9971 SIP Phones fail to register to CME

Other than the below roughly make sure your SIP is bound to the same interface as your "source address" under "voice register global"... 

voice service voip

sip

bind control source-interface Loopback0

bind media source-interface Loopback0

** above the sip is bound to my interface Loopback0 and below the voice register global has the source-address mentioned as my voice vlan interface ip..... this was what prevented the 9971 phone from registering to CME, once I changed the sip bind to gi0/0.11 (same as the source-address... the phones registered)

voice service voip

    allow-connections h323 to h323

    allow-connections h323 to sip

    allow-connections sip to h323

    allow-connections sip to sip

sip

     bind control source-interface gi0/0.11 (was earlier set to Loopback0)

     bind media source-interface gi0/0.11 (was earlier set to Loopback0)

     registrar server expires max 300 min 60

voice register global

    max-dn 25

    max-pool 10

    mode cme

    source-address 142.102.66.254 port 5060

    load 9971 sip9971.9-4-1SR1-2

    authenticate register

    tftp-path flash:

    create profile

voice register dn  1

    number 3001

voice register pool  1

    id mac 5ca4.8a64.5e87

    type 9971

    number 1 dn 1

    username cisco password cisco

tftp-server flash:dkern9971.100609R2-9-4-1SR1-2.sebn

tftp-server flash:kern9971.9-4-1SR1-2.sebn

tftp-server flash:rootfs9971.9-4-1SR1-2.sebn

tftp-server flash:sboot9971.031610R1-9-4-1SR1-2.sebn

tftp-server flash:sip9971.9-4-1SR1-2.loads

tftp-server flash:skern9971.022809R2-9-4-1SR1-2.sebn

Calls from SIP 9971 phones to internal extensions takes too long to dial out (roughly 15 seconds)

adjust the t302 timer in CUCM from the Service Parameters (the lowest you can go as of CUCM version 9.x) is 3000 ms or 3 seconds.

Problem with AAR Calls - Remote H323 Gateway is not able to route the call to the CUCM

When AAR is triggered in CUCM (I mean once the phone says "Not Enough Bandwidth. Rerouting"... and attempts to dial the External Phone Number Mask entry of the destination phone, the call once routing over the PSTN with the E.164 reaches the far end router over PSTN but the return VOIP Dial Peer to CUCM does not seem to reach CUCM.

Fix:

"Bind your H323 traffic to a Voice Interface on the Remote Voice Gateway that is returning the call to CUCM and reregister the remote gateway in CUCM as H.323.



Also do the following (although the above H323 traffic binding fixed the problem)

voice service voip on both gateways and allowed-connections h to h

Also if it's a 15.0 IOS then either say "no IP Address Trusted Authenticate" or add the ip address of the source to the "IP Addresss Trusted List" under "voice service voip"

Overall UCS Connectivity between the VM's to the FI's

The above pic on shows one Blade Server on a Chassis connected to one side of the Fabric... the other side of the Fabric would have the same logic.

D-channel interface does not appear in the " sh ip int b"

make sure you've got enough DSP's in the router...

"An error has occurred while trying to resynchronized the Cisco JTAPI client ,Please try Cisco JTAPI Resync in Cisco Unified CCX Administration again,keep getting this and the uc telephony subsystem service still shows partially up.any clues?

josh finke had posted a resolution to this that I believe would solve that issue, i've posted the link below.....

you'll as well need to make sure you have ur Unified CM Telephony Subsystem configured with the group and trigger and all and restart the unified ccx engine, and after u have configured the UCM Tele Subsystem, login in to CUCM and see if your CTI ports and CTI Route Point are created and registered and also check if your jtapi user that was created during the integration has the cti ports and route point in the Controlled Devices section, just to make sure that uccx and cucm are talking to each other smoothly...

https://learningnetwork.cisco.com/thread/10881

Steps from his post:

1. Create a folder named WINNT in c:

1. Copy c:WINDOWSJava to WINNT

1. Go to Cisco JTAPI Resync in the Unified CM Telephony Subsystem to install the Cisco JTAPI Client

4. Go to System>Control Center and restart Cisco Unified CCX Engine

Integrating CUCM with CUE - JTAPI Login Failed. Error while determining CCM Version-1

..was testing the CUE (Ver. 7.0.6) integration with CUCM ver 7.0.1, all went well however when I key in the jtapi username and password (created in CUCM with control of the CTI_RP and the CTI_Ports along with the Standard CTI Enabled Role) in the CUE GUI (after entering the cucm ip and web username and password obviously), it takes like 3 mins (this is because I have a WIC-1T FRelay connection and not an HWIC-1T), it finally gives me the error "JTAPI Login Failed. Error while determining CCM Version-1"... tried reloading the image (same version) didn't work (maybe another version would work... instead of the 70.6 that i have).... what I did next was to configure the CUE module from the CLI with the cucm info... (jtapi user, ctip ports... etc..)... before you do this make sure you have the right license file for cue loaded, you could verify this using "show software licenses", if you have the cme license file loaded something like "cue-vm-license_12mbx_cme_7.0.6.pkg", then you better load the ccm license file "cue-vm-license_12mbx_ccm_7.0.6.pkg" by issuing the command "sofware install clean url ftp://x.x.x.x/cue-vm-license_12mbx_ccm_7.0.6.pkg" where (x.x.x.x is the ftp server, install the ftp software like ftpd or filezilla on your pc and give the ip of your pc in the x.x.x.x above).... will ask you to reload and will take like 7mins... you could then re-issue the "show software licenses" command to view the license... should be ccm now... .. you could then try the normal recommended config to have it integrated... Alex has posted a good post on it... or a post from "voiceonbits" at the links below;

http://alexsj9.blogspot.com/2010/03/integrating-cue-and-cucm.htmlhttp://voiceonbits.com/2010/07/17/cisco-unity-express-setup-cue-cucm/

and if all hell brakes loose with the " JTAPI Login Failed. Error while determining CCM Version-1" error then try reloading the image and still nothing then....

CUE CLI Config for CUE-CUCM Integration

ccn subsystem jtapi
ctiport 4441 4442 (note: these are already created in cucm)
ccm-manager address x.x.x.x
ccm-manager username jtapi_cue password cisco
end subsystem

reloaded the cue module and then configured the 

ccn trigger jtapi phonenumber 4440 (this is the cti route point & voice mail pilot configured in cucm)
application "voicemail"
enabled
end trigger

add the below for fallback to srst

cue subsystem sip
gateway address x.x.x.x (cue router address)
mwi sip unsolicited
exit

ccn trigger sip phonenumber 4440
application "voicemail"
enabled
end trigger

with the cti ports and all that and reloaded the module again and still did not see it register in cucm, had a reload of the router.. check to confirm if your cti route point and cti ports have registered in cucm... also issue the "show ccn status ccm-manager" command in cue cli to check the status of the integration, should show up as registered.....

Test by calling the voicemail from the phone and leaving a voicemail from another phone and test the mwi on and off... all works great although the cue module will still show up as uninitialized....

CUCM |IP Phone| Call forwarding to PSTN |mobile number| showing "High Traffic Try Again Later"!

Under CUCM H.323 Gateway Configuration: 

Call Routing Information - Outbound Calls 
Calling Party Selection* Originator

phones don't register to the subscriber by auto register but register to the pub... cucm pub and sub don't sync

if you tried the utils dbreplication repair all or the utils dbreplication reset all and are still stuck with a different replication count in the RTMT Database Summary then go ahead with the 
util dbreplication forcedatasyncsub cucm-sub (where cucm-sub is the host name of my subscriber)... 

I recently faced an issue where my phones were not auto-registering to the subscriber, gave me the Rejected or Error DBConfig (tried restarting the services.... unchecked checked and uncheked the auto registration disabled for the sub server, deleted the phones and tried to have them reregister... played with the order of the pub and sub in the cucm group.... nothing worked) then finally I checked the db replication state by cli and rtmt (note: both can show you different states sometimes... they both should be consistent).... finally ran the above command that I got of voiceie.com link below.... worked a wonder

http://www.voiceie.com/cgi-bin/ultimatebb.cgi?ubb=get_topic;f=8;t=002500

AAR over RSVP CAC

AAR over RSVP CAC

2001 (HQ Ph1) dials 4001 (Branch 2 Ph 1) or vice versa

1. Enable AAR globally from the Service Parameters
2. Configure 1 AAR Group called AAR_General with prefix 9
3. Make sure EPNM is configured for both phones
a. HQ_PH: +14082022XXX
b. SC_PH: +85224044XXX

Now since the AAR Group Prefix is 9, this will be prepended to the EPNM when the call is re-routed due to network
congestion, so next step create a Route Pattern to dial this new modified EPNM.

Route Pattern on CUCM

9\+.85224044XXX (this will be dialed by HQ PH)

Choose RL with Standard Local Route Group and modify the called number as "NANP:PreDot" since HQ Phones can
dial Intl numbers without any leading digits (011), add this route pattern to a partition PT_HQ_AAR and this partition
should be added to a CSS CSS_AAR_HQ and this CCS should be added to each HQPH under the AAR CSS along with
the AAR Group (don’t forget to add the AAR group under the line as well)

9\+.14082022XXX (this will be dialed by SC PH)

1. Choose the standard local route group (or otherwise just the gateway) and add a route list with the NANP
Predot and prepend 900 as SC is h323 and the dial-peer in the SC Gateway would be 900T
2. Add this route pattern to a partition PT_AAR_SC and to a CSS CSS_AAR_SC and assign this CSS to the AAR
CSS For SC Phones along with the AAR Group (under the line as well).
3. Go to the location settings and from HQ to SC (this relationship) set the RSVP to mandatory and similarly
from SC to HQ as well...
4. Configure one MTP hardware resource on the HQ and SC Router and register it to CUCM and assign them
respectively to the MRGL on the phones....

Configuring CME Auto Attendant/BACD Aamer Ainapore

before you start entering the below commands, enable the debugs... 

debug voip application script 
debug ephone moh 

You can try to stop and restart the script by doing 
- show call application session //to get the session id 
- call application session stop id xx //to stop the script and restart it by issuing 
- call application voice load queue //assuming queue is the name of the bacd.tcl service. 
- call application voice load aa //assuming queue is the name of the bacd-aa.tcl service 

application 
service aa flash:app-b-acd-aa-2.1.2.2.tcl 
paramspace english index 1 
param number-of-hunt-grps 2 
param menu-timeout 5 
param dial-by-extension-option 5 
param handoff-string aa 
paramspace english language en 
param max-time-vm-retry 2 
param aa-pilot 4042833 
param max-extension-length 4 
paramspace english location flash: 
param second-greeting-time 60 
param welcome-prompt _bacd_welcome.au //caution: many make a mistake and type en_bacd_welcome.au instead of _bacd_welcome.au .... make sure not to type the en at the begining... i made that mistake too initially...// 
param call-retry-timer 15 
param voice-mail 3001 
param max-time-call-retry 600 
paramspace english prefix en 
param service-name queue 

service queue flash:app-b-acd-2.1.2.2.tcl 
param queue-len 10 
param aa-hunt3 1111 
param aa-hunt4 2222 
param queue-manager-debugs 1 
param number-of-hunt-grps 2 

tftp-server flash:music-on-hold.au 

dial-peer voice 13 voip 
service aa 
destination-pattern 4042833 
session target ipv4:142.102.66.254 
incoming called-number 4042833 
dtmf-relay h245-alphanumeric 
codec g711ulaw 
no vad 

telephony-service 
authentication credential admin cisco 
max-ephones 4 
max-dn 10 
ip source-address 142.102.66.254 port 2000 
system message HK SITE (R3) CME 
url services http://142.102.66.253/voiceview/common/login.do 
url authentication http://142.102.66.254/CCMCIP/authenticate.asp 
load 7970 SCCP70.8-3-3S 
time-zone 37 
date-format dd-mm-yy 
voicemail 4220 
max-conferences 8 gain -6 
moh music-on-hold.au 
web admin system name admin password cisco 
web admin customer name aamer password cisco 
dn-webedit 
time-webedit 
transfer-system full-consult 
create cnf-files version-stamp Jan 01 2002 00:00:00 

ephone-hunt 1 longest-idle 
pilot 1111 
list 3001, 3002, 3004 
timeout 18, 18, 18 
statistics collect 

ephone-hunt 2 longest-idle 
pilot 2222 
list 3003 
timeout 18 
statistics collect

SIP Phone Registration Config... CME

voice service voip 
allow-connections h323 to h323 
allow-connections h323 to sip 
allow-connections sip to h323 
allow-connections sip to sip 

sip 
bind control source-interface Loopback0 (Optional, but it is better to bind the traffic) 
bind media source-interface Loopback0 (Optional, but it is better to bind the traffic) 
registrar server expires max 300 min 60 



voice register global 
mode cme 
source-address 177.3.254.1 port 5060 
max-dn 25 
max-pool 10 
load 7970 SIP70.8-3-1S 
authenticate register 
tftp-path flash: 
create profile sync 001513024159287A 



voice register dn 1 
number 3001 

voice register dn 2 
number 3002 

voice register dn 3 
number 3003 

voice register dn 4 
number 3004 


voice register pool 1 
id mac 0024.C442.D5E7 
type 7970 
number 1 dn 1 
username cisco password cisco 

voice register pool 2 
id mac 0024.C442.D5E3 
type 7970 
number 1 dn 2 
username cisco password cisco 



voice register pool 3 
id mac 0024.C442.D620 
type 7970 
number 1 dn 3 
username cisco password cisco 


voice register pool 4 
id mac 0024.C442.D5ED 
type 7970 
number 1 dn 4 
username cisco password cisco 


tftp-server flash:cnu70.8-3-0-50.sbn 
tftp-server flash:apps70.8-3-0-50.sbn 
tftp-server flash:cvm70sip.8-3-0-50.sbn 
tftp-server flash:dsp70.8-3-0-50.sbn 
tftp-server flash:jar70sip.8-3-0-50.sbn 
tftp-server flash:SIP70.8-3-1S.loads 
tftp-server flash:term70.default.loads

How to test your FXO ports to see if they have an voltage coming in from the PSTN

test voice port 0/0/0 si-reg-read 29 1 

// in the above command we are testing FXO voice port 0/0/0 

if the output of the above command (make sure you enable "terminal monitor" if you are accessing the router via telenet and not console) is anything other than "0x00" you do have voltage but if you receive a "0x00" then that line is dead .... either make sure the line is properly terminated to the FXO port or call up the PSTN to check it out..

Important/Usefull CME Commands for Auto Attendant/BACD...

CME Auto Attendant (Usefull commands) 

show call application sessions //show all active sessions and their resepctive id's 
call application session stop id 17 //stop a current session by specifying the id retrieved in the previous command above 
call application voice load aa //load the aa service which will load the bacd-aa.tcl 
call application voice load queue //load the queue service which will load the bacd.tcl 
audio-prompt load flash:en_bacd_welcome.au // use this command to verify if the audio file you uploaded is valid on cme 

debug voice application script 

To record greeting use windows sound recorder for xp and 
save the file in the format CCITT u-law format with 8khz 8-bit mono attributes

Unable to register the gateway in CUCM as MGCP... just does'nt register...

Check the following: 

1) Check if a domain name is configured, if it is then either remove it or suffix it after the router hostname (example; instead of mentioning only the hostname as "R1" while registering the gateway in CUCM, register it as "R1.xyz.com" or whatever your domain name is in the router" 

2) shut and bring up mgcp process by issue a "no mgcp" "mgcp" 

3) issue the show ccc-manager command to see the status of the gateway in the router... if it's registered and active you might want to restart the "RIS Data Collector" service in cucm, it does sometimes happen that the cucm does not reflect the values in it's informix database... 

4) make sure the following commands are present in the router for mgcp.... 

ccm-manager redundant-host 177.1.10.11 
ccm-manager mgcp 
! 
mgcp 
mgcp call-agent 177.1.10.1 service-type mgcp version 0.1 
mgcp dtmf-relay voip codec all mode out-of-band 
! 
mgcp profile default 

5) hang the router from the wall and punch it !@!#!#@#@


Config Snippet:

mgcp 
mgcp call-agent 177.1.10.20 
mgcp bind control source lo 0 
mgcp bind media source lo 0 
mgcp dtmf-relay voice codec all mode out-of-band 
ccm-manager redundant host 177.1.10.10 
ccm-manager mgcp 
ccm-manager switchback immediate

Configuring Extension Mobility on CUCM... Part 2

Create Device Profile Default for Each Phone Model that shall Support Cisco Extension Mobility 

Step 1 Select Device>Device Settings>Default Device Profile 
Step 2 From the drop down list, select the phone model to be configured, for example, Cisco 7960. 
Step 3 Under Description, enter a description of this profile. 
Step 4 Under Phone Button Template, select Standard 7960 SCCP. 
Step 5 Click Save 
Step 6 Repeat for each model phone to be configured 

Create Device User Profile for a User 

Step 1 Choose Device>Device Settings>Device Profile and click Add New. 
Step 2 From the drop down list, select the phone model to be configured, for example, Cisco 7960 
Step 3 Click Next 
Step 4 Enter a Device Profile Name, for example “Your Name” . 
Step 5 From the Phone Button Template field, select Standard 7960 SCCP. 
Step 6 Click Save. 
Step 7 On the left hand side of the screen, click the link Line [1] – Add a new DN. 
Step 8 Choose a valid DN from your NIP, enter that DN in the Directory Number field. 
Step 9 Under Route Partition, select your city’s Headquarters Partition. 
Step 10 Under Directory Number Settings choose a CSS of appropriate access. 
Step 11 Enter any Call Forward and Call Pickup Settings as necessary. 
Step 12 In the Display (Internal Caller ID), enter the User’s name that was created in a previous Lab. 
Step 13 Click Save. 
Step 14 From the Related Links: menu, select Subscribe/Unsubscribe Services. 
Step 15 In the Select a Service, select Extension Mobility, then click Next. 
Step 16 Click Subscribe. 
Step 17 Click Save. 
Step 18 Repeat steps 7-13 for any additional lines. 

Associate User Device Profile to a User 

Step 1 From the menu, select User Management>End User. 
Step 2 Click Find 
Step 3 Select the user from the list that matches the profile that was created. 
Step 4 Under Extension Mobility>Available Profiles, select the profile that was created in the previous exercise and move it to the Controlled Profiles selection. 
Step 5 Under Default Profile, select the profile. 
Step 6 Click Save. 

Configure and Subscribe Cisco Unified Ip Phones to Service and Enable it. 

Step 1 Select Device>Phone from the menu. 
Step 2 Select the phone from the list of devices. 
Step 3 In the Related Links: field, select Subscribe/Unsubscribe Services and click Go 
Step 4 In the pop-up window, under Service Information, in the Select a Service pull down menu, select Extension Mobility. 
Step 5 Click Next 
Step 6 Click Subscribe 
Step 7 Click Save 
Step 8 Close the pop-up window. 
Step 9 Under Extension Information , check the Enable Extension Mobility box. 
Step 10 Under the Logout Profile field, select – Use Current Device Settings – 
Step 11 Click Save. 
Step 12 Click Ok from the pop-up warning. 
Step 13 Click Reset 
Step 14 In the pop-up window select Reset. 
Step 15 Click Close.