but does not offer the flexability of adding AC user and line # as a line group member.
Conclusion: Each model has features the other needs. This should be reconciled.
Thoughts?

Same thing happened to me with a cataylst driver on my laptop. Hangs forever. I had just updated my system about a day before I applied this update, and the kernel upgrade definitely wasn't the culprit. I get a console login on my laptop, but whenever I startx it'll just hang forever and not respond to any input other than holding down the power button. Everything was working fine with the kernel, but then the last update, which included
*libxcb
*xcb-proto
*xcb-util
*mlocate
*libx11
*xorg-server
*xf86-input-evdev
*flac
*hal-info
Shared between my two systems (one of which is my desktop, with a nvidia graphics chip and the nvidia proprietary drivers).
I tried downgrading x-server and all the libs that were related to X, but that only broke dbus.
I'm afraid to reboot my desktop now...
Last edited by Subject-17 (2009-08-02 09:01:18)

How to create 20000 route pattern at the same time?

I would like to create a partition which contains 20000 different number which a list of ip phone are going to be able to call.
It would be difficult for me of create route pattern one by one. Please How may I proceed?
Thanks in advance.
Best Regards,

If I am understanding you correctly you're trying to allow a subset of the phones in your organization to dial numbers other phones cannot.
Assuming I understand you correctly, you would be better off to go about this in the opposite direction. Current SRND recommendations indicate that the Device CSS is capable of reaching all destinations while the Line CSS imposes any dialing restrictions. It may very well be that you need to create 20,000 route or translation patterns; however, the advise is to place these in a partition that is listed on the Directory Number [Line] CSS and blocks the destination from those phones that should not be able to reach it. This becomes especially useful with other features such as Extension Mobility as the unique list of restrictions follow that DN regardless of the device it is being used on.
As for the "how" part: Current versions of CUCM include an Import/Export menu within BAT. Export the route patterns you have in the cluster now so that you have a properly formatted CSV and accompanying header text file. Replace the rows of the existing records with what you want to add to the database. You can use tools such as 7-Zip to extract and rebuild the TAR file.
Lastly, be mindful of the capacity limits of the database:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/8x/callpros.html#wp1156290
A Unified CM cluster with a very large dial plan containing many gateways, route patterns, translation patterns, and partitions, can take an extended amount of time to initialize when the Cisco CallManager Service is first started. If the system does not initialize within the default time, you can modify the system initialization timer (a Unified CM service parameter) to allow additional time for the configuration to initialize. For details on the system initialization time, refer to the online help for Service Parameters in Unified CM Administration.
You should work with your Cisco Partner who has access to sizing tools before doing this. These tools can calculate the impact that the additional route patterns will have on the cluster.
For Cisco partners and employees, the tools are available at the following locations:•Cisco Unified Communications Sizing Tool is available athttp://tools.cisco.com/cucst•Cisco Unified CM Capacity Tool is available athttp://www.cisco.com/go/cucmct

Detecting a Route Pattern using JTAPI

Hello,
We are trying to develop a JTAPI application that can detect which route pattern is used when a specific Directory Number dials a number. We are using CallManager 7.0
For example: Extension 1001 Dials a land line number 555-1234. I want to detect the name of the route pattern that extension 1001 used to place the call.
I need this information while the call is placed, so CDR is not an option. Thanks

I think the OP is thinking of real-time CTI monitoring. Unfortunately, there's no way to figure this out then and there.. the number you see in (J)TAPI is already translated. Perhaps using the originally dialed number and if there's no translation of the calling number you could check available patterns in the SQL DB, but it would be one heck of a job to get it done seeing that there can be multiple patterns that can apply (in CCM7 you could have transformation patterns, translation patterns and route patterns). Internally, I'm sure CCM has a way to do all this since it's what is needed to route calls, but that functionality isn't exposed.
Now I guess it comes down to how much you need to know this information - is it worth spending considerable effort to replicate what is basically a dialed number analyzer or not?

Adding a Route Pattern to a Line Group?

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