Class Snap
Saturday, April 12, 2014
Thursday, April 3, 2014
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
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....
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
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
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....
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....
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