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
Similar Messages
-
Route Pattern/Filter Issues
Hello - I had a question that I was hoping someone could help me with. I recently had an issue on CallManager 4.1.3sr6a. I have 10 sites, each using their own set of 6 9.@ route patterns for outgoing calls. Although the patterns are distinct, they all use the same route filters. Today, all of the sites began matching on the route pattern with the seven digit filter rather than matching on the long distance filter when a 91XXXXXXXXXX number was dialed (we tested with a number of numbers in different area codes). I wasn't able to input more digits after the 7th and DNA confirmed that it matched on the seven digit filter. The filters are as follows:
Long Distance =(AREA-CODE EXISTS AND LONG-DISTANCE-DIRECT-DIAL EXISTS)
SevenDigit = (LOCAL-AREA-CODE DOES-NOT-EXIST AND INTERNATIONAL-DIRECT-DIAL DOES-NOT-EXIST AND AREA-CODE DOES-NOT-EXIST AND SERVICE DOES-NOT-EXIST).
I was able to resolve the issue by going to each site, removing the long distance 9.@ pattern and simply readding it (no changes were made to configuration).
It was like it just stopped seeing all patterns using the long distance filter.
I'm wondering if anyone is aware of a bug that may have caused this.
Thanks very much!For detailed descriptions and examples, refer to the Understanding Route Pattern Wildcards and Special Characters section of Understanding Route Plans.
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/3_0_9/p1rtundr.html -
9@ Route Pattern Matched Issues
Unfortunately I have to deal with a lot of 9@ route patterns in our deployment. I understand weird things happen when 9@ is used, but even this one is boggling my mind. So I was hoping someone could help me understand why it's doing what it's doing.
I have a CSS with a collection of partitions. I'll call the 3 I'm interested in the following: One-PT, Two-PT, Three-PT.
One-PT has a route pattern of 9@ with the Local filter applied going to Gateway 1.
Two-PT has a route pattern of 9@ with the Local filter applied going to Gateway 2.
Three-PT has a route pattern of 9.XXXXXXXXXX with no filter applied (those are 10 Xs) going to Gateway 3.
My phone is assigned to the CSS with these 3 partitions. When I dial 9 981 xxx xxxx DNA says that 9@ from One-PT is always matched. If I remove One-PT from the CSS, then 9@ in Two-PT is matched. Only if I remove those 2 partitions does Three-PT get matched.
Now, as I said above I understand 9@ can introduce weird routing issues, but I thought that the route pattern with 9 and 10 Xs would be more specific and it would be matched. Obviously I was wrong, but I'm trying to understand why I was wrong. Is this because the 10 digit number dialed matches the NANP and the Local filter matches a NANP area code? Thus it's the more exact match?
Thanks!Hi,
As per the following link
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/5_0_4/ccmsys/ccmsys/a03rp.html#wp1050657
"Using the @ Wildcard Character in Route Patterns
Using the @ wildcard character in a route pattern provides a single route pattern to match all NANP numbers, and requires additional consideration.
The number 92578912 matches both of the following route patterns: 9.@ and 9.XXXXXXX. Even though both these route patterns seem to equally match the address, the 9.@ route pattern actually provides the closest match. The @ wildcard character encompasses many different route patterns, and one of those route patterns is [2-9][02-9]XXXXX. Because the number 2578912 more closely matches [2-9][02-9]XXXXX than it does XXXXXXX, the 9.@ route pattern provides the closest match for routing."
Also, check the following post
https://supportforums.cisco.com/discussion/10698966/9-route-pattern
HTH
Manish -
Route Pattern / Extension Issue
Hello! I'm currently having an issue with a new block of extensions in Call Manager 7.1.5.
We recently ordered a new block of DIDs. The telco sends us the last 4 digits. The last 4 are 9XXX. When dialing these extensions internally, there is an 8-10 second pause of dead air and then it eventually rings through to the phone.
All of my Route Patterns for each site start with a " 9. "
My guess is that when dialing the extension, Call Manager tries to match it to a Route Pattern since it starts with a "9" followed by 3 digits. After it cycles through the Route Patterns and doesn't match it, it eventually routes it to the phone. Is this what's happening? And if so, any thoughts to how I alleviate this issue?
Any recommendations are appreciated.
Thanks!Hi James,
It appears to be correct. Due to overlapping pattern, the call manager would wait for an inter-digit timeout to expire and route the calls to the correct pattern.
As a workaround, you can try to lower the T302 Timer(inter-digit timeout) in call manager to 4-5 seconds (Service Parameters -> Cisco Call Manager -> T302 Timer).
As this issue is experienced with overlapping pattern, the best way to resolve it to design your dial plan efficient so that there is no overlapping.
HTH,
Jagpreet Singh Barmi -
CUCM 9.1.2
I am trying to get a better understanding of what happens when you add a Route Pattern. When you click save a warning pops up saying
"Any update to this Route Pattern automatically resets the associated gateway or route list."
This pattern is tied to a route list which is then tied to the gateway. What exactly is reset? Will this effect any current calls? How long does this reset take?
Thanks,
AlexIf your route patterns point to route list nothing will be affected, if you point your RP directly to GW or trunk, depending on what is connected to the GW (trunk type) calls may be interrupted. Since you should never point RPs to trunk/GW directly, this is atypically not an issue.
HTH,
Chris -
CCM 4.1 route list issue
We are trying to transfer a calls to an after-hours call center. Unfortunately we are getting a fast busy when we attempt to forward. When I run 'debug isdn q931' I can see the reason we are getting a fast busy is the number prefix is incorrect. It's trying to forward the number with our 7 digit prefix in front of the calling number. So our 7 digit prefix is 989-2525 and the calling party for example is 909-656-2222, so when it trys to forward the call it shows the calling party as '98925259096562222' which is why we get a fast buy. The 7 digit prefix 989-2525 is defined in the Route List Configuration, which has to be set otherwise I'm unable to call inbound. What's adding the prefix? Sometimes I'll see the calling party as '98925259892525' and the call does go through.
Heya Carlo!
When you say check if they are the same configured, you are referring to them both having the same selected partitions? I'm able to call the call center from my phone with the correct calling party number.
I'm using a translation pattern where I have a CTI port configured in the Called Party Transform Mask. Under Called Party Transformations I have:
Discard Digits: NONE
Called Party Transform Mask: 2600 <---- this is the CTI port I have configured
Prefix Digits (Outgoing Calls): BLANK
Calling Party Transformation settings are blank.
As for the Route Pattern, I have one configured using the route pattern* 9.1909[2-9]XXXXXX where the discard digits is set for PreDot. I'm actually thinking the issue with discarding digits incorrectly is in the route pattern, still reading up on that part now. What do you think with this info provided? -
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
www.cisco.com/go/pdihelpdesk -
I have an Airport Extreme as my router and am using time capsule to extend the network in my new house. My ISP is only providing me 4-5 ip addresses and wants me to set up my router to issue out new ip addresses for all my devices.How do I fix this?Help.
They said I need to change my settings to NAT settings. I haven't been able to figure out or find anything. I have also spoken to Apple Support on the phone for hours without being able to figure out how to do this ( i don't think he knew much either lol.) Please help me because I've got about 15-20 devices in my house that require to be connected to the internet and this is just making things ridiculously slow and painful for me.
Thanks!It is on DHCP & NAT under router mode yet my isp is still the one issuing ip addresses to my devices instead of the router issuing them
-
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.
ThanksHi 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
www.cisco.com/go/pdihelpdesk -
Route Pattern CSV File Removes "0" from Called Party Prefix Digits Field
I want to upload over 350 route patterns using BAT Tool in CUCM 9.1. All patterns must have their Called Party Prefix Digits (Outgoing Calls) Field containing 10 numbers with the number "0" at the begining. The Problem is the CSV file removes the leading "0" form the digits. I tried to make the cell type in Excel as "Text" and it worked and the "0" is kept normally, but when I save and close the file then open it again, the cell is defaulted to "General" type and the "0" is disappeared again! Changing the CSV file format to any other one and uploading it to CUCM system generates an error stating that the file format is not supported.
Attached is a sample entry of the CSV file. I want to preserve the whole number "0541234567" in "PREFIX_DIGITS_CALLED_PARTY" Field.
Anyone can help me how to upload this big number of route patterns while preserving the number "0" at the begining?Make sure you change the csv file when using Excel to "Text" on the cell where the string starts with 0, otherwise Excel assumes this is a number and strips it.
HTH, please rate all useful posts!
Chris -
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 -
CUCM to CVP calls. CTI-RP vs Route Pattern
CVP 9 or above
CUCM 9 or above
Requirement:
1. Consultive Warm Transfer - The agents to be able to transfer calls to a a different department by dialing an internal number and wait in the queue until answered.
2. Internal - Back-office people to dial internal IT-Helpdesk or HR
I see the above call flows as same, i.e. a Call Originating from CUCM to CVP .... correct me please?
I have tested both and they both work exactly the same way, i.e. using a CTI-RP associated to PGUSER, ICM answers it sends correlation id to CUCM and CUCM sends this to CVP ...AND... using a simple route patters instead point to CUCM-CVP SIP trunk. Functionally they behave same way - ICM/CVP answers and queues call until answered.
But the documentation confuses me, below snippet from CVP Config Guide
"... Calls Originated by Unified CM
Internal Help Desk calls: For these calls, the Unified Communication Manager (CM) phone user calls a CTI Route Point
Consultative Warm Transfer: For these calls, a Unified CM agent places the caller on hold and dials in to Unified ICME to reach a second agent .... "
And then on the same doc, there a Note
(*1) Note For warm transfers, the call from Agent 1 to Agent 2 does not typically use a SIP Trunk, but you must configure the CTI Route Point for that dialed number on the Unified CM server and associate that number with your peripheral gateway user (PGUSER)
(*2) And then again on the same doc under 'Unified ICME Warm Consult Transfer/Conference to Unified CVP' chapter/section it mentiones doing this using a Route Patter 'Create a route pattern and assign the route list to the route pattern'
So the confusion is
1. Why treat these call flows as Internal and Warm Transfer - they are calls from CUCM to CVP for the same end result - queue the call and transfer to an agent?
2. Route pattern or CTI-RP, what diff it makes? They both behave the same way, so is there a diff from reporting point of view that a call to CTI-RP are treated as Transferred rather than new calls or what?
3. Also if you compare (*1) & (*2) above, they both talk about Warm Transfer and *1 says 'must use CTI-RP' and *2 says use a Route Pattern?
Please assist.
Thanks & Regards,
KartikKartik,
The Route Pattern that is mentioned is used for connecting a call leg through CVP to a local VXML Gateway for media playback. The CTI Route Point is entirely different from the Route Pattern/Route List setup. Here's the basic call flow:
Internal caller dials DN
DN hits CTI RP in CUCM. CTI RP sends call to ICM.
ICM matches DN to Call Type to Script, executes Script.
At some point, Script has either Send To VRU or a Run Ext. Script node.
ICM sends CUCM Network VRU label back to CUCM.
CUCM routes label using Route Pattern and Route List. The CSS of the internal caller determines how this is modified, i.e. which prefix digits to add for determining VXML gateway to route to.
Call is sent to CVP through SIP trunk
CVP receives call, tells ICM it has the call.
CVP starts new call leg to VXML gateway with digit string to match bootstrap dial-peer.
VXML Gateway receives call, initiates bootstrap TCL and VXML magic.
Yes, this is basically the same call flow for a fresh internal call to a queue, or an internal warm transfer to a queue. The CTI Route Point is needed in both cases. The Route Pattern/Route List combo is needed in both cases.
When you start looking at reporting, yes of course the two call scenarios are different. One is a transfer, the other isn't. The transferred call will have a more complex call history if you look at it in the TCDR.
From the standpoint of call legs, you will use less legs if you do a direct (one-step) transfer instead of a warm transfer. It is also simpler to maintain the call context in that case. In a warm transfer scenario, the agent is putting a caller on hold, then starting a new call, and joining the two calls together. The new call is coming from the agent, not the original caller. In a direct transfer, CVP just takes back the original call, potentially does more queuing, then sends the original caller to a new agent target.
-Jameson -
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
ThanksHello,
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. -
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
shameerYes, 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
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