CME-BACD

I have a problem . this is the first time to install CME-BACD . when I finshed the instllation there are an error apper :
Oct 24 06:55:46.027: //3//TCL :/tcl_PutsObjCmd: TCL AA: +++ B-ACD-SERVICE not registered, Starting B-ACD-SERVICE +++
Oct 24 06:55:46.027: //3//AFW_:/AFW_FSM_Drive: Tcl_Eval to drive FSM inside Tcl modulespace. code=1 code=ERROR
Oct 24 06:55:46.027: TCL script failure
        Result:
                         Handoff Failed
Oct 24 06:55:46.027:    TCL script failure errorInfo:
                        Handoff Failed
    while executing
"handoff appl leg_incoming $serviceName -s $hString"
    (procedure "act_Setup" line 37)
    invoked from within
"act_Setup"
My configuration in attached... Please help me

Hi Alhassan
please find after update configuration:-
f
dial-peer voice 40 pots
service app-b-acd-aa
incoming called-number 29001
direct-inward-dial
dial-peer voice 2 voip
description Dialpeer to Branches
service app-b-acd-aa
destination-pattern 2....
session target ipv4:10.x.x.x
incoming called-number 29001
codec g711ulaw
application
service app-b-acd-aa
param max-time-call-retry 600
param voice-mail 29003
paramspace english index 1
param service-name app-b-acd
param queue-exit-extension1 29001
param number-of-hunt-grps 2
param menu-timeout 6
param handoff-string app-b-acd-aa
param dial-by-extension-option 3
paramspace english language en
param max-time-vm-retry 2
param max-extension-length 3
param aa-pilot 29001
paramspace english location flash:
param second-greeting-time 30
param welcome-prompt en_bacd_welcome.au
param queue-manager-debugs 1
param call-retry-timer 10
service app-b-acd
param queue-len 10
param aa-hunt1 29006
param queue-manager-debugs 1
param aa-hunt2 29007
param number-of-hunt-grps 2
Thank you
Please rate all useful information

Similar Messages

CME B-ACD on Cisco 2911 with IOS 15.2(4)M5 not working

Hi Folks,
I am currently setting up CME version 9.1 with B-ACD (app-b-acd-aa-3.0.0.2.tcl & app-b-acd-3.0.0.2.tcl), running on
Cisco 2911 with IOS ver 15.2(4)M5, this is for lab purposes.
Below is my CME & B-ACD configuration :
voice service voip
ip address trusted list
ipv4 0.0.0.0 0.0.0.0
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
h323
h225 listen-port 1820
no call service stop
sip
bind control source-interface Vlan400
bind media source-interface Vlan400
registrar server expires max 600 min 60
voice register global
mode cme
source-address 172.25.202.1 port 5060
max-dn 2
max-pool 2
load 9971 sip9971.9-2-2SR1-9
authenticate register
timezone 28
time-format 24
date-format D/M/Y
tftp-path flash:
create profile sync 0004714411607756
voice register dn 1
number 3005
name br2phn2
voice register dn 2
number 3006
name br2phn4
voice register template 1
dialplan 1
voice register dialplan 1
type 7940-7960-others
pattern 1 3...
pattern 2 999
voice register pool 1
id mac 1C1D.86C4.0D6D
type 9971
number 1 dn 1
template 1
dtmf-relay rtp-nte
username 3005 password cisco
description 3214-3005
codec g711ulaw
voice register pool 2
id mac 1C1D.86C4.A574
type 9971
number 1 dn 2
template 1
dtmf-relay rtp-nte
username 3006 password cisco
description 3214-3006
codec g711ulaw
voice hunt-group 1 parallel
list 3002,3006
pilot 3210
application
service aa flash:/app-b-acd-aa-3.0.0.2.tcl
paramspace english index 1
param number-of-hunt-grps 2
param handoff-string aa
paramspace english language en
param max-time-vm-retry 2
param aa-pilot 3500
paramspace english location flash://
param second-greeting-time 60
param welcome-prompt _bacd_welcome.au
param call-retry-timer 15
param voice-mail 3001
param max-time-call-retry 90
param service-name queue
service aa-drop flash:/app-b-acd-aa-3.0.0.2.tcl
paramspace english index 1
param service-name queue
param drop-through-option 2
param second-greeting-time 60
paramspace english language en
param max-time-vm-retry 2
param max-time-call-retry 90
param voice-mail 3001
paramspace english location flash://
param aa-pilot 3501
param number-of-hunt-grps 1
param handoff-string aa-drop
param call-retry-timer 15
service queue flash:/app-b-acd-3.0.0.2.tcl
param queue-len 15
param aa-hunt10 3006
param queue-manager-debugs 1
param number-of-hunt-grps 2
param aa-hunt2 3210
interface Loopback0
ip address 172.25.110.3 255.255.255.255
ip ospf network point-to-point
h323-gateway voip interface
h323-gateway voip id Spain ipaddr 172.25.110.1 1719
h323-gateway voip h323-id BR2-RTR
h323-gateway voip tech-prefix 1#
h323-gateway voip bind srcaddr 172.25.110.3
interface Vlan400
ip address 172.25.202.1 255.255.255.0
ip pim dense-mode
dial-peer voice 3500 voip
service aa
destination-pattern 3500
session target ipv4:172.25.110.3
incoming called-number 3500
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
dial-peer voice 3501 voip
service aa-drop
destination-pattern 3501
session target ipv4:172.25.110.3
incoming called-number 3501
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
telephony-service
no auto-reg-ephone
max-ephones 2
max-dn 2 no-reg both
ip source-address 172.25.110.3 port 2000
cnf-file location flash:
load 7965 term65.default.loads
time-zone 28
time-format 24
date-format dd-mm-yy
max-conferences 8 gain -6
moh "music-on-hold.au"
web admin system name admin password cisco
dn-webedit
transfer-system full-consult
create cnf-files version-stamp 7960 Feb 14 2014 05:54:44
ephone-template 1
softkeys connected Endcall Hold Park Trnsfer Acct Flash
ephone-dn 1 octo-line
number 3001 no-reg both
description 3214-3001
name br2phn1
ephone-dn 2 octo-line
number 3002 no-reg both
description 3214-3002
name br2phn3
ephone 1
device-security-mode none
mac-address 189C.5DB6.D303
ephone-template 1
max-calls-per-button 5
busy-trigger-per-button 3
type 7965
button 1:1
ephone 2
device-security-mode none
description 3214-3002
mac-address 984B.E194.FDDD
ephone-template 1
max-calls-per-button 5
busy-trigger-per-button 3
type 7960
button 1:2
Problem :
1. When I test call from CME Phone both SIP and SCCP Phone by dial 3500 or 3501, I get the busy tone.
2. Debug voip dial-peer, match with dial-peer voice 3500 for (aa service) & 3501 for (aa-drop service).
3. Debug voice application script, show nothing.
Is there something wrong with my configuration ?
Rgds
Novri

Hi Novriadi,
In your configuration
service aa flash:/app-b-acd-aa-3.0.0.2.tcl
service queue flash:/app-b-acd-3.0.0.2.tcl
paramspace english location flash://
Remove "/" and "//" from the configuration
Then use the call application voice load command in privileged EXEC mode to reload the scripts.
Router# call application voice load aa
Router# call application voice load queue
Router# call application voice load aa-drop
You can refer to following document as well for more info
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucme/bacd/configuration/guide/cme40tcl/40bacd.html#wp1018270
Please find the sample configuration that is required to configure b-acd in CME for reference.
telephony-service
moh music-on-hold.au
multicast moh 239.1.1.1 port 2000
application
service queue flash:app-b-acd-2.1.0.0.tcl
param number-of-hunt-grps 2
param aa-hunt2 1111
param aa-hunt3 1222
param queue-len 15
param queue-manager-debugs 1
service aa flash:app-b-acd-aa-2.1.0.0.tcl
paramspace english index 1
paramspace english language en
paramspace english location flash:
param service-name queue
param handoff-string aa
param aa-pilot 8005550123
param welcome-prompt _bacd_welcome.au
param number-of-hunt-grps 2
param dial-by-extension-option 1
param second-greeting-time 60
param call-retry-timer 15
param max-time-call-retry 700
param max-time-vm-retry 2
param voice-mail 5003
dial-peer voice 222 voip
service aa
destination-pattern 8005550123
session target ipv4:192.168.1.1
incoming called-number 8005550123
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
Thanks & Regards,
Mudit Mathur

CME Basic Question

I have call manager express 7.1 installed. Can I configure greeting message for one extension ? does it requires unity express as well ?

Hello Syed,
if you can see the both scenarios there are much differences in both design. CME with BACD (AA) has limited features when comparing CME with CUE AA in terms of features. below is the synopsis just go through once you will know differences and their use.
Scenario 1 is most applicable when your Cisco CME AA is limited to providing a menu to direct calls into ACD groups. In this scenario, Cisco CME is used as the AA destination for incoming calls, and Cisco Unity Express is used exclusively as a voice mail server when the ACD agents are not available or do not answer the calls in a certain amount of time. The Cisco Unity Express AA is not used in this scenario.
Scenario 2 is the most flexible overall configuration for integrating Cisco CME basic ACD and Cisco Unity Express AA. The design used in Scenario 2 features Cisco Unity Express AA for general call-handling automation and for handing off calls to ACD agents. In this scenario, Cisco Unity Express is used as the AA destination for incoming calls; Cisco Unity Express also provides a variety of menu choices to the caller (such as dial-by-name, dial-by-extension, and recorded information segments). One of these choices directs some calls to the Cisco CME ACD groups. Cisco Unity Express voice mail is used in a similar manner as Scenario 1.
Br,
Nadeem
Please rate all useful post.

Define a default destination on CME B-ACD AA

Hi,
we have a CME B-ACD configured with an Auto Attendant prompt menu and then two hunt-group options.
The problem we are having is that when calling from certains old analog phones the system doesn't recognize the inputs ("1" or "2"), so the call is not forwarded to any of the hunt-groups and the Auto Attendant prompt menu keeps playing on loop.
Is there any way to configure a default destination when no option is pressed after a defined timeout?
I attach the config.
Thanks,
Isaac.

Hallo Oscar,
when u need calls to be directed to the huntgroup without waiting for the callers input, u need to configure param-drop-through-option, under the the call queing section.
But u will need to change the voice prompt that tells the callers to press a number to reach a call agent with another that has some kind of advertisement.
The voice prompt is to be given a file name "en_bacd_options_menu.au"
Read the "Setting Up Call-Queue and AA Services" section in "Cisco Unified CME B-ACD and Tcl Call- handling applications" .
get the book from this link.
http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/bacd/configuration/guide/40tcl.pdf
check out this configs
application
service queue flash:app-b-acd-2.1.2.2.tcl
param queue-len 10
param aa-hunt1 7597
param aa-hunt2 7598
param queue-manager-debugs 1
param number-of-hunt-grps 2
service aa flash:app-b-acd-aa-2.1.2.2.tcl
param number-of-hunt-grps 1
paramspace english index 1
param drop-through-option 1
param handoff-string aa
paramspace english language en
param max-time-vm-retry 2
param aa-pilot 7599
param max-extension-length 3
paramspace english location flash:
param drop-through-prompt _bacd_options_menu.au
param second-greeting-time 1
param welcome-prompt _bacd_welcome.au
param call-retry-timer 15
param max-time-call-retry 60
param voice-mail 598
param service-name queue
service aa1 flash:app-b-acd-aa-2.1.2.2.tcl
param number-of-hunt-grps 1
paramspace english index 1
param drop-through-option 2
param handoff-string aa1
paramspace english language en
param max-time-vm-retry 2
param aa-pilot 7598
param max-extension-length 3
paramspace english location flash:
param drop-through-prompt _bacd_options_menu.au
param second-greeting-time 1
param welcome-prompt _bacd_welcome.au
param call-retry-timer 15
param max-time-call-retry 120
param voice-mail 598
param service-name queue
since there is no option of choosing huntgroups, u will need to create a second incoming dial-peer for the second huntgroup. in that scenario u will be having 2 pilot numbers each directed to a particular aa service, i.e aa and aa1
Good luck.
Symon

How to restart CME Auto Attendant ?

What command can reload AA but do not reload CME?
When change a CME on AA configure or change the AA en_welcome.au file.
application
service aa flash:its-CISCO.2.0.2.0.tcl
param operator 8000
paramspace english language en
paramspace english index 0
paramspace english location flash:
paramspace english prefix en
param aa-pilot 8000
param max-extension-length 4
param welcome-prompt en_welcome.au

Here you are:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/bacd/configuration/guide/40bacd.html#wp1055309
Please remember to rate helpful responses and identify helpful or correct answers.

Cisco Unified CME B-ACD and Tcl

Good afternoon,
I have router Cisco IOS Software, C2900 Software (C2900-UNIVERSALK9-M), Version 15.2(4)M6a, with app-b-acd-2.1.2.3.tcl (this i upload to router because i dont find the scripts or TAR archive in the system)
I just need B-ACD to run the welcome greeting and then ask the user to dial option 1 to the extension or 0 for recepcionist.
I don't want the B-ACD to pass the user to different queues.
the audio files are in spanish.
My application configuration is:
application
service aa flash0:/cme-b-acd/app-b-acd-aa-2.1.2.3.tcl
paramspace english index 1
param number-of-hunt-grps 1
param menu-timeout 6
param handoff-string aa
param dial-by-extension-option 0
paramspace english language en
param aa-pilot XXXXXXX (public number)
param max-extension-length 5
paramspace english location flash0:/cme-b-acd/
param second-greeting-time 120
param welcome-prompt _bacd_welcome.au
param call-retry-timer 30
param voice-mail XXXXX (is this necesary?????)
param max-time-call-retry 600
param service-name queue
service queue flash0:/cme-b-acd/app-b-acd-2.1.2.3.tcl
param aa-hunt1 XXXXX (extension of receptionist)
param number-of-hunt-grps 1
Please can you help me and say if this correct ???????
Other questions: i upload the audio files in the same directory that de aplications programs in my case:
Directory of flash0:/cme-b-acd/
431 -rw-       26087 Feb 11 2015 15:30:42 -03:00 app-b-acd-2.1.2.3.tcl
432 -rw-       19191 Feb 11 2015 15:32:06 -03:00 app-b-acd-2.1.2.3-ReadMe.txt
433 -rw-       37673 Feb 11 2015 15:32:30 -03:00 app-b-acd-aa-2.1.2.3.tcl
Here?????
Thanks and best regards.,
Camilo
mmxv

Hello,
Change this (dial-by-extension-option 1) and (param aa-hunt1) is an ephone hunt group that has operator's extension.
Several complete examples with explanation can be found at the link below:
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucme/bacd/configuration/guide/cme40tcl/40bacd.html#wp1003141
Thank you,
Shadi

Cisco SIP Phone 9971 won't register on CME 8.6 or 8.5 Please HELP

Please help me , I have problem with registering Cisco SIP phone 9971 with CME 8.6 on ISR 2901.
I configured CME for SIP clients, then I add configuration for 9971 phone and create profiles. Phone downloaded SEP...xml file from CME,after that phone look for g4-tones.xml and gd-sip.jar files, I added them to CME after that phone downloaded them and reboot. Now phone is stuck in some kind of loop and does not register on CME.
On phone log I can see repeting next few messeges.
12:01:58a No DNS Server IP
12:01:59a Updating Trust list
12:01:59a No Trust List instaled
12:01:59a SEP04C5AB03B0D.cnf.xml (TFTP) // at this time phone download SEP...xml file from CME
12:02:00a VPN Error: VPN is not Configured
on CME if issue DEBUG TFTP EVENTS i receive next few lines
*Aug 18 18:20:19.891: TFTP: Looking for CTLSEP04C5A4B03B0D.tlv
*Aug 18 18:20:19.987: TFTP: Looking for ITLSEP04C5A4B03B0D.tlv
*Aug 18 18:20:20.083: TFTP: Looking for ITLFile.tlv
*Aug 18 18:20:20.347: TFTP: Looking for SEP04C5A4B03B0D.cnf.xml
*Aug 18 18:20:20.351: TFTP: Opened flash:/SEP04C5A4B03B0D.cnf.xml, fd 14, size 4585 for process 141
*Aug 18 18:20:20.363: TFTP: Finished flash:/SEP04C5A4B03B0D.cnf.xml, time 00:00:00 for process 141
here you can see verison info of CME
Cisco IOS Software, C2900 Software (C2900-UNIVERSALK9-M), Version 15.1(4)M, RELEASE SOFTWARE (fc1)
Technical Support: http://www.cisco.com/techsupport
Copyright (c) 1986-2011 by Cisco Systems, Inc.
Compiled Thu 24-Mar-11 15:31 by prod_rel_team
ROM: System Bootstrap, Version 15.0(1r)M9, RELEASE SOFTWARE (fc1)
ELTOSAN_ROUTER uptime is 1 hour, 50 minutes
System returned to ROM by reload at 16:29:20 UTC Thu Aug 18 2011
System image file is "flash:/c2900-universalk9-mz.SPA.151-4.M.bin"
Last reload type: Normal Reload
Last reload reason: Reload Command
Cisco CISCO2901/K9 (revision 1.0) with 471040K/53248K bytes of memory.
Processor board ID FGL1508252Y
3 Gigabit Ethernet interfaces
2 terminal lines
1 Virtual Private Network (VPN) Module
4 Voice FXO interfaces
4 Voice FXS interfaces
1 Internal Services Module (ISM) with Services Ready Engine (SRE)
   Survivable Remote Site Voicemail (SRSV) on Cisco Unity Express (CUE) 8.5.1 in slot/sub-slot 0/0
DRAM configuration is 64 bits wide with parity enabled.
255K bytes of non-volatile configuration memory.
254464K bytes of ATA System CompactFlash 0 (Read/Write)
License Info:
License UDI:
Device#   PID                   SN
*0        CISCO2901/K9          xxxxxxxxxxxxx
Technology Package License Information for Module:'c2900'
Technology    Technology-package          Technology-package
              Current       Type          Next reboot
ipbase        ipbasek9      Permanent     ipbasek9
security      securityk9    Permanent     securityk9
uc            uck9          Permanent     uck9
data          None          None          None
Configuration register is 0x2102
this is RUNNING CONFIGURATION
! Last configuration change at 16:10:12 UTC Thu Aug 18 2011
version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname ELTOSAN_ROUTER
boot-start-marker
boot system flash:/c2900-universalk9-mz.SPA.151-4.M.bin
boot-end-marker
no aaa new-model
no ipv6 cef
ip source-route
no ip routing
no ip cef
no ip dhcp use vrf connected
ip dhcp excluded-address 192.168.5.1 192.168.5.10
ip dhcp excluded-address 192.168.5.200 192.168.5.255
ip dhcp pool phone
   network 192.168.5.0 255.255.255.0
   default-router 192.168.5.251
   option 150 ip 192.168.5.251
ip dhcp pool data
   relay source 192.168.2.0 255.255.255.0
   relay destination 192.168.2.201
multilink bundle-name authenticated
crypto pki token default removal timeout 0
voice-card 0
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
fax protocol pass-through g711alaw
sip
registrar server expires max 3600 min 120
voice register global
mode cme
source-address 192.168.5.251 port 5060
max-dn 6
max-pool 6
load 9971 sip9971.9-1-1SR1.loads
authenticate register
tftp-path flash:
create profile sync 0005135312289902
voice register dn 1
number 207
allow watch
name GossaVM
label 207
voice register dn 3
number 101
name Dejan
label 101
mwi
voice register pool 1
id mac 000C.29C5.0011
number 1 dn 1
dtmf-relay sip-notify
username testvm password testera
codec g711alaw
voice register pool 3
id mac 04C5.A4B0.3B0D
type 9971
number 3 dn 3
presence call-list
dtmf-relay rtp-nte
username dejan password 1234
codec g711alaw
no vad
license udi pid CISCO2901/K9 sn xxxxxxxxxxxx
hw-module ism 0
hw-module pvdm 0/0
redundancy
interface GigabitEthernet0/0
description INTERFACE INTERNAL
no ip address
no ip route-cache
duplex auto
speed auto
no mop enabled
interface GigabitEthernet0/0.2
description LAN DATA
encapsulation dot1Q 2
ip address 192.168.2.251 255.255.255.0
no ip route-cache
interface GigabitEthernet0/0.5
description LAN VOICE
encapsulation dot1Q 5
ip address 192.168.5.251 255.255.255.0
no ip route-cache
interface ISM0/0
no ip address
no ip route-cache
shutdown
!Application: SRSV-CUE Running on ISM
interface GigabitEthernet0/1
no ip address
no ip route-cache
shutdown
duplex auto
speed auto
interface ISM0/1
description Internal switch interface connected to Internal Service Module
shutdown
interface Vlan1
no ip address
no ip route-cache
shutdown
ip forward-protocol nd
no ip http server
no ip http secure-server
snmp-server community public RO
tftp-server flash:dkern9971.100609R2-9-1-1SR1.sebn alias dkern9971.100609R2-9-1-1SR1.sebn
tftp-server flash:kern9971.9-1-1SR1.sebn alias kern9971.9-1-1SR1.sebn
tftp-server flash:rootfs9971.9-1-1SR1.sebn alias rootfs9971.9-1-1SR1.sebn
tftp-server flash:sboot9971.031610R1-9-1-1SR1.sebn alias sboot9971.031610R1-9-1-1SR1.sebn
tftp-server flash:skern9971.022809R2-9-1-1SR1.sebn alias skern9971.022809R2-9-1-1SR1.sebn
tftp-server flash:sip9971.9-1-1SR1.loads alias sip9971.9-1-1SR1.loads
tftp-server flash:United_States/g4-tones.xml
tftp-server flash:English_United_States/gd-sip.jar
control-plane
voice-port 0/0/0
voice-port 0/0/1
voice-port 0/0/2
voice-port 0/0/3
voice-port 0/1/0
voice-port 0/1/1
voice-port 0/1/2
voice-port 0/1/3
mgcp profile default
gatekeeper
shutdown
line con 0
line aux 0
line 67
no activation-character
no exec
transport preferred none
transport input all
transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
stopbits 1
line vty 0 4
password jebiga
login
transport input all
end
I did not have any kind of problem with X-LITE to register to CME. also try with few SCCP phones 7940 and I did not any kind of problem .
this is content of SEP....xml file for 9971
<device>
<deviceProtocol>SIP</deviceProtocol>
<devicePool>
<dateTimeSetting>
<dateTemplate>M/D/YA</dateTemplate>
<timeZone>Pacific Standard/Daylight Time</timeZone>
<ntps>
<ntp priority="0">
<name>0.0.0.0</name>
<ntpMode>unicast</ntpMode>
</ntp>
</ntps>
</dateTimeSetting>
<callManagerGroup>
<members>
<member priority="0">
<callManager>
<ports>
<sipPort>5060</sipPort>
</ports>
<processNodeName>192.168.5.251</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
</devicePool>
<sipProfile>
<sipProxies>
<registerWithProxy>true</registerWithProxy>
</sipProxies>
<sipCallFeatures>
<cnfJoinEnabled>true</cnfJoinEnabled>
<localCfwdEnable>true</localCfwdEnable>
<callForwardURI>service-uri-cfwdall</callForwardURI>
<callPickupURI>service-uri-pickup</callPickupURI>
<callPickupGroupURI>service-uri-gpickup</callPickupGroupURI>
<callHoldRingback>2</callHoldRingback>
<semiAttendedTransfer>true</semiAttendedTransfer>
<anonymousCallBlock>2</anonymousCallBlock>
<callerIdBlocking>2</callerIdBlocking>
<dndControl>2</dndControl>
<remoteCcEnable>true</remoteCcEnable>
</sipCallFeatures>
<sipStack>
<remotePartyID>true</remotePartyID>
</sipStack>
<sipLines>
<line button="1" lineIndex="1">
<featureID>9</featureID>
<featureLabel></featureLabel>
<proxy>USECALLMANAGER</proxy>
<port>5060</port>
<name></name>
<displayName></displayName>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>1</callWaiting>
<authName>dejan</authName>
<authPassword>1234</authPassword>
<sharedLine>false</sharedLine>
<messagesNumber></messagesNumber>
<ringSettingActive>5</ringSettingActive>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>true</callerNumber>
<redirectedNumber>true</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>
<line button="2" lineIndex="2">
<featureID>9</featureID>
<featureLabel>101</featureLabel>
<proxy>USECALLMANAGER</proxy>
<port>5060</port>
<name>101</name>
<displayName>Dejan Rakic</displayName>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>1</callWaiting>
<authName>dejan</authName>
<authPassword>1234</authPassword>
<sharedLine>false</sharedLine>
<messagesNumber></messagesNumber>
<ringSettingActive>5</ringSettingActive>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>true</callerNumber>
<redirectedNumber>true</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>
</sipLines>
<enableVad>true</enableVad>
<preferredCodec>g711alaw</preferredCodec>
<dialTemplate></dialTemplate>
<kpml>1</kpml>
<phoneLabel></phoneLabel>
<stutterMsgWaiting>2</stutterMsgWaiting>
<disableLocalSpeedDialConfig>true</disableLocalSpeedDialConfig>
<dscpForAudio>184</dscpForAudio>
<dscpVideo>136</dscpVideo>
</sipProfile>
<commonProfile>
<phonePassword>1234</phonePassword>
<callLogBlfEnabled>2</callLogBlfEnabled>
</commonProfile>
<featurePolicyFile>featurePolicyDefault.xml</featurePolicyFile>
<loadInformation>sip9971.9-1-1SR1.loads</loadInformation>
<vendorConfig>
</vendorConfig>
<commonConfig>
<videoCapability>0</videoCapability>
<ciscoCamera>0</ciscoCamera>
</commonConfig>
<sshUserId>dejan</sshUserId>
<sshPassword>1234</sshPassword>
<userId></userId>
<phoneServices>
<provisioning>2</provisioning>
<phoneService type="1" category="0">
<name>Missed Calls</name>
<phoneLabel></phoneLabel>
<url>Application:Cisco/MissedCalls</url>
<vendor></vendor>
<version></version>
</phoneService>
<phoneService type="1" category="0">
<name>Received Calls</name>
<phoneLabel></phoneLabel>
<url>Application:Cisco/ReceivedCalls</url>
<vendor></vendor>
<version></version>
</phoneService>
<phoneService type="1" category="0">
<name>Placed Calls</name>
<phoneLabel></phoneLabel>
<url>Application:Cisco/PlacedCalls</url>
<vendor></vendor>
<version></version>
</phoneService>
<phoneService type="2" category="0">
<name>Voicemail</name>
<phoneLabel></phoneLabel>
<url>Application:Cisco/Voicemail</url>
<vendor></vendor>
<version></version>
</phoneService>
</phoneServices>
<versionStamp>0131511014412102</versionStamp>
<userLocale>
<name>English_United_States</name>
<langCode>en</langCode>
</userLocale>
<networkLocale>United_States</networkLocale>
<networkLocaleInfo>
<name>United_States</name>
</networkLocaleInfo>
<authenticationURL></authenticationURL>
<directoryURL></directoryURL>
<servicesURL>http://192.168.5.251:80/CMEserverForPhone/serviceurl</servicesURL>
<dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices>
<dscpForCm2Dvce>96</dscpForCm2Dvce>
<transportLayerProtocol>2</transportLayerProtocol>
</device>

Hello,
I'm facing exactly the same problem, that is:
a Cisco SIP Phone 9971 won't register on CME 8.6 running on a 2811
I have read all the postings to this Forum, but I have not been able to solve it.
In my case the commands voice register dn and voice register pool are OK.
So frankly, I have no idea what I could be missing.
I'm pasting the Router's config.
I hope somebody is able to point me in the right direction.
Here is the config. Thank you!
C2811#sh run
Building configuration...
version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname C2811
no aaa new-model
dot11 syslog
ip source-route
ip cef
ip dhcp excluded-address 172.25.140.1 172.25.140.10
ip dhcp excluded-address 172.35.140.1 172.35.140.10
ip dhcp pool Data
network 172.25.140.0 255.255.255.0
default-router 172.25.140.1
option 150 ip 172.25.140.1
dns-server 172.25.140.1
ip dhcp pool Voice
network 172.35.140.0 255.255.255.0
default-router 172.35.140.1
option 150 ip 172.35.140.1
dns-server 172.35.140.1
no ip domain lookup
no ipv6 cef
multilink bundle-name authenticated
voice service voip
allow-connections sip to sip
sip
registrar server expires max 3600 min 120
voice register global
mode cme
source-address 172.25.140.1 port 5060
max-dn 40
max-pool 42
load 9971 sip9971.9-4-1-9.loads
authenticate register
authenticate realm cisco
tftp-path flash:
create profile sync 0004820400584603
voice register dn 1
number 1010
allow watch
name Phone10
label Phone10
mwi
voice register pool 1
id mac 189C.5DB6.BD09
type 9971
number 1 dn 1
presence call-list
dtmf-relay rtp-nte
username adm password adm
call-forward b2bua busy 68600
codec g711ulaw
no vad
camera
video
voice-card 0
crypto pki token default removal timeout 0
crypto pki trustpoint TP-self-signed-1879153754
enrollment selfsigned
subject-name cn=IOS-Self-Signed-Certificate-1879153754
revocation-check none
rsakeypair TP-self-signed-1879153754
crypto pki certificate chain TP-self-signed-1879153754
certificate self-signed 01
(details ommited)
license udi pid CISCO2811 sn FTX1146A44H
username admin privilege 15 password 0 admin
redundancy
interface FastEthernet0/0
no ip address
duplex auto
speed auto
interface FastEthernet0/0.25
description Data VLAN
encapsulation dot1Q 25
ip address 172.25.140.1 255.255.255.0
interface FastEthernet0/0.35
description Voice VLAN
encapsulation dot1Q 35
ip address 172.35.140.1 255.255.255.0
interface FastEthernet0/1
no ip address
shutdown
duplex auto
speed auto
ip forward-protocol nd
ip http server
ip http authentication local
ip http secure-server
ip http timeout-policy idle 600 life 86400 requests 10000
tftp-server flash:P00308010200.bin
tftp-server flash:P00308010200.sbn
tftp-server flash:P00308010200.sb2
tftp-server flash:P00308010200.loads
tftp-server flash:SCCP42.9-3-1SR3-1S.loads
tftp-server flash:apps42.9-3-1ES19.sbn
tftp-server flash:cnu42.9-3-1ES19.sbn
tftp-server flash:cvm42sccp.9-3-1ES19.sbn
tftp-server flash:dsp42.9-3-1ES19.sbn
tftp-server flash:jar42sccp.9-3-1ES19.sbn
tftp-server flash:term42.default.loads
tftp-server flash:term62.default.loads
tftp-server flash:SCCP45.9-3-1SR3-1S.loads
tftp-server flash:apps45.9-3-1ES19.sbn
tftp-server flash:cnu45.9-3-1ES19.sbn
tftp-server flash:cvm45sccp.9-3-1ES19.sbn
tftp-server flash:dsp45.9-3-1ES19.sbn
tftp-server flash:jar45sccp.9-3-1ES19.sbn
tftp-server flash:term45.default.loads
tftp-server flash:term65.default.loads
tftp-server flash:/Ringtones/Ringlist.xml alias Ringlist.xml
tftp-server flash:/Ringtones/DistinctiveRingList.xml alias DistinctiveRingList.x
ml
tftp-server flash:sip9971.9-4-1-9.loads
tftp-server flash:kern9971.9-4-1-9.sebn
tftp-server flash:rootfs9971.9-4-1-9.sebn
tftp-server flash:dkern9971.100609R2-9-4-1-9.sebn
tftp-server flash:sboot9971.031610R1-9-4-1-9.sebn
tftp-server flash:skern9971.022809R2-9-4-1-9.sebn
tftp-server flash:/g4-tones.xml alias United_States/g4-tones.xml
tftp-server flash:/gd-sip.jar alias English_United_States/gd-sip.jar
control-plane
mgcp profile default
telephony-service
max-ephones 24
max-dn 48
ip source-address 172.25.140.1 port 2000
cnf-file location flash:
load 7960-7940 P00308010200
load 7942 SCCP42.9-3-1SR3-1S.loads
load 7945 SCCP45.9-3-1SR3-1S.loads
load 7962 SCCP42.9-3-1SR3-1S.loads
load 7965 SCCP45.9-3-1SR3-1S.loads
max-conferences 8 gain -6
dn-webedit
transfer-system full-consult
create cnf-files version-stamp 7960 Feb 11 2014 07:18:32
ephone-dn 1
number 1001
description Phone 1
name Phone 1
hold-alert 30 originator
ephone-dn 2
number 1002
description Phone 2
name Phone 2
hold-alert 30 originator
ephone-dn 3
number 1003
description Phone 3
name Phone 3
hold-alert 30 originator
ephone 1
device-security-mode none
mac-address 001C.58FB.6E0F
button 1:1
ephone 2
device-security-mode none
mac-address 0014.A981.7F8A
button 1:2
ephone 3
device-security-mode none
mac-address 0006.5356.A4B8
button 1:3
alias exec con conf t
alias exec sib show ip int brief
alias exec srb show run | b
alias exec sri show run int
line con 0
exec-timeout 0 0
logging synchronous
line aux 0
line vty 0 4
privilege level 15
login local
transport input telnet ssh
transport output telnet ssh
line vty 5 15
privilege level 15
login local
transport input telnet ssh
transport output telnet ssh
scheduler allocate 20000 1000
ntp master 1
end
C2811#

Issue with LPCOR on CME 10.5

Dear All,
I am facing issues with LPCOR configuration on CME 10.5. For International calls the Authentication Prompts triggers some times and some times doen not.
Also when a local call is dialed the Authentication Prompt is triggered some times.Below is the config and debug logs. Need your help to resolve this.
voice lpcor enable
voice lpcor custom
group 10 endusers
group 11 pstn
voice lpcor policy endusers
service fac
accept endusers fac
accept pstn fac
voice lpcor policy pstn
service fac
accept endusers fac
accept pstn fac
application
package auth
param passwd-prompt flash:enter_pin.au
param max-retries 0
param abort-digit *
param term-digit #
param user-prompt flash:enter_account.au
param passwd 12345
param max-digits 32
interface GigabitEthernet0/1.1
encapsulation dot1Q 1 native
ip address 10.25.76.1 255.255.255.0
interface GigabitEthernet0/1.201
encapsulation dot1Q 201
ip address 10.25.77.1 255.255.255.0
voice-port 0/0/0
lpcor outgoing pstn
trunk-group ALL_FXO 1
supervisory disconnect dualtone mid-call
supervisory custom-cptone 2n-gsm
no battery-reversal
input gain -6
output attenuation -3
cptone SA
timeouts call-disconnect 1
timeouts wait-release 1
timing sup-disconnect 50
connection plar 5040
caller-id enable
cable-detect
dial-peer cor custom
name local
name longdistance
name 911
name Internal
name fac-int
name user-fac
dial-peer cor list local
member local
dial-peer cor list call-local
member local
dial-peer cor list call-longdistance
member longdistance
dial-peer cor list user1
member local
member 911
dial-peer cor list user2
member local
member longdistance
member 911
member user-fac
dial-peer cor list user3
member 911
dial-peer cor list call-911
member 911
dial-peer cor list call-internal
member Internal
dial-peer cor list fac-int
member local
member 911
member fac-int
dial-peer cor list user-fac
member user-fac
dial-peer voice 96 pots
trunkgroup ALL_FXO
corlist outgoing call-911
destination-pattern 9[2-6]......
forward-digits 7
dial-peer voice 901 pots
trunkgroup ALL_FXO
corlist outgoing call-911
destination-pattern 901[2-4,6-8].......
forward-digits 10
dial-peer voice 800 pots
trunkgroup ALL_FXO
destination-pattern 9800T
prefix 800
dial-peer voice 900 pots
destination-pattern 9T
port 0/0/3
prefix 9
dial-peer voice 11 pots
destination-pattern 901........
port 0/0/3
forward-digits 10
dial-peer voice 9051 pots
trunkgroup ALL_FXO
corlist outgoing call-local
destination-pattern 905........
forward-digits 10
dial-peer voice 19 pots
trunkgroup ALL_FXO
corlist outgoing fac-int
destination-pattern 900T
translate-outgoing called 1
forward-digits all
dial-peer voice 20 voip
description International calling
service clid_authen_collect
destination-pattern 900T
lpcor outgoing pstn
session target ipv4:10.25.76.1
incoming called-number 9T
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
ephone-dn 1
number 4121
name John
corlist incoming fac-int
ephone 1
lpcor type local
lpcor incoming endusers
mac-address E0D1.730A.21DE
ephone-template 2
type 7942
button 1:1
voice register dn 33
number 4163
call-forward b2bua busy 5000
call-forward b2bua noan 5000 timeout 20
call-forward b2bua unregistered 5000
allow watch
name Joseph
mwi
voice register pool 33
busy-trigger-per-button 4
id mac BC67.1C31.C8AA
type 7821
number 1 dn 33
cor incoming fac-int 1 4163
dtmf-relay rtp-nte
codec g711ulaw
transfer max-length 4
Debug Logs
DAMAC-CME-ANOUD#DEBUg VOIce lpcor all
voip lpcor all debugging is on
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#term
DAMAC-CME-ANOUD#terminal i
DAMAC-CME-ANOUD#terminal i
Apr 12 16:22:39.825: %VOIPAAA-5-VOIP_CALL_HISTORY: CallLegType 1, ConnectionId F692C420E06611E4BB0CE7FDC5486EA5, SetupTime 16:22:35.615 UTC Sun Apr 12 2015, PeerAddress 4130, PeerSubAddress , DisconnectCause 10 , DisconnectText normal call clearing (16), ConnectTime 16:22:39.825 UTC Sun Apr 12 2015, DisconnectTime 16:22:39.825 UTC Sun Apr 12 2015, CallOrigin 2, ChargedUnits 0, InfoType 2, TransmitPackets 0, TransmitBytes 0, ReceivePackets 0, ReceiveBytes 0
Apr 12 16:22:39.825: %VOIPAAA-5-VOIP_FEAT_HISTORY: FEAT_VSA=fn:TWC,ft:04/12/2015 16:22:35.609,cgn:4130,cdn:,frs:0,fid:2599,fcid:F692C420E06611E4BB0CE7FDC5486EA5,legID:284C,bguid:F692C420E06611E4BB0CE7FDC5486EA5mon
DAMAC-CME-ANOUD#terminal imon
                          ^
% Invalid input detected at '^' marker.
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
Apr 12 16:22:44.089: //-1/xxxxxxxxxxxx/LPCOR/lpcor_get_index_by_name:
   lpcor endusers
Apr 12 16:22:44.089: //-1/xxxxxxxxxxxx/LPCOR/lpcor_get_index_by_name:
   lpcor endusers index 10
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#show debug
VOIP LPCOR:
debug voip lpcor error call is ON (filter is OFF)
debug voip lpcor error call informational is ON (filter is OFF)
debug voip lpcor error software is ON
debug voip lpcor error software informational is ON
debug voip lpcor detail is ON (filter is OFF)
debug voip lpcor function is ON (filter is OFF)
debug voip lpcor inout is ON (filter is OFF)
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
Apr 12 16:23:22.889: %VOIPAAA-5-VOIP_CALL_HISTORY: CallLegType 1, ConnectionId FBA1532AE06611E4BB10E7FDC5486EA5, SetupTime 16:22:44.089 UTC Sun Apr 12 2015, PeerAddress 4130, PeerSubAddress , DisconnectCause 10 , DisconnectText normal call clearing (16), ConnectTime 16:23:02.009 UTC Sun Apr 12 2015, DisconnectTime 16:23:22.889 UTC Sun Apr 12 2015, CallOrigin 2, ChargedUnits 0, InfoType 2, TransmitPackets 0, TransmitBytes 0, ReceivePackets 1038, ReceiveBytes 166080
Apr 12 16:23:22.889: %VOIPAAA-5-VOIP_FEAT_HISTORY: FEAT_VSA=fn:TWC,ft:04/12/2015 16:22:44.093,cgn:4130,cdn:,frs:0,fid:2600,fcid:FBA1532AE06611E4BB10E7FDC5486EA5,legID:284D,bguid:FBA1532AE06611E4BB10E7FDC5486EA5
Apr 12 16:23:22.905: %VOIPAAA-5-VOIP_CALL_HISTORY: CallLegType 1, ConnectionId FBA1532AE06611E4BB10E7FDC5486EA5, SetupTime 16:22:57.795 UTC Sun Apr 12 2015, PeerAddress 0097150107659, PeerSubAddress , DisconnectCause 10 , DisconnectText normal call clearing (16), ConnectTime 16:23:02.015 UTC Sun Apr 12 2015, DisconnectTime 16:23:22.905 UTC Sun Apr 12 2015, CallOrigin 1, ChargedUnits 0, InfoType 2, TransmitPackets 1038, TransmitBytes 174384, ReceivePackets 1043, ReceiveBytes 166880
Apr 12 16:23:22.905: %VOIPAAA-5-VOIP_FEAT_HISTORY: FEAT_VSA=fn:TWC,ft:04/12/2015 16:22:57.785,cgn:4130,cdn:0097150107659,frs:0,fid:2601,fcid:FBA1532AE06611E4BB10E7FDC5486EA5,legID:284E,bguid:FBA1532AE06611E4BB10E7FDC5486EA5
Apr 12 16:23:25.317: //-1/xxxxxxxxxxxx/LPCOR/lpcor_get_index_by_name:
   lpcor endusers
Apr 12 16:23:25.317: //-1/xxxxxxxxxxxx/LPCOR/lpcor_get_index_by_name:
   lpcor endusers index 10
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#un all
All possible debugging has been turned off
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#!ok just send me these logs
DAMAC-CME-ANOUD#!i have to move from here
Apr 12 16:24:02.153: %VOIPAAA-5-VOIP_CALL_HISTORY: CallLegType 1, ConnectionId 14343755E06711E4BB16E7FDC5486EA5, SetupTime 16:23:25.323 UTC Sun Apr 12 2015, PeerAddress 4130, PeerSubAddress , DisconnectCause 10 , DisconnectText normal call clearing (16), ConnectTime 16:23:43.393 UTC Sun Apr 12 2015, DisconnectTime 16:24:02.153 UTC Sun Apr 12 2015, CallOrigin 2, ChargedUnits 0, InfoType 2, TransmitPackets 0, TransmitBytes 0, ReceivePackets 930, ReceiveBytes 148800
Apr 12 16:24:02.153: %VOIPAAA-5-VOIP_FEAT_HISTORY: FEAT_VSA=fn:Tnow
DAMAC-CME-ANOUD#\WC,ft:04/12/2015 16:23:25.321,cgn:4130,cdn:,frs:0,fid:2602,fcid:14343755E06711E4BB16E7FDC5486EA5,legID:2850,bguid:14343755E06711E4BB16E7FDC5486EA5
Apr 12 16:24:02.169: %VOIPAAA-5-VOIP_CALL_HISTORY: CallLegType 1, ConnectionId 14343755E06711E4BB16E7FDC5486EA5, SetupTime 16:23:39.169 UTC Sun Apr 12 2015, PeerAddress 0097150107659, PeerSubAddress , DisconnectCause 10 , DisconnectText normal call clearing (16), ConnectTime 16:23:43.389 UTC Sun Apr 12 2015, DisconnectTime 16:24:02.169 UTC Sun Apr 12 2015, CallOrigin 1, ChargedUnits 0, InfoType 2, TransmitPackets 930, TransmitBytes 156240, ReceivePackets 937, ReceiveBytes 149920
Apr 12 16:24:02.169: %VOIPAAA-5-VOIP_FEAT_HISTORY: FEAT_VSA=fn:TWC,ft:04/12/2015 16:23:39.169,cgn:4130,cdn:0097150107659,frs:0,fid:2603,fcid:14343755E06711E4BB16E7FDC5486EA5,legID:2851,bguid:14343755E06711E4BB16E7FDC5486EA5

We have come across this issue today in 10.9.5 (so affects 10.9.4 as well) but it was occurring in Sydney as well with a client and for me in Melbourne.

Jabber to 9971 CME no video via CUBE

I can make calls Jabber sotphone/Jabber softphone with video fine.
I can make calls Jabber softphone to same cluster CUCM 9971 with video fine.
When I make a call Jabber Softphone call via CUCM SIP Trunk to CUBE to CME 9.1 15.2 I get no video. Any ideas? Is there a setting for RTCP on a sip trunk or dial-peer or somewhere else in CUBE or in CME? As I saw a similar issue on NetPro resolve that.
CUCM 9.1
CUE IOS 15.1
CME 9.1 / IOS 15.2M5
9971 SIP9.4.1.9
Jabber for Windows 9.7.2

Hello!
Does video work inside CME?
Did you configure 'video codec' inside dial-peer?
Regards,
Kirill

SIP to SIP Call Failures on CME to CME - sip-ua conflict/issue?

Hi,
I have two existing CME systems which I wish to allow internal calls between. These calls will go over an IPSec VPN. However the calls are failing.
Phones DN22xx - London CME 2801 - PIX505 --- Internet ---ASA5505 - India CME 2801 - Phones DN400x
I have configured dial peers on both CME's and the IPSec VPN. I can ping between both systems. The VPN allows traffic between the interface IP's of the CME systems only.
London CME (local SCCP phones 22xx):
interface FastEthernet0/0.100
encapsulation dot1Q 100 native
ip address 10.0.10.250 255.255.255.0
voice class codec 101
codec preference 1 g729r8
codec preference 2 g711ulaw
codec preference 3 g711alaw
dial-peer voice 25 voip
description *** SIP Peer to India ***
answer-address 400.
destination-pattern 400.
voice-class codec 101
session protocol sipv2
session target ipv4:192.168.15.10
incoming called-number 400.
no vad
India CME (Local SSCP phones 400x):
interface FastEthernet0/0
ip address 192.168.15.10 255.255.255.0
voice class codec 100
codec preference 1 g729r8
codec preference 2 g711ulaw
codec preference 3 g711alaw
dial-peer voice 10 voip
description *** SIP Peer to London UK ***
answer-address 22..
destination-pattern 22..
voice-class codec 100
session protocol sipv2
session target ipv4:10.0.10.250
incoming called-number 22..
no vad
The CME system at India also has an existing SIP dial peer to a service provider and has sip-ua configured (username, password, realm and registrar).
A call from India (4005) to London (DN2207) fails, the ccsip debug attached. I'm assuming its because the sip-ua configuration is being used for these calls to when I don't want it to be. The from field shows âFrom: <sip:[email protected]â when I need this to be the internal IP 192.168.15.10.
Can anyone offer any assistance with this?
Regards,
Chris

Hi,
thanks for your input however thats not the problem. 201.196.128.56 isn't an address on the router, it only has one IP and its 192.168.15.10.
The 201.196.128.56 address is the NAT'd address on the firewall. So that when a SIP call is made to the internet with sip-ua the from address is the public IP.
Chris

CME 7.1 with SCCP 7940G phones and SIP connection to a VOIP provider - inbound outbound fails

Here's a quick and dirty diagram of a CME 7.1 configuration. The phone can all call each other but something is not quite right with the SIP provider. The registrar and SIP registration pieces are working but most of the configuration examples that I've seen make me think that the CME router was being used as the edge device to the internet. From my drawing, you can see that is not the case here. My edge device is a Cisco ASA5505 with 9.2.x software running. I might be missing something in the SIP gateway knowledge department. Without diving into the configuration, I'm wondering if SIP messages are failing for calls because of NAT'ing? Trying to do searches has been tricky because I keep running into information that is more about setting up CME for SIP phones or just getting SIP to work between CME and a SIP provider. I have that part working. I'm just a bit unsure about how an SCCP 7940G gets an outbound call or even gets one to come in.
When I dial from my cell phone to the pilot number, there are no rings, it just goes to the VOIP provider's voice mail. When I try to dial out, I get a fast busy.
So, is NAT a consideration? Will the SIP gateway set up a call (forward) via the pre-established SIP connection? Yeah, I do sound like a newb.
If anyone has good information about, let's say, an inbound call and how that traffic flow works.
Thanks!

Have you configured your ASA to either NAT the IP address of the CME router or to do port forwarding for port 5060?

CISCO Jabber 8.6.2 and CME 8.6

Hello,
I want to use Cisco Jabber 8.6.2 with Call manager Express 8.6
I configured the IPhone on CME and is working ok on local wireless LAN,
When I’m using the VPN I can place call's inside the network but I can't use on outside line. I have no sound.
Also if I place a call and the I end the call, the called phone rings and is not stopping. So I think is a call disconnect problem.
Bellow is a part of the configuration.
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
h323
call start slow
sip
session transport tcp
registrar server
voice class codec 1
codec preference 1 g729r8
voice class custom-cptone romania
dualtone busy
frequency 450
cadence 170 170
dualtone disconnect
frequency 450
cadence 170 170
voice register global
mode cme
source-address 10.12.4.252 port 5060
max-dn 10
max-pool 10
authenticate register
hold-alert
tftp-path flash:
create profile sync 0000545624458818
voice register dn 1
number 5146
call-forward b2bua all 5108
call-forward b2bua busy 5160
call-forward b2bua noan 5160 timeout 18
name Ioan Stanciu
shared-line
label Ioan
voice register pool 10
registration-timer max 720 min 660
id mac 68A8.6D91.3FE0
session-transport tcp
type CiscoMobile-iOS
number 1 dn 1
username 5146 password 5146
voice-port 0/0/0
supervisory disconnect dualtone mid-call
supervisory custom-cptone romania
no battery-reversal
timeouts call-disconnect 2
timeouts wait-release 2
connection plar opx immediate 5150
caller-id enable
voice-port 0/0/1
supervisory disconnect dualtone mid-call
supervisory custom-cptone romania
no battery-reversal
timeouts call-disconnect 2
timeouts wait-release 2
connection plar opx immediate 5150
caller-id enable
dspfarm profile 1 conference
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
codec g722-64
maximum sessions 1
associate application SCCP
dial-peer voice 2 pots
preference 2
destination-pattern 9T
incoming called-number .
direct-inward-dial
port 0/0/1
telephony-service
video
maximum bit-rate 256
no auto-reg-ephone
em logout 0:0 0:0 0:0
max-ephones 25
max-dn 120
ip source-address 10.12.4.252 port 2000
max-redirect 7
auto assign 1 to 8
service phone videoCapability 1
service phone paramEdibility 1
no service directed-pickup
timeouts interdigit 6
timeouts ringing 15
Any idea what can be the problem ?

Hello,
Is the connect over 3G enabled on your iPhone? Try this.
Btw, what have you put as device ID on your iPhone? the MAC address?
To my knowledge, the device ID is TCTXXX for CUCM, there is no info related on how to add it for CME 8.6
Regards

Jabber Client for CME license

i have CME 2921 ISR router and i need to implement cisco jabber client for 50 user
is this needs a license or not
if yes, please provide me how to orser it and what is its part number
thank you very much

If you are going to use Jabber, that counts like a regular SIP Phone, just like 7945, or 99XX phones, you don't need an additional license:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmelabel.html#wp1058022
HTH
Jorge Armijo
Please remember to rate helpful responses and identify helpful or correct answers.

CME 8.6 Call Disconnect Issue

Hi All -
My company has a small CME deployment in Australia and yesterday the local network resource made some changes to the network related to other work but following their changes, a very weird problem is occuring on the system.
The site has 4 POTS lines coming in on a VIC-4FXO card and when calls come in, the IP phones ring but as soon as they try to answer the call disconnects. I was suspecting a codec negotiation issue but since the call is coming from the PSTN, I'm not real sure because it should be G.711 which the IP phones should handle fine.
my ccapi debugs show the call disconnecting with the typical 16 cause code (normal clearing). I can provide any part of the config or debug output required for some ideas.....
I appreciate the help.
Tom

Hey Paolo,
I am facing the same issue but only with a single user.
Using ISDN2e (bri). She answers the call and it disconnects, can you please shine some light please
Please rate the post
Shibly Ibrahim

CME, Call forward to CUE from CCM IP phone

I want to call forward the call from CCM IP phone to CME ephone's voicemail which setup in CUE. works okay between CME ephones. configured voice service as follows but no luck. what did I missing to implement?
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service h450.2
no supplementary-service h450.3
-CCM4.1.3, configured H225 trunk. leave uncheck the MTP on the trunk device
-gatekeeper to connect between CCM and CME
-CME3.3, h323 to gatekeeper and sip to CUE
-CUE2.1
Thanks in advance,

It works by restart the CME router and have a question the sip-ua output. I have two media streams but the 2nd shows "STREAM_IDLE". I think this is for g729 connected to CCM via h323 gk. Can I get an explanation why?
CME#sh sip-ua calls
SIP UAC CALL INFO
Call 1
SIP Call ID : [email protected]
State of the call : STATE_ACTIVE (7)
Substate of the call : SUBSTATE_NONE (0)
Calling Number : 4083132006
Called Number : 4211
Bit Flags : 0x101A0030 0x100000 0x500
CC Call ID : 95
Source IP Address (Sig ): 10.253.66.254
Destn SIP Req Addr:Port : 10.253.66.2:5060
Destn SIP Resp Addr:Port: 10.253.66.2:5060
Destination Name :
Number of Media Streams : 2
Number of Active Streams: 1
RTP Fork Object : 0x0
Media Stream 1
State of the stream : STREAM_ACTIVE
Stream Call ID : 95
Stream Type : voice-only (0)
Negotiated Codec : g711ulaw (160 bytes)
Codec Payload Type : 0
Negotiated Dtmf-relay : inband-voice
Dtmf-relay Payload Type : 0
Media Source IP Addr:Port: 10.253.66.254:16998
Media Dest IP Addr:Port : 10.253.66.2:16904
Orig Media Dest IP Addr:Port : 0.0.0.0:0
Media Stream 2
State of the stream : STREAM_IDLE
Stream Call ID : -1
Stream Type : voice+dtmf (1)
Negotiated Codec : No Codec (0 bytes)
Codec Payload Type : 255 (None)
Negotiated Dtmf-relay : inband-voice
Dtmf-relay Payload Type : 0
Media Source IP Addr:Port: 10.253.66.254:17120
Media Dest IP Addr:Port : 0.0.0.0:0
Orig Media Dest IP Addr:Port : 0.0.0.0:0
Number of SIP User Agent Client(UAC) calls: 1
SIP UAS CALL INFO
Number of SIP User Agent Server(UAS) calls: 0
CME#sh sccp connections
sess_id conn_id stype mode codec ripaddr rport sport
1 2 xcode sendrecv g711u 10.253.66.254 2000 16518
1 1 xcode sendrecv g729 10.253.66.254 2000 17620
Total number of active session(s) 1, and connection(s) 2

CME-BACD

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