Cisco CME licensing

Hello Cisco Team
I am planning to install 20 ip phones 7942 and 5 ip Communicators but I got some doubts in licensing,
First
What is the difference between FL-CME-SRST-25 and SW-CCME-UL-7942, do I have to buy both?
Can I take 5 licenses from FL-CME-SRST-25 to install 5 ip communicators?
Hope you can help me
Regards

Hi Hugo,
SW-CCME-UL-7942 is the IP phone CME license for 7942 model while FL-CME-SRST-25 is feature license of SRST and is for 25 seats.
for IP Communicator , u need to have separate licenses.Part Code would be SW-CCME-UL-IPCOMM= [you can get the same checked by Cisco AM]
All licenses are paper based.
you can refer the discussion
https://supportforums.cisco.com/thread/2075659
regds,
aman

Similar Messages

CUCM & CME licensing

Hello
I know this may seem a strange question but I am struggling to find an answer to it and am hoping this is the correct place to post the question and get an answer.
Our organisation currently has a CUCM cluster. When new phones are needed the organisation buys the phones and the required number of licenses for the phones.
Each of the 3 remote sites has a 2851 running in SRST mode if it looses contact with the CUCM servers.
I know all the theory about CUCM etc but I am new to administration on a live network. There are a couple of questions which have been asked lately that I am unclear about.
1. There are some plans to add 4 more remote sites to the system, the question has been asked if more phone licenses need to be bought. I have advised that licenses are bought with the phones (depending on the phone model). The phones are new not spares so they need the licenses.
2. If some of the original sites decided to break away from the current set-up and use their 2851 in CME mode will they need to buy more licenses for their phones. I was unclear of the position on this but I did not think it possible to transfer CUCM licenses to CME.
I am sorry if this is rather a basic question but although I have spent a bit of time looking through Cisco documentation on licensing I am still unclear on the answers to these questions.
I would appreciate any help on this or if someone could point me in the direction where I can find the information
Kind Regards

Thank you for the info but I might not have explained myself properly
We dont want to replace CUCM licenses with CME licenses, it is more a case of if one of the remote sites which currently is registered to the CUCM decides to go on their own and use their 2851 gateway as a CME router can the current licenses on the CUCM system be transfered to the CME.
I thought that the CUCM licenses could only be transferred between CUCM systems but I might be wrong
I was also a bit unsure of what type of phone licenses the CME actually needed.
Thanks for your input
Ellen

Cisco CME integration with NICE recorder

Hi Team,
Please let me know if we can integrate Cisco CME with NICE recorder. If yes, please share the steps involved in configuration on CME side and NICE.
Warm Regards,
Dinesh Rathi

Hi Anas,
If the panasonic prtocol is H323 will be no problem the CME dial peer will configure as below:-
dial-peer voice 121 voip
translation-profile outgoing prefix
destination-pattern x..
session target ipv4:192.x.x.x
dtmf-relay h245-alphanumeric h245-signal
but if the panasonic protocol is SIP the configuration on CME will be as below:-
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
supplementary-service h450.12
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
sip
registrar server expires max 250 min 200
no call service stop
dial-peer voice 2000 voip
destination-pattern y...
session protocol sipv2
session target ipv4:172.x.x.x
dtmf-relay sip-notify
codec g711ulaw
no vad
So please can you advise for the above if the two options are right or not??
Thanks

Cisco Enterprise License Manager

Is Cisco Enterprise License Manager the samething as Cisco License Manager? I just want to make sure I am downloading the correct thing to work with CUCM 9.0. This is the only thing I was able to find in the Support downloads when logged in with my CCO:
Cisco License Manager 3.2.3 Client and Server Package (Windows)

And thus, same HW requirements as CUCM if you want to do a standalone install.
Otherwise, just use the one that will get installed with CUCM or CUC.
HTH
java
if this helps, please rate
www.cisco.com/go/pdihelpdesk

Nortel VS Cisco CME

I have a client who liked Cisco initially but now likes Nortel IP-PBX.
I dont know much about Nortel to compare it to Cisco's CME.
Can anyone please provide me some links or information on why Cisco CME solution is better than Nortel's Hybrid solution.
Thanks
AD

What about applications? Is Nortel BCM Proprietary? Does it allow thir party integration for Phone applications?
Our main concerns are:
1. Management of the system.
2. Reliability.
3. Need for third party application integration.
4. What is the value of BCM vs CME?
Our company has 120 phones at the moment. But that number will increase to 180 in next 2 years. We may not reach 240 in next 5-7 years. So we are condidering call Manager express solution. Initially we were looking at Call Manager solution but got some bad feedback about the CCM. Is CCM 5.0 reliable?
What does CCM provide that CME does'nt ?
We will be adding 15 warehouse locations in the future. But each location will have 2 users. There will be 2 Analog lines for these locations and may be use Softphone through VPN tunnel. We do not have point to point T-1's for remote locations. Just a Internet T-1 for VPN tunnel to main location.
Thanks for your help
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Cisco BE6000 Licensing

Dear all,
I have a question concerning Cisco BE6000 licensing. I'ts clear that MD server support 1000 users and 1200 devices.
In this case, Can I add additional phones from UPM or CUCM without being associated to a user ? Means,that I can have 200 additional phones without users and 1000 users associated to a phone.
Can I add a phone directly from CUCM and not from UPM ? Does UPN updated directly its users and phone directory from CUCM ?
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Regards,

BE6000 licensing is based on licensing users and unassociated devices. An unassociated phone still consumes the "User Connect License" that is required for that type of phone. You are correct in saying that the MD server does support up to 1200 devices, so as long as you are licensing and deploying no more than 1,000 users (think total and configured licenses/users) and 1,200 devices... Then you are in good shape.
I presume that you are using Cisco Prime Collaboration Provisioning (PCP) instead of the older UPM. With CPC, you can add phones and users into UCM and they will be "synced" over to CPC at the next configured sync schedule. No problem.
It is still a good idea, from a change management perspective, to add/move/change user services in CPC. It will keep track of everything that has been done to/for a user.

Cisco CME and Calls through SIP provider

Hello, friends.
There are Cisco (C2801-ADVENTERPRISEK9_IVS-M), Version 15.1 (4) M7.
Telephones connected to SCCP, registered SIP from the provider.
When I try to call to test number 4444 through sip in debug I see:
*Feb 10 01:51:25.317: //53363/2739DFE79696/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP XXXXXXXXXXX:5060;branch=z9hG4bK100D02077;rport=5060
From: "TEST" <sip:[email protected]>;tag=131CC60C-1D40
To: <sip:[email protected]>;tag=b638310eda6e4a73cf10b7fe3c94c572.bef7
Call-ID: [email protected]
CSeq: 101 INVITE
Proxy-Authenticate: Digest realm="sip.zadarma.com", nonce="Uvf1OFL39Awnou/oMiaFQrf9jyybhFmf", qop="auth"
Server: kamailio (4.0.3 (x86_64/linux))
Content-Length: 0
*Feb 10 01:51:25.325: //53363/2739DFE79696/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP XXXXXXXXXX:5060;branch=z9hG4bK100D02077
From: "TEST" <sip:[email protected]>;tag=131CC60C-1D40
To: <sip:[email protected]>;tag=b638310eda6e4a73cf10b7fe3c94c572.bef7
Date: Sun, 09 Feb 2014 21:51:25 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
Cisco при этом зарегана у провайдера SIP
DC#show sip-ua register status
Line peer expires(sec) registered P-Associ-URI
Configuration:
voice service voip
ip address trusted list
ipv4 178.16.26.122 255.255.255.255
ipv4 144.76.42.108 255.255.255.255
ipv4 176.9.145.115 255.255.255.255
ipv4 5.9.108.25 255.255.255.255
ipv4 78.46.95.118 255.255.255.255
ipv4 89.249.23.194 255.255.255.255
ipv4 178.16.26.124 255.255.255.255
ipv4 176.9.85.133 255.255.255.255
ipv4 46.4.53.86 255.255.255.255
ipv4 5.9.84.165 255.255.255.255
ipv4 78.16.26.122 255.255.255.255
ipv4 77.235.62.222 255.255.255.255
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
sip
registrar server
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8
codec preference 3 g711alaw
voice register global
max-dn 10
max-pool 10
voice register dn 1
number 150
voice register dn 2
number 151
voice translation-rule 9
rule 1 /^95/ //
voice translation-rule 1020
rule 1 /^.$/ /40232/
voice translation-profile outgoing
translate calling 1020
translate called 9
mgcp fax t38 ecm
mgcp profile default
dial-peer voice 2 voip
translation-profile outgoing outgoing
destination-pattern 95....
session protocol sipv2
session target sip-server
voice-class codec 1
no voice-class sip outbound-proxy
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay rtp-nte
no vad
sip-ua
credentials username 40232 password 7 XXXXXXXXXX realm sip.zadarma.com
authentication username 40232 password 7 XXXXXXXXXXXX realm sip.zadarma.com
registrar dns:sip.zadarma.com:5060 expires 3600
sip-server dns:sip.zadarma.com:5060
connection-reuse
host-registrar
DC#show sip-ua register status
Line                             peer       expires(sec) registered P-Associ-URI
================================ ========== ============ ========== ============
150                              40001      12           no
40232                            -1         550          yes
SIP provider says cisco trying to call with the internal call number, and it is necessary in order that have an SIP provider:
Wrong Remote-Party-ID: "Vankuver" <sip:61@<my ip>>;party=calling;
Should be so sip:40232@<my ip>
Please help me!

Yes, I behind nat.
*Feb 10 18:11:53.425: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/TCP 192.168.11.14:42294;branch=z9hG4bK-d8754z-e645887cf7416a27-1---d8754z-;rport
Max-Forwards: 70
Contact:
To: "954444"
From: "150";tag=7b409f06
Call-ID: ZjUzNjkwMWMyZDAyYmY1OWU2NjgzYzQwZjYyZWM5ZGU.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 314
v=0
o=- 2 2 IN IP4 192.168.11.14
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.11.14
t=0 0
m=audio 5724 RTP/AVP 107 0 8 101
a=alt:1 2 : gNONJ/Dj BaLJhmb/ 10.200.16.55 5724
a=alt:2 1 : DQ3e8qud c1qVrWui 192.168.11.14 5724
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=sendrecv
*Feb 10 18:11:53.477: //54341/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK1038E7FF
From: "" >;tag=169E6BC4-1E16
To: [email protected]>
Date: Mon, 10 Feb 2014 14:11:53 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 0541864002-2442400227-2618163141-2285537806
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1392041513
Contact: outside ip cisco cme:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 262
v=0
o=CiscoSystemsSIP-GW-UserAgent 8076 2450 IN IP4 92.63.108.115
s=SIP Call
c=IN IP4 92.63.108.115
t=0 0
m=audio 18534 RTP/AVP 0 8 101
c=IN IP4 92.63.108.115
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
*Feb 10 18:11:53.481: //54340/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 192.168.11.14:42294;branch=z9hG4bK-d8754z-e645887cf7416a27-1---d8754z-;rport
From: "150";tag=7b409f06
To: "954444"
Date: Mon, 10 Feb 2014 14:11:53 GMT
Call-ID: ZjUzNjkwMWMyZDAyYmY1OWU2NjgzYzQwZjYyZWM5ZGU.
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
*Feb 10 18:11:53.625: //54341/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP outside ip cisco cme:5060;branch=z9hG4bK1038E7FF;rport=5060
From: "" ;tag=169E6BC4-1E16
To: [email protected]>;tag=9fedfddccf3bcc4a1975d2cdb2a664b8.7066
Call-ID: [email protected]
CSeq: 101 INVITE
Proxy-Authenticate: Digest realm="sip.zadarma.com", nonce="Uvja/1L42dNbKQpCc2GzgagslkjyE1Pn", qop="auth"
Server: kamailio (4.0.3 (x86_64/linux))
Content-Length: 0
*Feb 10 18:11:53.633: //54341/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDPoutside ip cisco cme:5060;branch=z9hG4bK1038E7FF
From: "150" [email protected]>;tag=169E6BC4-1E16
To: [email protected]>;tag=9fedfddccf3bcc4a1975d2cdb2a664b8.7066
Date: Mon, 10 Feb 2014 14:11:53 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
*Feb 10 18:11:53.637: //54341/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDPoutside ip cisco cme:5060;branch=z9hG4bK1038F25FC
From: "" ;tag=169E6BC4-1E16
To: [email protected]>
Date: Mon, 10 Feb 2014 14:11:53 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 0541864002-2442400227-2618163141-2285537806
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Timestamp: 1392041513
Contact: :5060>
Expires: 180
Allow-Events: telephone-event
Proxy-Authorization: Digest username="40232",realm="sip.zadarma.com",uri="sip:[email protected]:5060",response="df38cd7f4af8e4a808fbbfdf5a7dd6a1",nonce="Uvja/1L42dNbKQpCc2GzgagslkjyE1Pn",cnonce="E701683F",qop=auth,algorithm=md5,nc=00000001
Max-Forwards: 69
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 262
v=0
o=CiscoSystemsSIP-GW-UserAgent 8076 2450 IN IP4 92.63.108.115
s=SIP Call
c=IN IP4 92.63.108.115
t=0 0
m=audio 18534 RTP/AVP 0 8 101
c=IN IP4 92.63.108.115
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
*Feb 10 18:11:53.981: //54341/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK1038F25FC;rport=5060
From: "" ;tag=169E6BC4-1E16
To: [email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
Server: kamailio (4.0.3 (x86_64/linux))
Content-Length: 0
*Feb 10 18:11:54.385: //54341/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 92.63.X:5060;rport=5060;branch=z9hG4bK1038F25FC
Record-Route:
From: "k40232" ;tag=169E6BC4-1E16
To: [email protected]>;tag=as7e8de8e5
Call-ID: [email protected]
CSeq: 102 INVITE
Server: Zadarma Voip
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Type: application/sdp
Content-Length: 281
v=0
o=root 1942395501 1942395501 IN IP4 178.16.26.124
s=Asterisk PBX
c=IN IP4 178.16.26.124
t=0 0
m=audio 12164 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
*Feb 10 18:11:54.409: //54341/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 92.63.xxxx.xxxx:5060;branch=z9hG4bK10390E63
From: "150" [email protected]>;tag=169E6BC4-1E16
To: [email protected]>;tag=as7e8de8e5
Date: Mon, 10 Feb 2014 14:11:53 GMT
Call-ID: [email protected]
Route:
Max-Forwards: 70
CSeq: 102 ACK
Proxy-Authorization: Digest username="40232",realm="sip.zadarma.com",uri="sip:[email protected]:5060",response="df38cd7f4af8e4a808fbbfdf5a7dd6a1",nonce="Uvja/1L42dNbKQpCc2GzgagslkjyE1Pn",cnonce="E701683F",qop=auth,algorithm=md5,nc=00000001
Allow-Events: telephone-event
Content-Length: 0
*Feb 10 18:11:54.429: //54340/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.11.14:42294;branch=z9hG4bK-d8754z-e645887cf7416a27-1---d8754z-;rport
From: "150";tag=7b409f06
To: "954444";tag=169E6F78-88E
Date: Mon, 10 Feb 2014 14:11:53 GMT
Call-ID: ZjUzNjkwMWMyZDAyYmY1OWU2NjgzYzQwZjYyZWM5ZGU.
CSeq: 1 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: :5060;transport=tcp>
Supported: replaces
Server: Cisco-SIPGateway/IOS-12.x
Supported: timer
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 193
v=0
o=CiscoSystemsSIP-GW-UserAgent 149 3396 IN IP4 92.63.108.115
s=SIP Call
c=IN IP4 92.63.108.115
t=0 0
m=audio 17190 RTP/AVP 8
c=IN IP4 92.63.108.115
a=rtpmap:8 PCMA/8000
a=ptime:20
*Feb 10 18:11:54.653: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 91.231.141.230:42294;branch=z9hG4bK-d8754z-95374017c126c928-1---d8754z-;rport
Max-Forwards: 70
Contact:
To: "954444";tag=169E6F78-88E
From: "150";tag=7b409f06
Call-ID: ZjUzNjkwMWMyZDAyYmY1OWU2NjgzYzQwZjYyZWM5ZGU.
CSeq: 1 ACK
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 0

Cisco CME: calls through SIP-provider again

Hello,friends!
I have already published a discussion here https://supportforums.cisco.com/discussion/12089656/cisco-cme-and-calls-through-sip-provider and you helped me, everything works well for Russian numbers.
When I tried to add the configuration for calls to Belarus, again, there was a problem. I do not understand why, although the configuration ideintichnaya.
My config:
voice service voip
ip address trusted list
ipv4 178.16.26.122 255.255.255.255
ipv4 144.76.42.108 255.255.255.255
ipv4 176.9.145.115 255.255.255.255
ipv4 5.9.108.25 255.255.255.255
ipv4 78.46.95.118 255.255.255.255
ipv4 89.249.23.194 255.255.255.255
ipv4 178.16.26.124 255.255.255.255
ipv4 176.9.85.133 255.255.255.255
ipv4 46.4.53.86 255.255.255.255
ipv4 5.9.84.165 255.255.255.255
ipv4 78.16.26.122 255.255.255.255
ipv4 77.235.62.222 255.255.255.255
ipv4 81.88.86.11 255.255.255.255
ipv4 192.168.1.50 255.255.255.255
ipv4 217.150.198.44 255.255.255.255
ipv4 178.63.96.3 255.255.255.255
ipv4 178.63.96.28 255.255.255.255
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service sip moved-temporarily
sip
registrar server
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8
codec preference 3 g711alaw
voice class sip-profiles 20
request INVITE sip-header From modify "\"(.*)\" <sip:(.*)@(.*)>" "\"\" <sip:[email protected]>"
voice translation-rule 9
rule 1 /^98/ /7/
voice translation-rule 10
rule 1 /^9/ //
voice translation-rule 1020
rule 1 /^.*$/ /141756/
voice translation-rule 1030
rule 1 /^.*/ /141756/
voice translation-rule 1040
rule 1 /^.*$/ /21/
voice translation-profile incoming
translate called 1040
voice translation-profile outgoing
translate calling 1030
translate called 9
voice translation-profile outgoing-mezhdunarod
translate calling 1030
translate called 10
voice-card 0
dial-peer voice 2 voip
description TO-RUSSIA
translation-profile outgoing outgoing
preference 1
destination-pattern 98..........
session protocol sipv2
session target sip-server
voice-class codec 1
no voice-class sip outbound-proxy
voice-class sip profiles 20
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay rtp-nte sip-notify
no vad
dial-peer voice 3 voip
translation-profile incoming incoming
incoming called-number 141756
voice-class codec 1
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay rtp-nte
no vad
dial-peer voice 4 voip
description To-Belarus
translation-profile outgoing outgoing-mezhdunarod
destination-pattern 9375.........
session protocol sipv2
session target sip-server
voice-class codec 1
no voice-class sip outbound-proxy
voice-class sip profiles 20
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay rtp-nte sip-notify
no vad
sip-ua
credentials username 141756 password 7<pass> realm sip.zadarma.com
authentication username 141756 password 7 <pass>
no remote-party-id
registrar 1 dns:sip.zadarma.com expires 3600
sip-server dns:sip.zadarma.com
connection-reuse
host-registrar
DEBUG ccsip message:
Jun 17 14:23:09.033: //14293/D2C3B137AE52/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK36571F65
From: "" <sip:[email protected]>;tag=40FCB218-23D7
To: <sip:[email protected]>
Date: Tue, 17 Jun 2014 09:23:09 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3536040247-4114026979-2924673736-0741251102
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1402996989
Contact: <sip:[email protected]:5060>
Call-Info: <sip:92.63.108.115:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 309
v=0
o=CiscoSystemsSIP-GW-UserAgent 6656 8059 IN IP4 92.63.108.115
s=SIP Call
c=IN IP4 92.63.108.115
t=0 0
m=audio 18252 RTP/AVP 0 18 8 101
c=IN IP4 92.63.108.115
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
Jun 17 14:23:09.089: //14293/D2C3B137AE52/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK36571F65;rport=5060
From: "" <sip:[email protected]>;tag=40FCB218-23D7
To: <sip:[email protected]>;tag=b638310eda6e4a73cf10b7fe3c94c572.6d40
Call-ID: [email protected]
CSeq: 101 INVITE
Proxy-Authenticate: Digest realm="sip.zadarma.com", nonce="U6AYAFOgFtT86kmu2Fr5tYxLYGEexIl1", qop="auth"
Server: kamailio (4.1.2 (x86_64/linux))
Content-Length: 0
Jun 17 14:23:09.169: //14293/D2C3B137AE52/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK36571F65
From: "Vankuver" <sip:[email protected]>;tag=40FCB218-23D7
To: <sip:[email protected]>;tag=b638310eda6e4a73cf10b7fe3c94c572.6d40
Date: Tue, 17 Jun 2014 09:23:09 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
Jun 17 14:23:09.169: //14293/D2C3B137AE52/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK365820F2
From: "" <sip:[email protected]>;tag=40FCB218-23D7
To: <sip:[email protected]>
Date: Tue, 17 Jun 2014 09:23:09 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3536040247-4114026979-2924673736-0741251102
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Max-Forwards: 70
Timestamp: 1402996989
Contact: <sip:[email protected]:5060>
Call-Info: <sip:92.63.108.115:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Proxy-Authorization: Digest username="141756",realm="sip.zadarma.com",uri="sip:[email protected]:5060",response="9534322838cbf2e265b2004bc0aa240e",nonce="U6AYAFOgFtT86kmu2Fr5tYxLYGEexIl1",cnonce="FFF9A231",qop=auth,algorithm=md5,nc=00000001
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 309
v=0
o=CiscoSystemsSIP-GW-UserAgent 6656 8059 IN IP4 92.63.108.115
s=SIP Call
c=IN IP4 92.63.108.115
t=0 0
m=audio 18252 RTP/AVP 0 18 8 101
c=IN IP4 92.63.108.115
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
Jun 17 14:23:09.637: //14293/D2C3B137AE52/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK365820F2
From: "" <sip:[email protected]>;tag=40FCB218-23D7
To: <sip:[email protected]>
Date: Tue, 17 Jun 2014 09:23:09 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3536040247-4114026979-2924673736-0741251102
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Max-Forwards: 70
Timestamp: 1402996989
Contact: <sip:[email protected]:5060>
Call-Info: <sip:92.63.108.115:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Proxy-Authorization: Digest username="141756",realm="sip.zadarma.com",uri="sip:[email protected]:5060",response="9534322838cbf2e265b2004bc0aa240e",nonce="U6AYAFOgFtT86kmu2Fr5tYxLYGEexIl1",cnonce="FFF9A231",qop=auth,algorithm=md5,nc=00000001
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 309
v=0
o=CiscoSystemsSIP-GW-UserAgent 6656 8059 IN IP4 92.63.108.115
s=SIP Call
c=IN IP4 92.63.108.115
t=0 0
m=audio 18252 RTP/AVP 0 18 8 101
c=IN IP4 92.63.108.115
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
Jun 17 14:23:10.621: //14293/D2C3B137AE52/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK365820F2
From: "" <sip:[email protected]>;tag=40FCB218-23D7
To: <sip:[email protected]>
Date: Tue, 17 Jun 2014 09:23:10 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3536040247-4114026979-2924673736-0741251102
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Max-Forwards: 70
Timestamp: 1402996990
Contact: <sip:[email protected]:5060>
Call-Info: <sip:92.63.108.115:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Proxy-Authorization: Digest username="141756",realm="sip.zadarma.com",uri="sip:[email protected]:5060",response="9534322838cbf2e265b2004bc0aa240e",nonce="U6AYAFOgFtT86kmu2Fr5tYxLYGEexIl1",cnonce="FFF9A
All possible debugging has been turned off
DC#231",qop=auth,algorithm=md5,nc=00000001
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 309
v=0
o=CiscoSystemsSIP-GW-UserAgent 6656 8059 IN IP4 92.63.108.115
s=SIP Call
c=IN IP4 92.63.108.115
t=0 0
m=audio 18252 RTP/AVP 0 18 8 101
c=IN IP4 92.63.108.115
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
Debug voice ccapi inout:
Destination Pattern=9375........., Called Number=375298911396, Digit Strip=FALSE
Jun 17 15:22:13.073: //14425/13366763AF35/CCAPI/ccCallSetupRequest:
Calling Number=141756(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
Called Number=375298911396(TON=Unknown, NPI=Unknown),
Redirect Number=, Display Info=Vankuver
Account Number=, Final Destination Flag=FALSE,
Guid=13366763-F540-11E3-AF35-FAC82C2E981E, Outgoing Dial-peer=4
Jun 17 15:22:13.073: //14425/13366763AF35/CCAPI/cc_api_display_ie_subfields:
ccCallSetupRequest:
cisco-username=
----- ccCallInfo IE subfields -----
cisco-ani=141756
cisco-anitype=0
cisco-aniplan=0
cisco-anipi=0
cisco-anisi=0
dest=375298911396
cisco-desttype=0
cisco-destplan=0
cisco-rdie=FFFFFFFF
cisco-rdn=
cisco-rdntype=0
cisco-rdnplan=0
cisco-rdnpi=0
cisco-rdnsi=0
cisco-redirectreason=0 fwd_final_type =0
final_redirectNumber =
hunt_group_timeout =0
Jun 17 15:22:13.073: //14425/13366763AF35/CCAPI/ccIFCallSetupRequestPrivate:
Interface=0x6968AA04, Interface Type=3, Destination=, Mode=0x0,
Call Params(Calling Number=141756,(Calling Name=Vankuver)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
Called Number=375298911396(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,
Subscriber Type Str=RegularLine, FinalDestinationFlag=FALSE, Outgoing Dial-peer=4, Call Count On=FALSE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Application Call Id=)
Jun 17 15:22:13.073: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Jun 17 15:22:13.073: :cc_get_feature_vsa malloc success
Jun 17 15:22:13.073: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Jun 17 15:22:13.077: cc_get_feature_vsa count is 2
Jun 17 15:22:13.077: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Jun 17 15:22:13.077: :FEATURE_VSA attributes are: feature_name:0,feature_time:1819298856,feature_id:3371
Jun 17 15:22:13.077: //14427/13366763AF35/CCAPI/ccIFCallSetupRequestPrivate:
SPI Call Setup Request Is Success; Interface Type=3, FlowMode=1
Jun 17 15:22:13.077: //14427/13366763AF35/CCAPI/ccCallSetContext:
Context=0x6C726BF4
Jun 17 15:22:13.077: //14425/13366763AF35/CCAPI/ccSaveDialpeerTag:
Outgoing Dial-peer=4
Jun 17 15:22:13.085: //14427/13366763AF35/CCAPI/cc_api_call_proceeding:
Please help me... I don't know what to do!

You need to contact service provider for this , after authentication challenge your sip provider is not sending any response.
Contact them and ask whether they had received INVITE with proxy authentication details or not.

Cisco ISE licensing...

Hi,
seeking help to reduce our ISE licensing cost, actually we are out budget and we planning to order ISE licenses less than what we required, and looking for efficiently using the same, is there any way, i mean if we reduce "user idle timeout" is it reduce our license consumption?
any kind help appreciated...
thank you,

License Count
A Cisco ISE user consumes a license during an active session. Once the sessions has ended, ISE releases the license for reuse by another user.
The Cisco ISE license is counted as follows:
A Base, Plus, or Advanced license is consumed based on the feature that is used.
An endpoint with multiple network connections can consume more than one license per MAC address. For example, a laptop connected to wired and also to wireless at the same time. Licenses for VPN connections are based on the IP address.
Licenses are counted against concurrent, active sessions. An active session is one for which a RADIUS Accounting Start is received but RADIUS Accounting Stop has not yet been received.

Cisco/go/license DOWN?

Hi I am trying to access the licensing page to obtain license from my PAK key urgently. It says internal server error when I access www.cisco/go/license
Please advise
Sent from Cisco Technical Support iPhone App

Hi Not too sure if your orgnial issue was a typo however as Rob indicates http://cisco.com/go/license redirects to https://tools.cisco.com/SWIFT/Licensing/PrivateRegistrationServlet.
Cheers
Mario

Login to cisco CME as Administrator failed check your call manager express

Hi Experts
CME and CUE in one router. when i access the CUE from the IE , i put the CUE username and password and i get in. After that it asks me to enter CME username and password to run the wizard. whenever i put the password i get this crappy message "Login to cisco CME as Administrator failed. check your call manager express config" i have cheked my config many times. Please let me know if someone has faced this problem or any suggestion on this. today is the 2nd time i have faced this problem , last time I cud not solve it and end up wasting 6 hours...
please help

Hi friend,
Here is the Prerequisites for Installing Cisco Unity Express Software. As David describes, may be the CME admin account is missing:
http://www.cisco.com/en/US/docs/voice_ip_comm/unity_exp/rel3_1/installation/guide/prereq31.html#wpmkr1112912
Try this, and let us know.
Best regards,
- AdriÃ¡n.

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.

Cisco sip ip phone CP-7841 couldn't registered in cisco CME SRST 2951

Hello,
My IP phone CP-7841 couldn't registered in my call manager. Can you help me please?
See the configuration below:
yourname#sh run
Building configuration...
Current configuration : 7013 bytes
! Last configuration change at 17:52:57 UTC Mon May 4 2015
version 15.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname yourname
boot-start-marker
boot-end-marker
logging buffered 51200 warnings
aaa new-model
aaa session-id common
ip dhcp excluded-address 192.168.2.1
ip dhcp excluded-address 158.113.41.1
ip dhcp excluded-address 10.10.10.1
ip dhcp pool voice
network 192.168.2.0 255.255.255.0
default-router 192.168.2.1
option 150 ip 192.168.2.1
lease 0 2
ip dhcp pool data
network 158.113.41.0 255.255.255.128
default-router 158.113.41.1
lease 0 2
voice service voip
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
registrar server expires max 1200 min 300
voice register pool-type 1
description Cisco IP phone 7841
reference-pooltype 6941
voice register global
mode cme
source-address 192.168.2.1 port 5060
max-dn 400
max-pool 150
load 7841 sip78xx.10-1-1SR1-4.loads
load 7861 sip78xx.10-1-1SR1-4.loads
authenticate register
authenticate realm all
tftp-path flash:
file text
create profile sync 0002065003436039
voice register dn 1
number 101
name phone 1
voice register dn 2
number 102
voice register pool 1
busy-trigger-per-button 2
id mac E0D1.73E4.0C58
type 7841
number 1 dn 1
voice register pool 2
busy-trigger-per-button 2
id mac E0D1.73E4.A54C
type 7861
number 1 dn 2
license udi pid CISCO2951/K9 sn FCZ185070TS
hw-module pvdm 0/0
username unicef privilege 15 secret 5 $1$pCbf$7NtwVixNLu1vbwJJLRTN5.
redundancy
interface Embedded-Service-Engine0/0
no ip address
shutdown
interface GigabitEthernet0/0
description $ETH-LAN$$ETH-SW-LAUNCH$$INTF-INFO-GE 0/0$
ip address 10.10.10.1 255.255.255.248
duplex auto
speed auto
interface GigabitEthernet0/1
no ip address
duplex auto
speed auto
interface GigabitEthernet0/1.10
description router interface for voice vlan
encapsulation dot1Q 10
ip address 192.168.2.1 255.255.255.0
interface GigabitEthernet0/1.50
description router interface for data vlan
encapsulation dot1Q 50
ip address 158.113.41.1 255.255.255.0
interface GigabitEthernet0/2
no ip address
shutdown
duplex auto
speed auto
ip forward-protocol nd
ip http server
ip http access-class 23
ip http authentication local
ip http secure-server
ip http timeout-policy idle 60 life 86400 requests 10000
nls resp-timeout 1
cpd cr-id 1
tftp-server flash:rootfs78xx.10-1-1SR1-4.sbn
tftp-server flash:sboot78xx.10-1-1SR1-4.sbn
tftp-server flash:sip78xx.10-1-1SR1-4.loads
access-list 23 permit 10.10.10.0 0.0.0.7
control-plane
mgcp behavior rsip-range tgcp-only
mgcp behavior comedia-role none
mgcp behavior comedia-check-media-src disable
mgcp behavior comedia-sdp-force disable
mgcp profile default
gatekeeper
shutdown
telephony-service
max-conferences 8 gain -6
transfer-system full-consult
banner exec ^C
% Password expiration warning.
Cisco Configuration Professional (Cisco CP) is installed on this device
and it provides the default username "cisco" for one-time use. If you have
already used the username "cisco" to login to the router and your IOS image
supports the "one-time" user option, then this username has already expired.
You will not be able to login to the router with this username after you exit
this session.
It is strongly suggested that you create a new username with a privilege level
of 15 using the following command.
username <myuser> privilege 15 secret 0 <mypassword>
Replace <myuser> and <mypassword> with the username and password you want to
use.
^C
banner login ^C
Cisco Configuration Professional (Cisco CP) is installed on this device.
This feature requires the one-time use of the username "cisco" with the
password "cisco". These default credentials have a privilege level of 15.
YOU MUST USE CISCO CP or the CISCO IOS CLI TO CHANGE THESE PUBLICLY-KNOWN
CREDENTIALS
Here are the Cisco IOS commands.
username <myuser> privilege 15 secret 0 <mypassword>
no username cisco
Replace <myuser> and <mypassword> with the username and password you want
to use.
IF YOU DO NOT CHANGE THE PUBLICLY-KNOWN CREDENTIALS, YOU WILL NOT BE ABLE
TO LOG INTO THE DEVICE AGAIN AFTER YOU HAVE LOGGED OFF.
For more information about Cisco CP please follow the instructions in the
QUICK START GUIDE for your router or go to http://www.cisco.com/go/ciscocp
^C
line con 0
line aux 0
line 2
no activation-character
no exec
transport preferred none
transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
stopbits 1
line vty 0 4
access-class 23 in
privilege level 15
transport input telnet ssh
line vty 5 15
access-class 23 in
privilege level 15
transport input telnet ssh
scheduler allocate 20000 1000
end

*****After looking at your config again, I see your "voice register pool-type 7841" had been added. I should have looked at the config a little closer.
Hello Akaffou Aristid,
The 7841 phone is a revision of the older 6941 phone and it needs to be referenced in the config.(Fast Track Configuration)
New SIP Phone models validated for CME using Fast-track configuration
voice register pool-type 7841
description Cisco IP Phone 7841
reference-pooltype 6941
"Once the new SIP phone model is configured using the fast-track configuration approach, the new phone model can be provisioned using the existing voice register pool configuration option as shown below."
Below is the link that references the above commands.
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucme/admin/configuration/guide/cmeadm/cmebasic.html#pgfId-1295535
Hope this helps.
Please rate helpful posts.
Thanks.

Not showing dashboard in cisco Enterprise License Manger

Dear All,
i have install CUCM 9.1 in UCS BE6000K
while installation i select only CUCM and installtion done
but when i access the CUCM after entering the ip address in the brower it's showing CUCM and Enterprise License Manger
CUCM is working fine i configured few parameter
issue when i click on Enterprise LIcense Manger i log in it's showing noting is there in the GUI page i can see only on top of the page left side corner Enterprise license manger and in right side top of the corner log in , about etc.
why is not showing this parameter like monitoring, dashboard, license usage,license management, inventory,administrion in the Enterprise License Manger

Hi Lalesh,
there is a bug
https://supportforums.cisco.com/thread/2252230
ELM GUI not displaying on Chrome 29+ and Firefox 25
CSCul30396
Description
Symptom:
ELM GUI does not display properly, only the page header is visible.
Conditions:
User is trying to access ELM using the latest version of Chrome or Firefox.
Workaround:
The ELM GUI displays properly on both Internet Explorer 9 and 10, as well as Safari (Mac and Windows).
regds,
aman

Cisco prime license confusion

Hi I have installed cisco prime 1.2 to manage router, AP, controller, switch and ISE
and I am confused wiht license
I have this 3 item
1. L-PILMS42-100
2. L-PINCSW11-100
3. L-PINCS11-100
I have already genereted and added item 3 on prime and it work
I gererated item 1 but I cannont add it on cisco prime, he dont reconnize the file
I am unnable to add my ISE on cisco prime
Do I need special licence fro ISE
Do I need to add the 3 license
Please advise

Do you want to use Prime Infrastructure or Prime LMS to manage the Catalyst 2960 switches? In either case it is possible - simply add the devices manually or discover them. Procedure for PI is here. Procedure for LMS is here.
The ISE appliances are not manageable in any but the most basic sense as they are not a supported Cisco device (for either Prime Infrastructure or Prime LMS) and will be seen the same as a generic non-Cisco deivce. i.e., only SNMP polling and traps (and, with LMS, potentially syslog data).

Cisco CME licensing

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