Archive for the ‘Cisco General’ Category

Show isdn service

Posted: September 18, 2006 by sankar in Cisco General

To find out which channels are in service, which are out of service, use command,

sh isdn service. Here is a sample output from a Cisco IOS router.

 BR1#sh isdn service
PRI Channel Statistics:

%Q.931 is backhauled to CCM MANAGER 0×0003 on DSL 0. Layer 3 output may not appl
y

ISDN Se0/0/1:23, Channel [1-24]
  Configured Isdn Interface (dsl) 0
   Channel State (0=Idle 1=Proposed 2=Busy 3=Reserved 4=Restart 5=Maint_Pend)
    Channel :  1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4
    State   :  0 0 0 3 3 3 3 3 3 3 3 3 3 3 3 3 3 3 3 3 3 3 3 3
   Service State (0=Inservice 1=Maint 2=Outofservice 8=MaintPend 9=OOSPend)
    Channel :  1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4
    State   :  0 0 0 2 2 2 2 2 2 2 2 2 2 2 2 2 2 2 2 2 2 2 2 2
BR1#

How to hide users in Corporate Directory?

Posted: September 18, 2006 by sankar in Cisco General

1. Create a file by the name hideuser.ldif (can be any name) in Notepad

2. Lets take the example of hiding ac user for Attendant Console. Put the following contents in the file and save it.

dn: cn=ac, ou=users, o=cisco.com
changeType:modify
replace:Description
Description:CiscoPrivateUser

3. Go to command line and run the following command.

ldapmodify -h <server name> -p 8404 -D “cn=Directory
Manager,o=cisco.com” -w <DCDAdmin Password> -c -f hideuser.ldif

Gatekeeper – Viazone behaviour

Posted: September 11, 2006 by cciestudy in Cisco General, Gatekeeper

How to trigger via zone behaviour?

Local zones – always enable outvia on the destination zone

Remote zones – use invia or outvia on the remote zone

Via-zone gatekeepers differ from legacy gatekeepers in how LRQ and ARQ messages are used for call routing. Using via-zone gatekeepers will maintain normal clusters and functionality. Legacy gatekeepers examine incoming LRQs based on the called number, and more specifically the dialedDigits field in the destinationInfo portion of the LRQ. Via-zone gatekeepers look at the origination point of the LRQ before looking at the called number. If an LRQ comes from a gatekeeper listed in the via-zone gatekeeper’s remote zone configurations, the gatekeeper checks to see that the zone remote configuration contains an invia or outvia keyword. If the configuration contains these keywords, the gatekeeper uses the new via-zone behavior; if not, it uses legacy behavior.

For ARQ messages, the gatekeeper determines if an outvia keyword is configured on the destination zone. If the outvia keyword is configured, and the zone named with the outvia keyword is local to the gatekeeper, the call is directed to a Cisco Multiservice IP-to-IP Gateway in that zone by returning an ACF pointing to the Cisco Multiservice IP-to-IP Gateway. If the zone named with the outvia keyword is remote, the gatekeeper sends a location request to the outvia gatekeeper rather than the remote zone gatekeeper. The invia keyword is not used in processing the ARQ.

CCM Security – MGCP

Posted: September 7, 2006 by cciestudy in Cisco General, CM4.1 Features

crypto isakmp policy 1
encr 3des
authentication pre-share
group 2
lifetime 10800
crypto isakmp key cisco address 10.1.1.1
crypto isakmp key cisco address 10.1.1.2

!
!
crypto ipsec transform-set CM esp-3des esp-sha-hmac
mode transport
!
crypto map CM 1 ipsec-isakmp
set peer 10.1.1.1
set transform-set CM
match address 101
crypto map CM 2 ipsec-isakmp
set peer 10.1.1.2
set transform-set CM
match address 102

!
access-list 101 permit ip host 10.2.2.2 host 10.1.1.1

access-list 102 permit ip host 10.2.2.2 host 10.1.1.2
interface Serial0/0.101 point-to-point
ip address 10.2.2.2 255.255.255.0
frame-relay interface-dlci 101
crypto map CM

voice translation-rule 1
rule 1 /\(^[2-9]………$\)/ /91\1/ — for national calls
rule 2 /\(.*\)/ /9011\1/ — for international calls

!
!
voice translation-profile ANI
translate calling 1

a. setup meetme conference, conf bridge as necessary. Let meetme number be 1900

b. setup a route point for calls from external and internal callers as the dial-in bridge number. Let this be  1800. Cfwdall this CTI route point to voice mail.

c. Setup a Call handler meetme in Unity. Under transfer settings, tell it to ring subscriber at extn 1900. Use Supervise transfer and Check Ask Caller’s name option.

d. Create a call routing rule under forwarded calls in Unity, and set teh Forwarding station to be 1800 (CTIRP). Attempt a transfer to callhandler meetme that you just created in step c.

Unity will ask the caller his name and transfer him to the conference bridge. Conference bridge members who already have joined the conference will hear, “Call from <callername>”

Allow caller input – Unity

Posted: August 8, 2006 by sankar in Cisco General, Unity

Under caller input page, lock a key (0 through 9, * #) to a particular action if you want Unity to perform that action as soon as you press that key. The “Allow callers to dial an extension durin greeting” has no effect if the keys are locked. To allow callers to dial an extension during the greeting, dont lock the key to a particular action

 If you have any of your keys set to Ignore key, the greeting that is played is usually System greeting “I do not recognize that as a valid selection”. This is the Error greeting. Expose the Error greeting using Advanced Settings tool from Tools Depot and you can re-record the Error greeting to a message of your choice.

Unity – Tips

Posted: August 2, 2006 by sankar in Cisco General, Unity

The following tip may be utilized when you are asked to forward a call from CM / CME to Voicemail and have the caller leave a message. Then the message should be delivered to multiple user inboxes and also  light up mwi on the appropriate phones.

a. Setup CM/CME to forward calls to VM. (say  a route point with number  1005)

b. Create a PDL in Unity with members as the individual subscribers who needs to receive the message.

c. Create a Callhandler CH with number 1005. Go to Messages – > Message recipient and select this PDL you created in step 2.

IPIPGW Example

Posted: July 31, 2006 by sankar in Cisco General

We setup a IPIPGW with 3 gatekeepers and 2 callmanagers.

IPPhone1—Callmanager 1—–GK1———–(IPIPGW/VIAZONEGK)———–GK2———Callmanager2—IPPhone2

GK1 configs:

int lo 0

ip address 172.12.100.1 255.255.255.0 

gatekeeper
 zone local HQ-RTR ipexpert.com 172.12.100.1
 zone remote VIAZONE ipexpert.com 10.12.200.2 1719
 zone prefix VIAZONE 20*
 gw-type-prefix 1#* default-technology
 no shutdown
!

GK2 Configs:

gatekeeper
 zone local LT-RTR ipexpert.com 192.168.10.46
 zone remote VIAZONE ipexpert.com 10.12.200.2 1719
 zone prefix VIAZONE 40*
 gw-type-prefix 1#* default-technology
 no shutdown

IPIPGW config:

 voice service voip
 allow-connections h323 to h323

dial-peer voice 1010 voip
 destination-pattern 20..
 session target ras
 incoming called-number 40..
 dtmf-relay h245-alphanumeric
 codec transparent
!
dial-peer voice 1020 voip
 destination-pattern 40..
 session target ras
 incoming called-number 20..
 dtmf-relay h245-alphanumeric
 codec transparent
!

int fa0/0

ip route-cache same-interface

h323-gateway voip interface
h323-gateway voip id VIAZONE ipaddr 10.12.200.2 1719
h323-gateway voip h323-id PSTNSw
h323-gateway voip tech-prefix 1#

VIAZONE GK Configs:

gatekeeper
 zone local VIAZONE ipexpert.com 10.12.200.2
 zone remote HQ-RTR ipexpert.com 172.12.100.1 1719 invia VIAZONE outvia VIAZONE
 zone remote LT-RTR ipexpert.com 192.168.10.46 1719 invia VIAZONE outvia VIAZON
E
 zone prefix LT-RTR 20*
 zone prefix HQ-RTR 40*
 no shutdown

IPCC High Availability

Posted: July 27, 2006 by sankar in Cisco General, IPCC Express

When you install two IPCC servers the following considerations hold true:

 a. The first server you activate (CRS Engine and/or Datastore services) will become the active CRS box.

 b. The second server you activate (CRS Engine) will become the standby box for the active CRS engine.

 c. The second server you activate (Datastore engine) will become the standby box for the active Datastore engine. (Agent, Historical etc).

 d. When you create a jtapi user in the active box (first server in your cluster), the user name that is created in Callmanager is “jtapi_1″ the one signifying that its the jtapi user for the first CRS server. When you install the second CRS server, it will create another jtapi user named “jtapi_2″.

 e. The second server that you install will use the administrator username / password of the first box during the initial server setup. You should not use Administrator / ciscocisco password.

 f. If the active box has a jtapi group with 20 CTI ports ranging from 3201 to 3220, when you initialize the standby box, it will automatically create 20 more CTI ports from 3221 to 3240. 

 g. JTAPI groups that are created in IPCC coressponds to CTI ports created in Callmanager. The ports in Callmanager that belong to the active CRS box will have a description (if you leave it at defaults) as JTAPI Group #0 – 1. The ports that belong to the standby CRS box will have a description JTAPI Group #0 – 2

Lets look at an example where you want to block international calls outside of business hours 8 am to 5 pm.

Remember the time you should configure is in 24 hours format. So 5 pm is 17:00.

after-hours block pattern 1 9011
after-hours day Sun 17:01 08:59
after-hours day Mon 17:00 08:59
after-hours day Tue 17:00 08:59
after-hours day Wed 17:00 08:59
after-hours day Thu 17:00 08:59
after-hours day Fri 17:00 17:00
after-hours day Sat 17:01 17:00

If you specify the time as 08:59, it includes the 60 seconds from 08:59:00 to 08:59:59. So calls will be blocked until 08:59:59. If you specify the time as 17:00, calls will be blocked till until 17:00:59.

VT advantage – unlimited

Posted: July 22, 2006 by sankar in Cisco General

VTA uses CAST protocol. Uses 4224 TCP port. This protocol helps in discovering remote VTA capable endpoints, communicates with IP phone as well as with Callmanager. To ensure proper QOS is given to CAST, on the phone port, create a service policy to mark TCP 4224 with DSCP CS3.

VTA uses two types of codecs.

a. H263 – min of 128kbps to a max of 1.5 Mbps per call

b. Wideband – 7.0 mbps per call

MGCP Call Preservation

Posted: July 19, 2006 by cciestudy in Cisco General, Gateways, IOS Gateways

MGCP PRI backhaul does not support call preservation when transitioned from Callmanager to SRST and vice versa.

H323/MGCP – Caller ID Display

Posted: July 19, 2006 by cciestudy in Cisco General, Gateways, IOS Gateways

H323 does not support Facility IE. H323 supports only Display IE

MGCP supports both Display and Facility IE

Frame-relay fragment

Posted: July 10, 2006 by sankar in Cisco General, QOS, WAN

Frame-relay fragment is typically not needed for link speeds above 768 kbps.

If CIR is 256k and link speed is 480kbps, frame-relay fragment should be based off of PVC CIR ie 256 K.

MGCP Fallback to H323

Posted: June 18, 2006 by cciestudy in Cisco General

application
global
service alternate Default

ccm-manager fallback-mgcp

sh ccm-manager
MGCP Domain Name:
Priority Status Host
============================================================
Primary Down 10.101.21.250
First Backup Down 10.102.21.251
Second Backup None

Current active Call Manager: None
Backhaul/Redundant link port: 2428
Failover Interval: 30 seconds
Keepalive Interval: 15 seconds
Last keepalive sent: 19:46:24 UTC Mar 5 1993 (elapsed time: 00:43:26)
Last MGCP traffic time: 19:46:33 UTC Mar 5 1993 (elapsed time: 00:43:17)
Last failover time: 14:36:07 UTC Mar 5 1993 from (10.100.1.50)
Last switchback time: 14:36:13 UTC Mar 5 1993 from (10.100.1.51)
Switchback mode: Graceful
MGCP Fallback mode: Enabled/ON
Last MGCP Fallback start time: 19:46:52 UTC Mar 5 1993
Last MGCP Fallback end time: 19:45:24 UTC Mar 5 1993
MGCP Download Tones: Disabled

Backhaul/Redundant link is down
Configuration Error History:
FAX mode: cisco
To simulate MGCP fallback, do not disable MGCP using the no mgcp command. This will not kick in the H323 fallback mode. Either apply an access list or create a null route for Callmanager server.

Dial-peer matching

Posted: June 17, 2006 by cciestudy in Cisco General

It follows the longest match rule first and then preference. In this example, if you dial 2001, dial-peer 10 will be matched first (longest match + preference 1) and then dial-peer 11 (longest match + preference 2) and then dial-peer 11 (next best match)

dial-peer voice 10 voip
preference 1
destination-pattern 2001
voice-class h323 1
session target ipv4:10.100.1.51
ip qos dscp cs3 signaling
!
dial-peer voice 11 voip
preference 2
destination-pattern 2001
voice-class h323 1
session target ipv4:10.100.1.51
!
dial-peer voice 12 voip
destination-pattern 200.
session target ipv4:10.100.1.50

IOS interpretation of “\”

Posted: June 7, 2006 by cciestudy in Cisco General

IOS maps "\" followed by a number to a character. For ex: "\320" maps to the alphabet "P".

If you need to use this combination, then use the escape key sequence "\134" before the numbers. In the same example, use "\134320"

MLPP

Posted: June 3, 2006 by sankar in Cisco General, CM4.1 Features, MLPP

How to setup basic MLPP ?

a. Take three phones 1025, 1026 and 1027. Let 1025 be the destination phone. 1026 be able to make Exec Override calls. 1027 be able to make Flash Override calls.

b. Set them all up in same MLPP domain 000011 (for ex:)

c. Set 1025 with MLPP indication enabled, Premption – Forceful

d. Set 1026 and 1027 with MLPP indication enabled, Premption – Can be disabled. No need to set them to forceful.

e. Create translation pattern 90.1025 in partition Exec and set the MLPP setting to Executive Override. This TP can be accessed only by 1026. Translate 90.1025 to 1025.

f. Create translation pattern 90.1025 in partition Flash and set the MLPP setting to Flash Override. This TP can be accessed only by 1027. Translate 90.1025 to 1025.

g. Place a flash override call from 1027 to 1025. (Dial 90.1025 from 1027.). You should note a different ring tone on 1025 and also a precedence ringer display.

h. Place a exec override call from 1026 to 1025 (Dial 90.1025 from 1026). You should immediately hear a precedence tone on 1025 and 1027. (Continous tone). Both parties 1025 and 1027 have to independently hangup. The Exec override call from 1026 is offered to 1025. Phone 1025 starts ringing.

i. If 1025 doesnt answer the exec override call from 1026, after 30 seconds, the call between 1025 and 1027 is dropped and the exec override call from 1026 is offered to 1025.

j. A flash override call cannot preempt a exec override call. If 1026 and 1025 are in a exec override call, a flash override call from 1027 cannot preempt the call between 1026 and 1025, as the flash override call has lower precedence. The flash override call will appear as call waiting on 1025. (No call waiting beep is heard).

k. An exec override call cannot preempt another exec override call (same domain), unless the service parameter Exec Override Call Preemptable is set to true. With the default setting (false), if 1026 places an exec override call to 1025, and 1027 places an exec override call to 1025, this call will appear as call waiting on 1025. (No call waiting beep is heard).

l. Do not set Preemption enabled to Forceful and Premption indication to Off of Default (If device pool is also off). This will prevent preemption on the destination device.

m. Only calls originating from devices in same domain can be prempted. If 1026 is set to domain 000011 and 1027 is set to domain 000022, 1026 cannot override 1027's call and vice versa.

n. If both 1026 and 1027 are set to flash override, One flash override call cannot preempt another flash override call. The call appears as call waiting (no beep is heard)

Fax passthrough – unlimited!!

Posted: May 27, 2006 by sankar in Analog, Cisco General, Fax

What is Fax passthrough ?

Fax passthrough encodes fax traffic with in a g711 voice codec and sends it across the voip network as a voice call. The call may use any codec (g711, g729, g723) etc initially and once a 2100 Hz CED tone is detected, the device (ATA for ex:) tells the far end gateway to switch over to G711 using a peer-to-peer message. This message is called a NSE message (Named Signalling Event) with in the RTP stream.

QoS templates for 3550

Posted: May 25, 2006 by sankar in 6500, Cisco General, QOS

1. mls qos
### enables qos globally#######

2. mls qos cos-map 0 8 16 26 34 46 48 56
### maps cos values to dscp values properly######

3. For IP phones ports, apply the following commands

int range fa 0/1 , fa 0/2
#### ip phone ports

flowcontrol receive off ***** important command********
flowcontrol send off *********important command*******

4. Mapping voice bearer traffic in priority queue

int fa0/1

wrr-queue cos-map 4 5
priority-queue out
### if asked to put Voice bearer in priority queue

5. Mapping voice signalling traffic in queue 3

wrr-queue cos-map 3 3

6. Port configuration

interface fa0/1

mls qos trust cos
#### trusts packet cos
mls qos trust device cisco-phone
#### trusts cos only if a phone is attached

switchport priority extend cos 0

### zeros out PC cos values.

7. If asked to modify bandwidth and buffer settings for each queue (only then do the following)

For fastE ports:

mls qos min-reserve 5 170
mls qos min-reserve 6 130
mls qos min-reserve 7 51
mls qos min-reserve 8 34
#### defines min-reserve levels (upto 8 levels may be defined, default buffer size is 100 for all levels)####

int range fa 0 /1, fa 0/2

wrr-queue bandwidth 20 20 60 1
### priority queue doesnt need wrr bandwidth allocation
wrr-queue min-reserve 1 5
wrr-queue min-reserve 2 6
wrr-queue min-reserve 3 7
wrr-queue min-reserve 4 8
### maps min-reserve levels to queues#####

For GigE ports:

int range gi 0/1 , gi 0/2
wrr-queue queue-limit 60 20 20 1
#### defines more buffer space for low priority queue ####
wrr-queue bandwidth 20 20 60 1

8. DSCP maps (optional)

For gig ports there is a dscp map that maps dscp values to thresholds.
Each queue has two thresholds, and by default all dscp values are mapped to threshold 1.
If asked to set voice traffic (may be video) to threshold 2, use command.

wrr-queue dscp-map 2 26 34 46 (this is higher threshold in the queue)

9. Tail Drop or WRED (optional)

For gig ports default drop mechanism is tail drop. Here is how you may modify these thresholds:

wrr-queue threshold 1 80 100
wrr-queue threshold 2 80 100
wrr-queue threshold 3 80 100

### no need to define drop thresholds for queue 4 if its priority queue

To enable WRED and specify thresholds, use following commands:

wrr-queue random-detect max-threshold 1 80 100
wrr-queue random-detect max-threshold 2 80 100
wrr-queue random-detect max-threshold 3 80 100

### WRED and tail drop are mutually exclusive

10. Classification using ACLs.

To classify based on subnet, define standard or extended acl's.

access-list 101 permit ip any any dscp 24

class-map test
match access-group 101

11 .Defining Policer and Remarking traffic
mls qos map policed-dscp-map 26 46 to 0
#### remarks voice control and bearer traffic to dscp 0. (Defined in policer)

mls qos aggregate-policer TestPolicer 256000 8000 exceed-action policed-dscp-transmit
#### defines an aggregate policer with a rate of 256kbps, burst of 8000 bits and remarks dscp for voice and bearer traffic based on above policed-dscp map
class-map match-all Voice
match ip dscp af31 ef

policy-map Voice
class Voice
trust dscp
police aggregate TestPolicer
#### applies aggregate policer to the class.
You cannot define same policer across multiple policy-maps.

int range fa 0/1 , fa0/2
service policy input Voice

Example configs:

1. To define a class-map that remarks traffic:
——————————————
class-map match-all VoiceControl
match ip dscp af31
class-map match-all VoiceBearer
match ip dscp ef

policy-map Voice
class VoiceControl
trust dscp
set ip dscp 40
class VoiceBearer
trust dscp
set ip dscp 24
int range fa 0/1 , fa0/2
service policy input Voice

2 . To perform per-vlan, per-port classification, marking, policing. (may be required on gateway ports which may be a trunk port)
—————————————————————
class-map match-all Voice
match ip dscp af31 ef

class-map match-all VoiceVLAN
match vlan 100 ————– defines which vlan you want to match
match class-map Voice ——- defines all traffic on voice vlan with dscp af31 or ef.
policy-map Voice
class VoiceVLAN
trust dscp
police aggregate TestPolicer

#### applies aggregate policer to vlan 100
You cannot define same policer across multiple policy-maps.

int range fa 0/3
Decription Gateway port
service policy input Voice

3. To perform individual policing on each class:
———————————————-

mls qos map policed-dscp-map 26 46 to 0

class-map match-all Voice
match ip dscp af31 ef

policy-map Voice
class Voice
trust dscp
police 256000 8000 exceed-action policed-dscp-transmit
####This is a individual policer

int range fa 0/1 , fa0/2
service policy input Voice

QOS template for 6500 – Egress

Posted: May 24, 2006 by sankar in 6500, Cisco General, QOS

Mapping packets to a particular queue / threshold

set qos map 2q2t tx 2 1 cos 3 (mandatory)

Optional commands:

set qos wrr 2q2t 5 255 (optional)

The values are absolute based on a scale of 255. To get the values in percent, you need to multiply it by 2.5.10% means 25 and 20% means 50 and so forth.

set qos drop-threshold 2q2t tx queue 1 80 100

OR

set qos wred 1p2q2t tx queue 1 80 100 (both optional)

set qos drop-threshold 2q2t tx queue 2 80 100

OR

set qos wred 1p2q2t tx queue 2 80 100 (both optional)

set qos txq-ratio 2q2t 80 20 (optional)

Intracluster ports:

SQL TCP 1433, 3372
SMB TCP 445
ICCS – TCP 8002, 8003

Windows common ports:

DHCP (if running) – UDP 67,68
TFTP – UDP 69

Signalling ports:

skinny TCP 2000, (from phone to CCM)
secure skinny TCP 2443 (from phone to ccm)
tftp -udp 69 and ephemeral ports
capf – tcp 3804 (phone to capf/ccm)
RTP – udp 16384 – 32768 (CM uses only 24576 – 32767)
VTAdvantage (TCP 4224) – PC to the phone

Callmanager to gateway

tcp port 11000 – 11999
tcp port 1024 – 4999
tcp port 1720 bothways (h225)

tcp port 2000 (skinny gateway to ccm)
udp 2427/ tcp 2428 (mgcp gateway control/backhaul)
tcp and udp port 5060 (SIP gtway and ICT)
udp port 16384-32767 – RTP between the gtwy and cm

Callmanager to gatekeeper

tcp port 1718 (ccm to gk)
udp port 1719 (gk to ccm)
Callmanager to gateway (for encryption)

ESP – 50 (ESP protocol itself)
IKE – 500 UDP

Callmanager to Secure SRST router  

SRST Certificate Provider Port: 2445 

When CTI Application use is checked, the user can control the devices associated to that user.

When CTI Super Provider is checked, the user can control all CTI ports, CTI Route Points and IP Phones.

For CUE you need to define mwi on and off as two different ephone-dn's. You cannot enable the mwi on-off using the same ephone-dn like you do with Unity.

Set up the mwi number following by dots equal to the extension length

ephone-dn 10

number 1599….

mwi on

ephone-dn 11

number 1598….

mwi off

The default ccn application for mwi uses the extensions 8000 for ON and 8001 for OFF. You can change this from CLI

ccn application ciscomwiapplication
description "ciscomwiapplication"
enabled
maxsessions 4
script "setmwi.aef"
parameter "strMWI_OFF_DN" "1598"
parameter "strMWI_ON_DN" "1599"
parameter "CallControlGroupID" "0"
end application

1. When the extension appears on line 1, red lamp MWI and envelope is used.

2. When the extension appears on any line other than line 1, only envelope is used.

After changing the IP Address in the interface service-module, calls to voicemail are getting answered, but callers does not hear anything.

The show call active voice brief shows that the call leg to unity express is getting connected to the old IP address.

Resetting the CUE fixed the issue.

Unity Live Reply – Gotchas

Posted: April 26, 2006 by sankar in Cisco General, Unity

If subscriber has Standard conversation, he needs to press 4-4 after the message has been played (including message footer).

If subscriber has Optional conversation1, he needs to press 8-8 after the message has been played (including message footer)

When subscriber (called) presses 4-4 or 8-8, Unity checks the original subscriber's (who left the voice message) transfer settings and sends the call based on his transfer settings. You may have to change the setting from "Send to greeting" To "Ring subscriber's extension" or "Ring subscriber at XXXX".

CME – Caller name display

Posted: April 13, 2006 by sankar in Callmanager Express, Cisco General

Prime line – Caller name display while ringing and when connected

Other lines – Caller name display only when connected

To enable Caller name display,

telephony-service

  service dnis overlay  (For overlaid phones)

  service dnis dir-lookup (Look up names from directory-entry command)

Soft phone local time display

Posted: April 13, 2006 by sankar in Cisco General

Soft phones derive time from the local PC and not from Callmanager / CME.