CUCM Configuration

Hi Everyone
I worked for a company in the past that you were able to dial *12345 as an example and by dialing that it would take you right to a users VM. I want to implement this feature at my new job and I wanted to know if someone could tell me what steps I would take to get this feature setup.
Eric

Hi Eric,
Just to add a note to the good tips from my friends Aman & Suresh (+5 each!)
I think the doc attached in the url below explains the steps you are looking for
http://www.cisco.com/c/en/us/support/docs/voice-unified-communications/unified-communications-manager-callmanager/26862-transfer-voicemail.html
Cheers!
Rob

Similar Messages

What is the best way to capture CUCM configurations?

Rather than clicking though the drop down menus in CUCM, what is the most efficient way to capture a complete configuration settings in CUCM? I know some of the settings can be done using BAT, but what about other settings like Enterprise Parameters, DHCP server/subnet, etc.? If the captured configurations are in human readable format will be even better.

[+5] to Rob and Brandon for sharing more info.
Hi Brandon,
I downlaoded the latest software cipinventory-v0.11 from the site and unzipped it .I could find four files and not any exe .I have attached the snapshot.
when I click on .pl file it asks which program to be used for opening.
Am I following the correct procedure or missing something?
regds,
aman

How to Backup CUCM Configuration

Dears,
Kinldy i have a UCS-C220M3 box using WMWare ESXi 5 installed with CUCM 9.1 that i have an issue that required to re-install the CUCM again. My qurey is that how i can copy the configuartions i did to re-call them back (IP phones devcies, Trunks, Route Calls, and others) kinldy adbvise?
Humam

Hi
In addition to the useful infromation as usual which Eng.Aman shared with you. Please find the below steps:-
Step 1
-On The Cisco Unified Communications System navigate to the Disaster Recovery System page and proceed to login.
Step 2
-Select the Backup tab
-Select the Backup Device Tab
-Click either "+" sign or the "Add New" button
Step 3
-Enter a name for your Backup device name*
-Select the Network Directory option
-For the Server name Enter the IP address of your sftp server
-For the Path name if everything on your Freesftpd server was left as default use \ otherwise
make the proper changes.
-For the username, use the username of the SFTP user that you just created in FreeFTPd, in my case is test
-For the password, use the password of the user that you have created in FreeFTPd.
-For the number of Backups to store on Network Directory I ussually select 3, but I leave this up to your discretion.
-Select the Save button.
*If everything went well the status alert message should now show Update succesful
Step 4
-Select the Backup tab
-Select the Scheduler tab
-Click either the "+" sign or the "Add New" button
Step 5
-For the Schedule Name enter an appropiate name for this operation
-For the Device Name select your newly created Backup Device
-Select the appropriate feature
-Proceed to enter a Backup start Date and time , this will be the date that you want the backup/schedule process to begin/start, also
keep in mind that time that you select here, it will be the time that the backup process will start according to your selections that you
make below.
-Select the Frequency* Basically when do you want to run a backup operation.
Once
Daily
Weekly (Mon, Tue, Web, Thur, Fri, Sat, Sun)
Monthly
-Click The Save Button
-Click the Enable Schedule button to enable this backup schedule, the status message alert will now show Status Enabled.
Please also find the below link
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_configuration_example09186a0080ab9fc0.shtml
Thank you
please rate all useful information

Jabber on Android and CUCM 8.6.2

Hello.
We are trying to use jabber on a galaxy android tablet, but we can't register client to CUCM 8.6.2.
I've followed this guide:
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_configuration_example09186a0080bb5104.shtml
for cucm configuration, I've installed cop files, restarted publisher and subscriber, and also tomcat service.
When we start jabber on galaxy tablet, we insert the two parameters needed (ip address of cucm and device name), but after a while it return an error message that say us to control configuration.
Tablet is connected to wifi network and can reach CUCM via ping or web page.
Thanks for any suggestion.
Daniele

Hi
Have you added the 'Dual mode for Android' device to CUCM? If so, perhaps screen grab your config.
Also unless you have other wireless voice devices already, verify that your wireless LAN can route to the CUCMs and can access TFTP and other ports on that subnet.
Aaron

CUE 8.5.1 NME-CUE web page setup no setup initialization and error: Login to CUCME failed

When first login in via the web page. When going to Configure menu and choosing CUCME to enter it manually, I get:
Error: Login to CUCME failed with the new values. Check the new CUCME configuration and enter the correct values.
hostname: 172.23.0.1
web user name: admin
web password: cisco
Sip gateway hostname: 172.23.0.1
ccn reporting historical
database local
description "se-172-23-0-2"
end reporting
ccn subsystem sip
gateway address "172.23.0.1"
mwi sip unsolicited
end subsystem
BR2-ROUTER#sh run
Building configuration...
Current configuration : 5264 bytes
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname BR2-ROUTER
boot-start-marker
boot-end-marker
card type t1 0 3
logging message-counter syslog
logging buffered 51200 warnings
no aaa new-model
clock timezone MST -7
clock summer-time MDT recurring
network-clock-participate wic 3
dot11 syslog
ip source-route
ip cef
ip dhcp excluded-address 172.21.0.1 172.21.0.49
ip dhcp excluded-address 172.21.0.59 172.21.0.254
ip dhcp excluded-address 172.20.0.1 172.20.0.10
ip dhcp pool CME
   network 172.21.0.0 255.255.255.0
   option 150 ip 172.21.0.1
   default-router 172.21.0.1
ip dhcp pool LAPTOPS
   network 172.20.0.0 255.255.255.0
   default-router 172.20.0.2
   dns-server 10.10.10.1
no ip domain lookup
ip domain name wilson.com
no ipv6 cef
multilink bundle-name authenticated
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 h225-notify cid-update
sip
bind control source-interface GigabitEthernet0/0.20
bind media source-interface GigabitEthernet0/0.20
registrar server expires max 600 min 60
voice register global
mode cme
source-address 172.21.0.1 port 5060
max-dn 4
max-pool 4
authenticate register
timezone 12
time-format 24
date-format YY-M-D
voicemail 3600
tftp-path flash:
create profile sync 0021447056000116
ntp-server 174.137.67.50 mode directedbroadcast
voice register dn 1
number 3006
call-forward b2bua busy 3600
call-forward b2bua mailbox 3006
call-forward b2bua noan 3600 timeout 12
name rp-sip-1-16
label SIP 511-5016
mwi
voice register pool 1
id mac FCFB.FBCA.30CE
type 7965
number 1 dn 1
dtmf-relay rtp-nte
username 3006 password cisco
description 687-3006
codec g711ulaw
voice-card 0
username admin privilege 15 secret 5 $1$..D.$orbTsqgPSvNkMpfjjkg5q.
archive
log config
hidekeys
controller T1 0/3/0
cablelength long 0db
controller T1 0/3/1
cablelength long 0db
interface Loopback0
ip address 172.23.0.1 255.255.255.252
ip ospf network point-to-point
interface GigabitEthernet0/0
description $ETH-LAN$$ETH-SW-LAUNCH$$INTF-INFO-GE 0/0$
no ip address
duplex auto
speed auto
interface GigabitEthernet0/0.10
encapsulation dot1Q 10 native
ip address 172.20.0.2 255.255.255.0
interface GigabitEthernet0/0.20
encapsulation dot1Q 20
ip address 172.21.0.1 255.255.255.0
interface GigabitEthernet0/0.30
encapsulation dot1Q 30
ip address 172.22.0.1 255.255.255.0
interface GigabitEthernet0/1
ip address 192.168.1.138 255.255.252.0
duplex auto
speed auto
interface Integrated-Service-Engine1/0
ip unnumbered Loopback0
service-module ip address 172.23.0.2 255.255.255.252
service-module ip default-gateway 172.23.0.1
no keepalive
ip forward-protocol nd
ip route 172.23.0.2 255.255.255.255 Integrated-Service-Engine1/0
ip http server
ip http access-class 23
ip http authentication local
no ip http secure-server
ip http timeout-policy idle 60 life 86400 requests 10000
ip http path flash:/gui
access-list 23 permit 10.10.10.0 0.0.0.7
nls resp-timeout 1
cpd cr-id 1
control-plane
ccm-manager fax protocol cisco
mgcp fax t38 ecm
dial-peer voice 3600 voip
destination-pattern 36..
session protocol sipv2
session target ipv4:192.168.1.144
dtmf-relay sip-notify
codec g711ulaw
no vad
sip-ua
retry invite 3
timers trying 400
mwi-server ipv4:192.168.1.144 expires 3600 port 5060 transport udp
gatekeeper
shutdown
telephony-service
no auto-reg-ephone
em logout 0:0 0:0 0:0
max-ephones 10
max-dn 10 no-reg both
ip source-address 172.23.0.1 port 2000
voicemail 3600
max-conferences 8 gain -6
call-forward pattern .T
web admin system name admin password cisco
dn-webedit
transfer-system full-consult
transfer-pattern .T
create cnf-files version-stamp Jan 01 2002 00:00:00
ephone-dn 1
number 3007
description 687-9898-3007
name Vatos locos
call-forward busy 3600
call-forward noan 3600 timeout 12
ephone-dn 2
number 3008
description 687-9898-3008
name Vatos locos2
call-forward busy 3600
call-forward noan 3600 timeout 12
ephone-dn 3 octo-line
number 3009
huntstop channel 6
ephone-dn 4
number 7999....
mwi on
ephone-dn 5
number 7998....
mwi off
ephone 1
device-security-mode none
description TESTTTTT
mac-address FCFB.FBCA.3406
max-calls-per-button 5
busy-trigger-per-button 4
type 7965
button 1:1 2:3
ephone 2
device-security-mode none
description TESTTTTT
mac-address FCFB.FBCA.3030
max-calls-per-button 4
busy-trigger-per-button 3
type 7965
button 1:2 2:3
line con 0
exec-timeout 0 0
logging synchronous
login local
line aux 0
line 66
no activation-character
no exec
transport preferred none
transport input all
transport output pad telnet rlogin lapb-ta mop udptn v120
line vty 0 4
access-class 23 in
privilege level 15
login local
transport input telnet
line vty 5 15
access-class 23 in
privilege level 15
login local
transport input telnet
scheduler allocate 20000 1000
ntp server 174.137.67.50
end
BR2-ROUTER#
Apr 12 2011 16:23:12 gui/admin_user.js
122     585532 Mar 30 2011 05:48:46 phone/7975/cnu75.8-3-2-27.sbn
123    2453636 Mar 30 2011 05:48:56 phone/7975/cvm75sccp.8-3-2-27.sbn
124     326315 Mar 30 2011 05:48:58 phone/7975/dsp75.8-3-2-27.sbn
125     557786 Mar 30 2011 05:49:00 phone/7975/jar75sccp.8-3-2-27.sbn
126        638 Mar 30 2011 05:49:02 phone/7975/SCCP75.8-3-3S.loads
127        642 Mar 30 2011 05:49:02 phone/7975/term75.default.loads
128          0 Mar 30 2011 05:49:02 phone/7941-7961
129    2494499 Mar 30 2011 05:49:12 phone/7941-7961/apps41.8-3-2-27.sbn
130     547146 Mar 30 2011 05:49:16 phone/7941-7961/cnu41.8-3-2-27.sbn
131       2340 Apr 02 2011 03:55:02 April012011.txt
132       3579 Apr 12 2011 03:52:42 softkeyDefault_kpml.xml
133         69 Apr 12 2011 03:52:40 syncinfo.xml
134       2682 Apr 12 2011 03:52:42 SEPFCFBFBCA30CE.cnf.xml
135       1882 Apr 12 2011 03:52:42 SIPDefault.cnf
136       3613 Apr 12 2011 03:52:42 softkeyDefault.xml
137       3987 Apr 12 2011 16:23:10 gui/admin_user.html
138       1029 Apr 12 2011 16:23:14 gui/CiscoLogo.gif
139        617 Apr 12 2011 16:23:14 gui/CME_GUI_README.TXT
140        953 Apr 12 2011 16:23:14 gui/Delete.gif
141      16344 Apr 12 2011 16:23:14 gui/dom.js
142        864 Apr 12 2011 16:23:16 gui/downarrow.gif
143       6146 Apr 12 2011 16:23:16 gui/ephone_admin.html
144       4558 Apr 12 2011 16:23:16 gui/logohome.gif
145       3866 Apr 12 2011 16:23:16 gui/normal_user.html
146      78428 Apr 12 2011 16:23:18 gui/normal_user.js
147       1347 Apr 12 2011 16:23:18 gui/Plus.gif
148        843 Apr 12 2011 16:23:18 gui/sxiconad.gif
149        174 Apr 12 2011 16:23:18 gui/Tab.gif
150       2431 Apr 12 2011 16:23:20 gui/telephony_service.html
151        870 Apr 12 2011 16:23:20 gui/uparrow.gif
152       9968 Apr 12 2011 16:23:20 gui/xml-test.html
153       3412 Apr 12 2011 16:23:20 gui/xml.template

Fixed. Routing issue:
Routing issue:
ip http access-class 23 !!!!!! Preconfigured from Factory
ip http authentication local
no ip http secure-server
ip http timeout-policy idle 60 life 86400 requests 10000
ip http path flash:/gui
access-list 23 permit 10.10.10.0 0.0.0.7 !!!!!! Preconfigured from Factory
To fix
No ip http access-class 23

CUCM 8.6 Dropped call transfers involving SIP phones

Hi All,
I am a developer who has been tasked with figuring out why call transfers are being dropped by Cisco CUCM when the original call comes from a SIP phone. This scenario works:
Cisco phone calls another Cisco phone, which transfers the original call to a SIP phone
These scenarios do not work:
SIP phone calls Cisco phone, which transfers the original call to another Cisco phone
SIP phone calls Cisco phone, which transfers the original call to another SIP phone
I have researched the Call Manager traces to the best of my ability, and I see some info in there that could potentially point to the source of the problem. I am just unable to understand what the trace means:
10:23:08.672 |//SIP/SIPCdpc(1,74,2342)/ci=24377698/ccbId=175645/scbId=0/active_CcDisconnReq: ccDisconnReq.onBehalfOf=Media : ccDisconnReq.s.sv=2 : ccDisconnReq.c.cv=47 |1,100,63,1.93259^10.10.10.85^*
10:23:08.672 |//SIP/Stack/Info/0x0/sipConstructContainerContext #### Created container=0xb0b42f58|1,100,71,1.1^*^*
10:23:08.672 |//SIP/SIPCdpc(1,74,2342)/ci=24377698/ccbId=175645/scbId=0/appendReasonHdr: appendReasonHdr - Invalid Disconnect Cause(cause=47), No Reason Header Appended|1,100,63,1.93259^10.10.10.85^*
10:23:08.672 |//SIP/SIPCdpc(1,74,2342)/ci=24377698/ccbId=175645/scbId=0/appendRPIDHdrForOriginalCalledParty: SIP device does not Support Orig Dialled Phone nego: 0|1,100,63,1.93259^10.10.10.85^*
I have been wondering whether this could be a codec issue, however the SIP phones we are using are configured with the following codecs:
G711U
G711A
G722
ILBC
GSM
and our SIP software is also set to accept the first codec offered by the remote side. It seems from the SIP client logs that G722 is being used as the codec to communicate with the Cisco phones, but perhaps I'm misinterpreting.
I have attached a CUCM trace of a call from a SIP phone (ext. 491) to a Cisco handset (ext. 170) where the Cisco handset attempts to transfer the call to another SIP phone (ext. 492). The trace snippet shown above is from this log.
I would really appreciate it if someone more experienced with VoIP/SIP/CUCM could take a look and offer any ideas on what the issue might be, and also how we might be able to address it. I can try to provide more info about our CUCM configuration if needed.
Thanks in advance!

Leslie, so here is what I found from the traces....
To understand the difference we need to understand how cucm performs call transfers from a sccp signalling point and a sip signalling point
SCCP
When the transfer key is pressed
1. CUCM sends a CloseReceiveChannel and StopMediaTransmission to the IP phone involved in active media (referenced by the callids)
NB, here CUCM updates the call state on the phone to a call state of 8 which is "Hold"
2.CUCM tells the held party to listen MOH from MOH server
3.CUCM establishes newcall leg with the intended transfered destination..Once this call is connected
4.CUCM receives a new Transfer instruction from the transferring phone to connect the held party
5..CUCM sends a CloseReceiveChannel between the held phone and MOH server (to tear down the media)
6. Next CUCM sends a CloseReceiveChannel and StopMediaTransmission to the transfering party & transfered party to remove Xferring party from call
7. finally CUCM sends OpenReceiveChannel between the original called party and the transfered party..and call is done
For SIP signalling. when the first transfer key is pressed
1. CUCM sends invite (re-invite) with an inactive SDP (a=inactive) to indicate a break in media path
2. CUCM sends a Delayed offer to insert MOH or to resume Media stream
NB: CUCM expects a sendrecv offer with SDP to the DO. (NB:if cucm gets an inactive offer SDP in the 200 OK instead of providing a send-recv offer SDP, the media path remains in an inactive state and causes calls to dropcall will drop),CUCM sends an ACK with sendonly to the 200 OK
3.CUCM establishes newcall leg with the intended transfered destination..Once this call is connected
4.CUCM receives a new Transfer instruction from the transferring phone to connect the held party
5. Next CUCM sends a re-invite with an inactive SDP to indicate a break in media path to MOH (in attempt to complete transfer)
6.Next CUCM sends an inactive SDP to indicate a break in media path between transfering party & transfered party to remove Xferring party from call
7. Next CUCM sends a DO re-invite to connect the transfered party. The far end then sends 200 OK with the required SDP to connect the call
Now having explained all of these, we need to look at where the call is failing for SIP-----SCCP----SIP calls without MTP
lets look at succesful SCCP-----SCCP-----SIP without MTP
Point 4 above
++++++++Extension 170 presses the transfer button to connect the two calls (Callid=24378483)+++++++++++++
(0003395) SoftKeyEvent softKeyEvent=4(Trnsfer) lineInstance=1 callReference=24378483
Point 5 above
++++Next CUCM closed the media between extension 160 and MOH server callid=24378480(this is the only active call on this callid)+++
(0003396) CloseReceiveChannel conferenceID=24378480 passThruPartyID=16777845. myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a89(10.10.10.137)
Point 6 Above
+++++Next cucm closes the call between extension 170 and 490 callid=(24378483)++++++++
(0003395) CloseReceiveChannel conferenceID=24378483 passThruPartyID=16777847. myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a8b(10.10.10.139)
(0003395) StopMediaTransmission conferenceID=24378483 passThruPartyID=16777847. myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a8b(10.10.10.139)
Point 6 above for the sip side (since the destination is SIP, to tear down media to SCCP phone, so as to connect the caller to the xfered party)
+++++++Next CUCM sends a re-invite with a=inactive SDP to the sip phone ++++++++++++
//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 62220 index 1890
[885626,NET]
INVITE sip:[email protected]:62220;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK23332dbee978
From: ;tag=192115~d8e94532-127d-4dca-bba0-64b1675da032-24378484
o=CiscoSystemsCCM-SIP 192115 2 IN IP4 10.10.10.195
s=SIP Call
c=IN IP4 0.0.0.0
m=audio 24560 RTP/AVP 9 101
a=rtpmap:9 G722/8000
a=ptime:20
a=inactive-----------------------------------------------------Inactive
Still part of Point 6 for SIP signalling
++++++++++++Next sip phone responds with a 200 OK recevonly SDP +++++++++++++++++++
//SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 10.10.10.104 on port 62220 index 1890 with 683 bytes:
[885628,NET]
SIP/2.0 200 OK
v=0
o=- 18077 11099 IN IP4 10.10.10.104
s=yasdjip
c=IN IP4 10.10.10.104
t=0 0
a=ptime:20
a=recvonly-------------------------------------a=recvonly
Finally Point 7 above..
+++++++++++++=Next cucm sends a DO re-invite to extension 492-sip phone++++++++++
//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 62220 index 1890
[885630,NET]
INVITE sip:[email protected]:62220;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK233534ffec4a
From: ;tag=192115~d8e94532-127d-4dca-bba0-64b1675da032-24378484
To: ;tag=5B0E9816C2CA6D70F3166FB972EDE4C2
+++++++Next we get a 200 OK from sip phone with sdp=sendrecv+++++++++=
/SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 10.10.10.104 on port 62220 index 1890 with 683 bytes:
[885634,NET]
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK233534ffec4a
Contact:
From: ;tag=192115~d8e94532-127d-4dca-bba0-64b1675da032-24378484
Call-ID: [email protected]
v=0
o=- 18077 11099 IN IP4 10.10.10.104
s=yasdjip
c=IN IP4 10.10.10.104
t=0 0
m=audio 16574 RTP/AVP 9 101
a=rtpmap:101 TELEPHONE-EVENT/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
+Now CUCM sends an OpenReceiveChannel and start media xmission to sccp phone (callid=24378480) with media parameters of sip phone++++++
(0003396) OpenReceiveChannel conferenceID=24378480 passThruPartyID=16777848 millisecondPacketSize=20 compressionType=6(Media_Payload_G722_64k) RFC2833PayloadType=101 qualifierIn=? sourceIpAddr=IpAddr.type:0 ipAddr:0x0a0a0a68000000000000000000000000(10.10.10.104). myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a89(10.10.10.137)
(0003396) startMediaTransmission conferenceID=24378480 passThruPartyID=16777848 remoteIpAddress=IpAddr.type:0 ipAddr:0x0a0a0a68000000000000000000000000(10.10.10.104)
remotePortNumber=16574 milliSecondPacketSize=20 compressType=6(Media_Payload_G722_64k) RFC2833PayloadType=101 qualifierOut=?. myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a89(10.10.10.137)
+++++++++++=Next Phone sends its ACK+++++++++++++++
(0003396) OpenReceiveChannelAck Status=0, IpAddr=IpAddr.type:0 ipAddr:0x0a0a0a89000000000000000000000000(10.10.10.137), Port=20352, PartyID=16777848
+++++++++++=Next CUCM sends ACK to 200 OK from SIP Phone+++++++++++
//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 62220 index 1890
[885635,NET]
ACK sip:[email protected]:62220;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK23366067b8c0
From: ;tag=192115~d8e94532-127d-4dca-bba0-64b1675da032-24378484
To: ;tag=5B0E9816C2CA6D70F3166FB972EDE4C2
Date: Tue, 19 Feb 2013 21:44:45 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 103 ACK
Allow-Events: presence
Content-Type: application/sdp
Content-Length: 237
v=0
o=CiscoSystemsCCM-SIP 192115 3 IN IP4 10.10.10.195
s=SIP Call
c=IN IP4 10.10.10.137
b=TIAS:64000
b=AS:64
t=0 0
m=audio 20352 RTP/AVP 9 101
a=rtpmap:9 G722/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Now at this point all is well...and the call is connected....
Now here is where the call is failing on the SIP-SCCP-SIP call without MTP
From Point 2 above, CUCM sends a DO to insert MOH, and then gets response, then sends an ACK to 200 Ok back to SIP Phone..
//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 53361 index 1810
[881160,NET]
ACK sip:[email protected]:53361;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK22035ecc1fcb
From: ;tag=190666~d8e94532-127d-4dca-bba0-64b1675da032-24378214
To: "492" ;tag=97C34E1FB9A11F83DD8D8F5BA4C87C57
Date: Tue, 19 Feb 2013 17:38:50 GMT
Call-ID: 1CCA5149B966DC89AE0F752B8EF86480BC7102DB
Max-Forwards: 70
CSeq: 102 ACK
o=CiscoSystemsCCM-SIP 190666 3 IN IP4 10.10.10.195
s=SIP Call
c=IN IP4 10.10.10.195---------------------------------------IP address of MOH server
t=0 0
m=audio 4000 RTP/AVP 0--------------------------------MOH port 4000
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendonly---------------------------------------------------------sendonly
+++++NOW Point 6 above (SIP) CUCM sends a=inactive to break media path to MOH server to connect caller and xfered party++++++
//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 53361 index 1810
[881161,NET]
INVITE sip:[email protected]:53361;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK22045bb7f918
From: ;tag=190666~d8e94532-127d-4dca-bba0-64b1675da032-24378214
To: "492" ;tag=97C34E1FB9A11F83DD8D8F5BA4C87C57
Date: Tue, 19 Feb 2013 17:39:04 GMT
Call-ID: 1CCA5149B966DC89AE0F752B8EF86480BC7102DB
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 103 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Remote-Party-ID: ;party=calling;screen=yes;privacy=off
Contact:
Content-Type: application/sdp
Content-Length: 164
v=0
o=CiscoSystemsCCM-SIP 190666 4 IN IP4 10.10.10.195
s=SIP Call
c=IN IP4 0.0.0.0--------------------------------------------------------------------Media IP is 0.0.0.0
t=0 0
m=audio 4000 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=inactive---------------------------------------------------------------------media inactive
At this point, we should get a response back from the sip phone...
and here is what we got..
++Trying which is expected++++
//SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 10.10.10.104 on port 53361 index 1810 with 331 bytes:
[881162,NET]
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK22045bb7f918
From: ;tag=190666~d8e94532-127d-4dca-bba0-64b1675da032-24378214
Call-ID: 1CCA5149B966DC89AE0F752B8EF86480BC7102DB
CSeq: 103 INVITE
To: "492" ;tag=97C34E1FB9A11F83DD8D8F5BA4C87C57
Content-Length: 0
++++++++Then we get a BYE+++++++++++++++
/SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 10.10.10.104 on port 53361 index 1810 with 576 bytes:
[881163,NET]
BYE sip:[email protected]:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.104:53361;branch=z9hG4bKa2vdQvR7J9OiMyjU;rport
Contact:
Max-Forwards: 70
From: "492" ;tag=97C34E1FB9A11F83DD8D8F5BA4C87C57
Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
Supported: replaces, path
User-Agent: Acrobits Softphone Business/2.4.8
To: ;tag=190666~d8e94532-127d-4dca-bba0-64b1675da032-24378214
Call-ID: 1CCA5149B966DC89AE0F752B8EF86480BC7102DB
CSeq: 3 BYE
Content-Length: 0
So this is the root cause of the problem. Your SIP phone does not know how to respond to multiple media break between it and the MOH server.
The difference between this and the succesful SCCP-SIP--SCCP, is that the held party was a sccp phone, hence the sip phone only has to process one a=inactive SDP message, where as in the SIP-SCCP-SIP, the help party was sip, so the sip phone has to process two a=inactive SDP messages
Now what is the difference when MTP is involved! A Big difference. MTP stays in the media path. So there is never a break in media and no inactive SDP attribute is sent. The flow looks like below:
for the initial call...The SIP phone sends its media to MTP and likewise the SCCP phone
SIP------Media------MTP------------Media-------SCCP Phone
When the new destination is dialled and transfer is commited,
SIP-------------media----MTP--------media---------MTP
The final invoite sent to connect 492 and 491 has MTP as the IP address to connect Media to.
++++++++Ivite to 492 ++++++++++++++
INVITE sip:[email protected]:61303;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK231a3b24b862
From: ;tag=192048~d8e94532-127d-4dca-bba0-64b1675da032-24378472
To: ;tag=78FF5BF6C019A55EA020B69BB6A767E2
Date: Tue, 19 Feb 2013 21:24:59 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 102 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Remote-Party-ID: ;party=calling;screen=yes;privacy=off
Contact:
Content-Type: application/sdp
Content-Length: 214
v=0
o=CiscoSystemsCCM-SIP 192048 1 IN IP4 10.10.10.195
s=SIP Call
c=IN IP4 10.10.10.195---------------------------------------------------------------the MTP ip address
t=0 0
m=audio 25038 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
+++++++Invite to 491 +++++++++++++++++
//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.94 on port 50376 index 1887
[885429,NET]
INVITE sip:[email protected]:50376;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK231b78d1b56
From: ;tag=192046~d8e94532-127d-4dca-bba0-64b1675da032-24378467
To: "491" ;tag=F13CE94DE942C47680356A647DC7F916
Date: Tue, 19 Feb 2013 21:24:59 GMT
Call-ID: AE7045FFB2D8D9C28D54651473A14A5D41B5B93C
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Remote-Party-ID: "Leslie2" ;party=calling;screen=no;privacy=off
Contact:
Content-Type: application/sdp
Content-Length: 237
v=0
o=CiscoSystemsCCM-SIP 192046 1 IN IP4 10.10.10.195
s=SIP Call
c=IN IP4 10.10.10.195----------------------------------------MTP
b=TIAS:64000
b=AS:64
t=0 0
m=audio 25030 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Wao! That was a long one isnt it...It was fun too.
So now you can look at your sip phones and see if they can accept two inactive sdp messages within the same call. That way you can remove MTP. otherwise you will have MTP involved in every single call involving a sip phone, even if they do not involve transfers
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"opportunity is a haughty goddess who waste no time with those who are unprepared"

UCCX Best Practice - UCM Agent Line Configuration documentation

With UCCX, I have always abided by some rules when it came to configuration of the agent's line in CUCM. At least to be a TAC supported solution anyway. For Example:
1. Agent Extension can not be shared
2. Agent Exension not part of any CUCM hunt group or call pickup group
3. Call Waiting Disabled Max/Busy 2/1
4. Agent Extension should not take inbound calls
5. Agent extensions not set to CFNA
6. etc....
I had someone ask me to back this up with some sort of documentation. I reviewed the UCCX 7.x SRND and could not find anywhere explicitly talking about CUCM configuration.
Does anyone know if this type of information is documented?
Thanks in advance,
Shane

Shane,
Look at the release notes for your version of UCCX. They typically have a
section called "Unsupported Features in Unified CM". There is also a
section on "Unsupported and Supported Actions" and general "Unsupported
Configurations in Cisco Unified CCX".
Release notes URL:
http://www.cisco.com/en/US/products/sw/custcosw/ps1846/prod_release_notes_li
st.html
You won't find the data in the SRND for whatever reason.
HTH.
Regards,
Bill
Please remember to rate helpful posts.
On 9/8/10 5:40 PM, "shane.orr"

Doubts with ICM / CUCM

Hi All,
I need your ideas to implement the below scenario for one of our customer.
Customer has 4 location E.g., North , South , East and West.
There is a TFN which is mapped to a script and when the user dials this TFN and selects the option North or South or East or West the call should go to that region.
Scenario :
The script has IF condition which checks which region the user chooses then it sends the call to a Label ( Extension number ) to the CUCM.
For Eg., when user chose North Region and ICM send that call to CUCM Label and suppose the North region user is Busy or logged out then the call should go to South Region extension.
How to implement the above scenario , is there option in Script to check with CUCM whether the extension is busy or loggedout and send the call to other region or it can be done with CUCM configuration.
Regards
Sathya

Sathya,
In Target Requery, When your Peripheral fails to Deliver call to Target or Label, it will submit re route request to Router and Router will resume the script execution from failure node and takes the routing Up.
So lets assume that call comes to CVP on TFN, and hits ICM Script.
In script you play menu to caller and get the selection(south,north,west,east)
now based on selection, you are using label node to send LABEL back to CVP and the same label will point to device or CTI RP on CUCM using CVP static routes.
now in the label node, check the target requery check box. i have attached screen shot.
enabling this will Add failure path to label node, and you can use the failure path to send one more label back to CVP which points to some other zone.
The Label can be dynamic or static, however you wish.
i have also attached one sample script, where call comes DN, and i do send to VRU, and after successful operation i send one label back to CVP and if that fails i send one more label back to CVP.
i hope this clears your doubts.
Regards
Chintan
~please rate all helpful posts and mark the answer as correct if any

How to find the installed CUCM iso is restricted or unrestricted?

How to find the installed software in CUCM9.1(iso) is restricted or unrestricted??

Hi,
what u can do is u can login into OS administration through Putty . u get the details of VMWare .
Same can be compared and checked with OVA templates in README file
http://www.cisco.com/web/software/283088407/97505/cucm_9.1_vmv8_v1.7.ova.README1.txt
CUCM 7500 user node:
     Cisco Unified Communications Manager (CUCM) configuration that
     supports up to 7500 users per node.
     Details:
     Red Hat Enterprise Linux 5 (32-bit)
     CPU: 2 vCPU with 3600 MHz reservation
     Memory: 6 GB with 6 GB reservation
     Disk: 1 - 110 GB disk with pre-aligned disk partitions
you can see UNRST in attached snapshot.
regds,
aman

Meeting Place Configuration Doc

Hi
I am configuring Meeting Place 8.5. Can anyone please provide me Meeting Place 8.5 basic configuration guide. I have the cisco guide with me. I have created sip trunk in cucm and done basic configuration in Meeting Place server as per cisco guide. But when I test calling meeting place number, the call gets disconnected. Something somewhere I have missed. I have successfully imported the users to the meeting place server. Thank u.

Hi,
Are you using EMS or HMS? Can you make sure you have correct configuration on SIP trunk (CUCM side) and make sure on MP you have configured the correct CUCM configuration under SIP config.
Login to your MP as root and see check below:
1. cd /var/mp
2. ls -l
and see if you see cca directory or not? If not then looks like you configured the MP for HMS but you don't have hardware MCU or possibly MCU is not online or not configure properly.
If you see above directory do below:
cd /var/mp/cca
tail -f tvsip* | grep -e '--'
make a test call and you will see sometihng on your screen, capture the whole stuff and post here will look tell you what's going on but if you don't see any calls coming that means call is not even landing to MP then check your CUCM for possibly PT, CSS, RP, RP, Trunk etc.
Please let me know if you need any more info.
If you need immediate assistance kindly open a TAC case.
HTH
Arun

ATA190: Web GUI page options are greyed out

ATA190: Web GUI page options are greyed out and device failed registration to CM 10.5
do anyone have a resolution o this issue?

Options Greyed out issue can be due to the below Bug, which is listed as just cosmetic:
ATA190: Web GUI page options are greyed out
CSCur53864
Description
Symptom:
ATA190: Web GUI page options are greyed out.
Most of the options in the Web GUI of the ATA 190 are greyed out and cannot be configured. There is no option in CUCM configuration page or the ATA190 Web GUI to unlock the configuration options.
Here is a list of a few of the pages that are greyed out and do not allow any editing:
Network Setup > Basic Setup > Internet Settings
Network Setup > Basic Setup > Time Settings
Network Setup > Advanced Settings > VLAN
Network Setup > Advanced Settings > CDP & LLDP
Administration> Log > Log Module
Administration> Log > Log Setting
Administration> Log > Log Viewer
Conditions:
ATA 190 Devices running firmware version running 1.1.0 (006)
Workaround:
These options are not configurable and are supposed to be view only. The issue is cosmetic.
Most of the Configurable settings are available under the Voice Menu and Administrative Menu.
-Terry
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Weird behavior in Flash CS6

Flash Professional CS6
Windows 7 64 bit
These problems were not present yesterday but are today, I was told that a system admin may have fubarred me. Thanks, System Admin!
So before I put in the request for a system restore, let's see if we can fix it first.
Problems:
1. Canvas is not redrawing completely. Some symbols which should be there are not there. The layer is not hidden, the symbol does not have its alpha turned down, the scrubber is on a frame with content. There is no reason for it to be missing and yet it is.
2. General Textbox malaise. When attempting to enter text into a textbox, nothing happens at first. Then, suddenly, everything typed will explode onto the screen, and a big white rectangular artifact will appear behind them. It is very difficult to edit a textbox when you cannot see the changes you are making until many seconds later.
3. The blue rectangle that appears around selected objects does not appear. The layer is not locked, the object is not a shape, and the object is selected and can be moved, but the indication that it has become selected never appears.
4. Stream audio does not stream. This is probably the most problematic, as I need to sync some events to the audio. The Sync of the audio is set to Stream, and scrubbing the audio on the timelime produces that familiar garbley sound as normal, but simply playing (by pressing enter or the play button) yields no audio preview.
5. General white rectangle artifact malaise. They are everywhere.
I have tried resizing the window, restarting flash, and restarting the computer all to no avail. Flash is the only application running at this time, although on previous days I did have Google chrome and Audacity open simultaneously.

I've seen this happen before because of a CUCM configuration issue. Check whether the CUCM Calling Search Space for the UCCX CTI ports can call the partition of the agent's extension. My guess is that the call attempts to connect but CUCM will not allow it to go through. UCCX then marks the agent as "Not Ready" and drops the call back in queue.

Weird behavior in UCCX 7.0.1 SR05

Hi team,
I've got a weird behavior in a UCCX 7.01 SR05 installation.
All incomming calls which go to an agent who is set "ready" set the agent to state "not ready" and go to queue...
This behavior also occurs with standard script icd.aef.
Had anybody has same behavior in an UCCX 7 installation? Or can anybody help me? I checked my configuration twise and couldn't find
any issue.
Thanks a lot,
Tobias

I've seen this happen before because of a CUCM configuration issue. Check whether the CUCM Calling Search Space for the UCCX CTI ports can call the partition of the agent's extension. My guess is that the call attempts to connect but CUCM will not allow it to go through. UCCX then marks the agent as "Not Ready" and drops the call back in queue.

Attendant Console 6.1.2 (yeah, I know it is unsupported)

So knowing full well that this is an unsupported software, I am reaching out anyway. I find myself in a pickle at my new company. We are still running Call Manager 6.1.5.11900-13 (don't ask) and we have many users using Attendant Console.
Last week, the line status of all the consoles changed to blue question marks. The phone control function of the application works great, just no line status.
We have restarted AC and CTI services on entire cluster, pointed the AC clients to multiple servers for testing, and uninstalled and reinstalled the AC client. Tonight we are punting and rebooting the Publisher.
Anyone out there unfortunate enough to be running this, and maybe have a suggestion to fix?
A suggestion to upgrade would be sound, but won't be happening for now.
Commence the laughing!

They usually do not simply "die" like this. Id check for any replication issues with your cluster first off.
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a00809643e8.shtml
Next, I would simply build an entire new AC in CUCM and try to use this one. I have had luck in the past with this where the only way I could get it to work was to delete everything off it that AC used. Pilot points, RPs, etc. Then rebuild it.Sounds odd, yes.
I find it odd though it stopped working with no changes in the network, CUCM configurations, etc.

Ask the Expert: Cisco TelePresence for the Enterprise

Welcome to the Cisco® Support Community Ask the Expert conversation. This is an opportunity to learn and ask questions about Cisco Telepresence® for the enterprise.
Cisco experts Jaret, Fernando, and Fred will be covering all Cisco TelePresence products. Topics include Cisco TelePresence endpoints and TelePresence infrastructure such as the Cisco TelePresence Video Communication Server (VCS), Cisco Expressway Series, Cisco Unified Communication Manager (CallManager), Cisco TelePresence Servers (MSE 8710, on Virtual Machine, etc.), MCU (MSE 8510, etc.), Cisco TelePresence Management Suite (TMS), and all other Cisco TelePresence related devices.
Jaret Osborne is an 8-year Cisco Advanced Services veteran. In his Advanced Services tour, Jaret has covered all aspects of Cisco Unified Communications and TelePresence products, including both enterprise and service provider verticals. Most recently Jaret has been working with global service providers supporting their Cisco TelePresence as a Service offerings while also incubating new cloud services at Cisco.
Fernando Rivas is a Cisco Advanced Services NCE, starting in the Cisco Technical Assistance Center (TAC), 2007, on the Collaboration Technology Team mastering the Cisco Unified Communication technologies and specialized in call control CUCM,VCS) and conferencing (MeetingPlace, Telepresence). In 2011, he joined Cisco Advanced Services as a member of the Cisco Collaboration team and participated in several Cisco TelePresence and video-related technologies deployments. Currently he is a member of the Video Cloud Technology Team, supporting video exchanges in several and architecting new private video cloud solutions for large enterprises. Fernando holds a routing and switching CCIE® certification (22975).
Fred Mollenkopf is a Cisco Advanced Services Network consulting engineer working at Cisco for the last 7 years. Fred has led some of the largest Cisco Unified Communication and Collaboration deployments done for Cisco customers and partners. Over 15 years’ experience in data networking with a specialization in Cisco Unified Communications in 2004. Currently he is a member of the SP Video Advanced Services Team, supporting SP video exchanges and the Cisco Telepresence solutions. Fred maintains an active CCIE® in Voice (17521).
Remember to use the rating system to let Jaret, Fernando, and Fred know if you have received an adequate response.
Because of the volume expected during this event, Jaret, Fred, and Fernando might not be able to answer every question. Remember that you can continue the conversation in the Collaboration, Voice and Video Community, under the sub-community TelePresence, shortly after the event. This event lasts through August 15, 2014. Visit this forum often to view responses to your questions and the questions of other Cisco Support Community members.

Tenaro,
Additionally here are the most common login issues. Unfortunately this includes items related to Presence implementation but I commented where we did not use these in our lab setup for CUCM Phone Capabilities only.
Login Issues
Problem:
Jabber Unable to Sign-in Through MRA
Solution
This can be caused by a number of things, a few of which are outlined below.
1. Collaboration Edge SRV record not created and/or port 8443 unreachable
For a jabber client to be able to login successfully using MRA, a specific collaboration edge SRV record must be created and accessible externally. When a jabber client is initially started it will make server DNS SRV queries:
_cisco-uds : this SRV record is used to determine if a CUCM server is available.
_cuplogin : this SRV record is used to determine if an IM&P server is available.
_collab-edge : this SRV record is used to determine if MRA is available.
If the jabber client is started and does not receive an SRV answer for _cisco-uds and _cuplogin, and does receive an answer for _collab-edge then it will use this answer to try to contact the Expressway-E listed in the SRV answer.
The _collab-edge SRV record should point to the FQDN of the Expressway-E using port 8443. If the _collab-edge SRV is not created, or is not externally available, or if it is available, but port 8443 is not reachable, then the jabber client will fail to login.
2.  Unacceptable or No Available Certificate on VCS Expressway
After the jabber client has received an answer for _collab-edge, it will then contact the expressway using TLS over port 8443 to try to retrieve the certificate from the expressway to setup TLS for communication between the jabber client and the expressway.
If the Expressway does not have a valid signed certificate that contains either the FQDN or domain of the Expressway, then this will fail and the jabber client will fail to login.
If this is occurring, the you should use the CSR tool on the Expressway, which will automatically include the FQDN of the expressway as a Subject Alternative Name.
MRA requires secure communication between the Expressway-C and Expressway-E, and between the Expressway-E and external endpoints.
Expressway-C Server Certificate Requirements:
The Chat Node Aliases configured on the IM&P servers. This is required if you are doing XMPP federation. The Expressway-C should automatically include these in the CSR provided that an IM&P server has already been discovered on the Expressway-C.
The names in FQDN format of all Phone Security Profiles in CUCM configured for TLS and used on devices configured for MRA. This allows for secure communication between the CUCM and Expressway-C for the devices using those Phone Security Profiles.
Expressway-E Server Certificate Requirements:
All domains configured for Unified Communications. This includes the domain of the Expressway-E and C, e-mail address domain configured for Jabber, and any presence domains.
The Chat Node Aliases configured on the IM&P servers. This is required if you are doing XMPP federation.
The MRA Deployment guide describes this in greater detail on pages 17-18. (http://www.cisco.com/c/dam/en/us/td/docs/voice_ip_comm/expressway/config_guide/X8-1/Mobile-Remote-Ac...
Note: In our lab for testing Phone Capabilities only, we did not include the Chat Node Aliases in the certificate as we were not using IM&P.
3.  No UDS Servers Found in Edge Config
After the Jabber client successfully establishes a secure connection with the Expressway-E, it will ask for its edge config. This edge config will contain the SRV records for _cuplogin and _cisco-uds. If these SRV records are not returned in the edge config, then the jabber client will not be able to proceed with trying to login.
To fix this, make sure that _cisco-uds and _cuplogin SRV records are created internally and resolvable by the Expressway-C
More information on the DNS SRV records can be found on page 10 of the MRA deployment guide for X8.1.1 (http://www.cisco.com/c/dam/en/us/td/docs/voice_ip_comm/expressway/config_guide/X8-1/Mobile-Remote-Access-via-Expressway-Deployment-Guide-X8-1-1.pdf)
Note: In our lab for testing Phone Capabilities only, we did not include the DNS SRV for _cuplogin.
4.  The Expressway-C logs will indicate the following error: XCP_JABBERD Detail="Unable to connect to host '%IP%', port 7400:(111) Connection refused"
If Expressway-E NIC is incorrectly configured, this can cause the XCP server to not be updated. If the Expressway-E meets the following criteria, then you will likely have this issue:
Using a single NIC
Advanced Networking Option Key is installed
Use Dual Network Interfaces option is set to “Yes”
To correct this problem, change the “Use Dual Network Interfaces” option to “No”
The reason this is a problem is because the Expressway-E will be listening for the XCP session on the wrong network interface, which will cause the connection to fail/timeout. The Expressway-E listens on TCP port 7400 for the XCP session. You can verify this by using the netstat command from the VCS as root.
Note: We used a Dual Network Interface Expressway for testing but were not using XCP, so this was not applicable to us.
5.  VCE-E Server hostname/domain name does not match what is configured in the _collab-edge SRV.
If the Expressway-E Server hostname/domain name does not match what was received in the _collab-edge SRV answer, the jabber client will not be able to communicate to the Expressway-E. The Jabber client uses the xmppEdgeServer/Address element in the get_edge_config response to establish the XMPP connection to the Expressway-E.
This is an example of what the xmppEdgeServer/Address would look like in the get_edge_config response from the Expressway-E to the Jabber client:
<xmppEdgeServer>
<server>
<address>ott-vcse1.vcx.cisco.com</address>
<tlsPort>5222</tlsPort>
</server>
</xmppEdgeServer>
To avoid this, make sure that the _collab-edge SRV record matches the Expressway-E hostname/domain name. Enhancement CSCuo83458 has been filed for this.
Note: This was one of our issues when we first setup. We adjusted our Expressway-E to insure the below:
System > Administration > System Name this was the FQDN
System > DNS > System Host Name was the host portion of the FQDN
System > DNS > Domain Name was the domain portion of the FQDN
System > Clustering > Cluster Name (FQDN for Provisioning) was the FQDN
6. Unable to log into certain IM&P servers. VCS logs say "No realm found for host cups-example.domain.com, check connect auth configuration"
From the Expressway-E, go to Configuration -> Unified Communications -> IM&P Servers. Open each server and click "Save" again. Not sure exactly why this happens.
Note: This was N/A to our test and can be ignored with Phone Capabilities only.
Thanks
Fred

CUCM Configuration

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