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
AD

Similar Messages

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 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.

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.

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.

Minimum Router's RAM requirement for Cisco CME

Hi,
Does anyone ever try to use below 1 GB RAM when installing CME for router 29XX? I know that Cisco recommend to use 1 GB RAM. I just want to know, what problem we will may face, if we implement CME with 2921 router for 50 users with default RAM (512 MB).

Hello,
As you mentioned; for 2911 routers the recommended is to use 1 GB of RAM as per the link below:
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucme/requirements/guide/cme105spc.html
As you also going to deploy it with its maximum ip phones that can support, I'd highly recommend you to use 1 GB. If you use 512 RAM, you may fall into performance issues and your setup will not be supported by Cisco TAC.
Thank you,
Shadi

Cisco CME 4.1 Factory Reset

Hello,
I hope this post finds itself in the correct place.
I have a Cisco 1760 Router with CME 4.1 running on it. We have had some changes and I would like to completely wipe the configuration on it. I would like to keep the actual firmware ios operating system on it but would like to get ride of any configuration changes made in the CLI. Is there a way to do this?
Thanks!

Hi
Can you find the below configuation?
http://www.cisco.com/c/en/us/support/docs/ios-nx-os-software/ios-software-releases-123-mainline/46509-factory-default.html
Thanks
Please rate all useful information

MoH on Cisco CME.

Hi to all,
I have a router Cisco 2801 (is a lab scenary) configured as call control. I have configured a CME 8.0. All is working properly but when I put a call waiting with an analogue phone connected to an ATA, MoH works fine but when I put the call waiting with a 7911 IP phone, the MoH isn't ear in the other peer.
Any Idea?? Seems something about DSP but, why works when I hold the call with the ATA??
The two types of phones are configured by the same mode. Tell me if you need something about.
Thanks for your help.

Hi,
May you please attach the out put of show voice dsp group all, show voice dsp detail for a working and non working call along with the output of debug ephone moh and debug voice ccapi inout for both the calls along with the calling, called and the party which pressed the hold key. Also send me the show tech from the router
Regards
Himank

Cisco cme paging security 3845 router password for paging access voip

Hi,
We have a Cisco 3845 router running CME, ephones and ephone-dn are configured, there is ephone-dn is configured for paging. The system and paging work fine. We want to give password for those who make announcement. There are currently more than 30 extensions but we only want to give paging access to three users. How can we do this?
Thank you

Sure. Using the same (or similar) example:
Extensions: 2XXX
Primary Phone: 2025552000
We can address the need with the following:
voice translation-rule 10
rule 1 /^2...$/ /2025552000/
voice translation-profile cme-to-itsp
translate calling 10
dial-p v 100 voip
description Example Egress Dial Peer to ITSP
destination-patt
translation-profile out cme-to-itsp
HTH.
-Bill (http://ucguerrila.com)

Cisco CME CfwdAll and Night Service

Hi,
we would like to forward a normal DN to a huntgroup (200 in the below example), but when the night-service is active,
the calls should be forwarded to a different number (300 in the below example):
ephone-dn 10 dual-line
number 100
call-forward all 200
call-forward night-service 300
However, it seems that "call-forward all" has priority over "call-forward night-service", so it won´t work like intended.
Any ideas how this could be achieved?
Many thanks
Heinz

Have you verified your night service is configured correctly and it works if call fwd all is not set? The below document says call-fwd night service should have precedence over call-fwd all:
Ref CME Admin guide: http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucme/admin/configuration/guide/cmeadm/cmetrans.html#pgfId-1106036
Call Forwarding
Call forwarding diverts calls to a specified number under one or more of the following conditions:
All calls—When all-call call forwarding is activated by a phone user, all incoming calls are diverted. The target destination for diverted calls can be specified in the router configuration or by the phone user with a soft key or feature access code. The most recently entered destination is recognized by Cisco Unified CME, regardless of how it was entered.
No answer—Incoming calls are diverted when the extension does not answer before the timeout expires. The target destination for diverted calls is specified in the router configuration.
Busy—Incoming calls are diverted when the extension is busy and call waiting is not active. The target destination for diverted calls is specified in the router configuration.
Night service—All incoming calls are automatically diverted during night-service hours. The target destination for diverted calls is specified in the router configuration.
A directory number can have all four types of call forwarding defined at the same time with a different forwarding destination defined for each type of call forwarding. If more than one type of call forwarding is active at one time, the order for evaluating the different types is as follows:
1. Call forward night-service
2. Call forward all
3. Call forward busy and call forward no-answer
-Terry
Please rate all helpful posts

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

Multicasting (IGMP Snoop) between Nortel and Cisco

We are currently having issues with Zen imaging (multicasting) and our setup is the following.
Please take into account, our knowledge is very limited with IGMP Snooping setup etc.
MDF = 6 Nortel 450-24T's using FirmWare -1.48 / SoftWare - 4.5.2.4
IGMP Settings are such :
VLAN: [ 1 ]
Snooping: [ Enabled ]
Proxy: [ Disabled ] -----> This was on...but once off, runs much smoother.
Robust Value: [ 2 ]
Query Time: [ 125 seconds ]
Set Router Ports: [ Version 1 ]
In the MDF (anythig directly in those switches) images fine now. (once I disabled PROXY)
However I have a few IDF's off the MDF that are using OLD Nortel 350F-HD's (no IGMP Snooping support) and it's horrible (can only do a few computers at a time.
So in one of the IDF's (the biggest one) I pulled out the 350F-HD and replaced it with a CISCO 2950 w/Fiber and it's using 12.1.20EA1 and I left IGMP Snooping on (thinking this will fix it) and couldn't even get ONE machine to connect and image in the multicast session. It's settings were (by default):
Global IGMP Snooping configuration:
IGMP snooping : Disabled
IGMPv3 snooping (minimal) : Enabled
Report suppression : Enabled
TCN solicit query : Disabled
TCN flood query count : 2
Vlan 1:
IGMP snooping : Disabled
Immediate leave : Disabled
Multicast router learning mode : pim-dvmrp
Source only learning age timer : 10
I then completly disabled IGMP Snooping on the CISCO and we're able to Image 5-7 Computers without a crash (more than that and it crashes - disconnects etc)
In the area's that I have All 450's or all CIsco's the imaging seems to go fine. (with minor errors)
Can any one give me some advice (or hopefully ran into this mixed setup before)?
Thank you.

Bosalaza,
Thank you for replying (and I read even more on the ip multicast routing). However I've not ran into the same issue at any school that has 100% cisco switches or 100% Nortels (that are setup correctly and not older than dirt). I think we've not needed the multicast routing setup as we only have one router on the network (and it's flat at the moment anyway). As long as IGMP Snooping is enabled correctly (on the switches) it seems to serve us well.
Although from what I've read (where you pointed me too) it seems even in our setup we would benifeit from taking time to setup "ip pim ....." etc.
I was able to scrounge from another network and change out a few very old Nortels (that didn't support IGMP Snoop) and all seems well now.
So long story short (and incase anyone else needs this info. The Nortel 350T and F - HD's were the main issue. It seems (for now) that a mixture of Nortel 350/450-24T's (any model that at least has IGMP Snooping) and Cisco's mixed (also Snoop on) works pretty well.
I'm going to consider this solved as I was able to fix it with changing out some old product. However I really appreciate your efforts and pointing my towards some good info. (Which I'm going to read up on more, as I'm sure we'll need to get it setup in the near future.)
Thanks again.

Urgent cisco cme caller-id

Hi,
i`m using a cisco call manager express with about 40 users using cisco ip phones 7931,7940....etc
when outside call ( eg.from my mobile or pstn) dial into the company and the 333(receptionist) phone rings - it is written from unknown number even after i add the command caller-id enable.
when i plugged the phone cable into a digital phone directly and make a call to this phone it works- caller-id has been showed.
how can i make the caller-id the outside caller appear on the cisco ip phone??
thanks

Hi
The provider may be providing caller id after couple of rings. When you enable caller-id, there are few options available. Try those.
By default, CME will expect the caller id after 1 ring. Also, make sure you configure "no battery-reversal" on the port. Some times, this causes the call to initiate again.
Let me know how it goes.
Thanks
- abu

Can't Web Browse to 6900 series phones running firmware 9.4.1.3 on cisco CME

Hi
I'm having issues with web browseing to cisco IP phones.
Historically we could web browse to a a basic settings/ dianostics page of the phone by simiply browseing to the phones IP address.
However after an upgrade to SCCP firmware 9.4.1.3 this is no longer possible? I have tested from the same lan/subnet but still it refuses connections. Did a telnet using port 80 and this is also refused.
Was this feature removed in the later firmware? I have always found this feature usful especially when doing degugging and remote testing.
If it is still possible to web to them does anyone know how this is enabled. I'm searching around but not found a solution yet.
Thanks
Simon

I think web access disabled by default started in 9.3(3) for these phone models:
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cuipph/6900_series/firmware/933/release_notes/P069_BK_R9A78776_00_rn-9_3_3-6900-series/P069_BK_R9A78776_00_rn-9_3_3-6900-series_chapter_00.html#P690_TP_WC33699B_00

Nortel VS Cisco CME

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