Cisco 79XX phones time question

I have a basic question that many may know the answer to. Cisco phones plugged into a CUCM 5,6,7,8x call manager - do the phones request the time from the CUCM cluster or does the CUCM cluster broadcast out the time to the phones? The question has been brought up and with a little searching the web and the forums - I have not found a clear answer.
Thanks.

In general all the phones get their time from call manager. Depending on model 7940/7960 will get the time with the offset applied from call manager, whereas the 7941/7961/7970 phones will get their time as UTC/GMT and then apply the offset locally. SIP phones use the NTP phone reference in call manager to contact the NTP server specified and then apply the offset based on their configuration file.

Similar Messages

Schedule time to shut down cisco IP phones

Dear All,
i would like to ask if there is any way to schedule time on CUCM 7 allowing us to shut down cisco ip phones during week ends, and which cisco ip phone models that support this feature.
regards

Hassan,
I'm no phone guy but if you want to POWER OFF the phones outside office hours there are serious implications to consider.
If your phones have PCs connected to them, there's a large chance that when the phones go off, the PCs will not be able to reach the network (because VoIP phones do not have bypass functionality).
Another thing is in an emergency, no one can make outgoing/incoming calls.
If this is what you want then you could try EnergyWise if you don't have acess to your CUCM services. Let's say you want to power off your phones at midnight and power up by 5am the next morning (weekdays):
!Globalenergywise domain security shared-secret energywise importance 60! Time Rangetime-range ONabsolute start 05:00 01 March 2013periodic weekday 5:00 to 23:59time-range OFFabsolute start 00:00 01 March 2013periodic weekday 0:0 to 5:00! Interfaceinterface GigabitEthernet0/1energywise level 10 recurrence importance 70 time-range ONenergywise level 0 recurrence importance 70 time-range OFFenergywise importance 60

Basic Cisco 7960 Phone Question

Hi, folks:
I am new to Cisco VoIP solution. I would like to set up a Cisco 7960 phone. When I provisioned the phone, I noticed that there is a parameter called "Admin VLAN ID".
Since my phone is connected to an access port of a switch, I believe it should not use any VLAN tagging.
So I am wondering whether this vlan tagging is only applied when a PC is subtended to the phone? If the PC belongs to a different VLAN, does it automatically change the connection to the switch to VLAN trunking?
If no PC is subtended to the phone, then, does the phone automatically turn off VLAN tagging?
I saw the phone got IP address from the local DHCP server, however, I cannot ping it. So does Cisco phone rejects ICMP packet by default?
Thanks a lot!
Shixiong

The "Cisco" solution to your problem would be to segment your voice and data networks and run QOS on the network to preserve voice quality. A typical Cisco switch port would look like this.
interface GigabitEthernet0/4
switchport trunk encapsulation dot1q
switchport mode trunk
switchport voice vlan 10
srr-queue bandwidth share 10 10 60 20
srr-queue bandwidth shape 10 0 0 0
queue-set 2
mls qos trust device cisco-phone
mls qos trust cos
no mdix auto
auto qos voip cisco-phone
spanning-tree portfast
This config would put a phone in VLAN 10 and either a directly attached PC or a PC hanging off the phone into VLAN 1. You would be responsible for making sure the proper routing and DHCP are available for both your Voice and Data network.
If you are just playing around and want to get a 7960 to work in the data network, remove all trunking info and voice vlan info and the phone should boot up with the same IP scheme as a PC on the data network (make sure this DHCP scope has correct option 150 settings - or this test will be very un-eventful for you - phone will not get a config.)
ICMP is not disabled by default, but if you have a phone that thinks it is in the voice vlan (called admin vlan on phone) but an address on the data network, it isn't going to work.
-nate

Using 802.1X and non-Cisco IP Phones

Hi there,
Having some questions about an 802.1x/non-Cisco ip phone setup and was hoping to find some answers/user-experience with this setup.
Main questions i'm facing:
1) When using non-Cisco ip phones (eg Nortel or Siemens) and a previous authorized client connected behind this ip phone gets disconnected. What will this action do with the authorized state of 802.1X on the switch port? WIll it stay authorized until the reauth timer expires or does it reject communication from any other device?
2) What about EAPOL-Logoff messages from the ip phone to the switch. Are these only used by Cisco phones when they experience a link-status change on data ports?
Thanks for sharing your thoughts

Overall, you need to try and deal with the fact that a machine can disappear from the network and the network may not know about it directly (i.e. Link doesn't go down).
I have no idea what other phones do, but Cisco phones send an EAPOL-Logoff when something is unplugged. This lets the switch know directly, and 1X session start is torn down immediately, closing what would be a security hole.
Fundamentally, re-auth is a workaround only, and this is not the reason to enable re-auth to begin with.
If your phone doesn't send an EAPOL-Logoff in this case, the switch might be left thinking an attack is underway when someone else tries to plug in (with presumably a different MAC). You do NOT want this to occur.
Hope this helps,

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

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

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

CME SIP issue - Cisco 7821 phone not registering

Hi
I am having issues with getting a Cisco 7821 phone to register.
Current deployment is with Cisco 6921 phones SCCP registration
SIP integration with CUE
SIP integration with Mitel system
c2951-universalk9-mz.SPA.154-3.M1.bin (CME 10.5)
In flash:
rootfs78xx.10-1-1SR1-4.sbn
kern78xx.10-1-1SR1-4.sbn
sboot78xx.10-1-1SR1-4.sbn
sip78xx.10-1-1SR1-4.loads
The 7821 phone gets IP address but fails to register. Please could somebody let me know why phone is not registering.
Configuration below (10.245.226.132 is CME address) .
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol pass-through g711ulaw
modem passthrough nse codec g711ulaw redundancy maximum-sessions 5
h323
sip
registrar server expires max 600 min 60
options-ping 90
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729r8
voice register global
mode cme
source-address 10.245.226.132 port 5060
max-dn 30
max-pool 10
load 7821 sip78xx.10-1-1SR1-4
authenticate register
authenticate realm all
timezone 22
date-format D/M/Y
voicemail 590
tftp-path flash:
create profile sync 0061443538560005
network-locale GB
voice register dn 1
number 1010
name user1
label user1
mwi
voice register pool 1
busy-trigger-per-button 2
id mac F09E.636E.63F2
type 7821
number 1 dn 1
presence call-list
dtmf-relay rtp-nte
username 1010 password 123
codec g711ulaw
no vad
dial-peer voice 391 voip
description *** Auto Attendant ***
destination-pattern 399
session protocol sipv2
session target ipv4:10.245.226.131
dtmf-relay sip-notify
codec g711ulaw
no vad
dial-peer voice 392 voip
description *** Administration Via Telephone ***
destination-pattern 392
session protocol sipv2
session target ipv4:10.245.226.131
dtmf-relay sip-notify
codec g711ulaw
no vad
dial-peer voice 393 voip
description *** Extension Assigner ***
service ea out-bound
destination-pattern 393
session target ipv4:10.245.226.132
dial-peer voice 590 voip
description *** Voice Mail Pilot ***
destination-pattern 590
b2bua
session protocol sipv2
session target ipv4:10.245.226.131
dtmf-relay sip-notify
codec g711ulaw
no vad
dial-peer voice 1 pots
description ** Match all incoming POTS calls **
translation-profile incoming IncomingPSTNcalls
incoming called-number .
direct-inward-dial
dial-peer voice 899 voip
description Call to Mitel
translation-profile incoming Prefix9
translation-profile outgoing rem44
destination-pattern [23]..
session protocol sipv2
session target ipv4:192.168.114.2
voice-class codec 1
dtmf-relay rtp-nte
no vad
interface GigabitEthernet0/0
description *** Connection to Mitel Phone System ***
ip address 192.168.114.5 255.255.255.248
duplex auto
speed auto
interface ISM0/0
description *** Connection to Cisco Unity Express ***
ip unnumbered GigabitEthernet0/1
service-module ip address 10.245.226.131 255.255.255.128
!Application: CUE Running on ISM
service-module ip default-gateway 10.245.226.132
interface GigabitEthernet0/1
description *** Connection to IP Phone LAN ***
ip address 10.245.226.132 255.255.255.128
duplex auto
speed auto
ip http server
ip http authentication local
ip http secure-server
ip http timeout-policy idle 60 life 86400 requests 10000
ip http path flash:
ip route 0.0.0.0 0.0.0.0 10.245.226.129
ip route 10.245.226.131 255.255.2
tftp-server flash:apps37sccp.1-4-4-0.bin
tftp-server flash:sip78xx.10-1-1SR1-4.loads
tftp-server flash:rootfs78xx.10-1-1SR1-4.sbn
tftp-server flash:sboot78xx.10-1-1SR1-4.sbn
sip-ua
mwi-server ipv4:10.245.226.131 expires 3600 port 5060 transport udp
registrar ipv4:10.245.226.132 expires 600
gatekeeper
shutdown
telephony-service
authentication credential cmeadmin c4p1ta2012
xml user xmladmin password xmladmin 15
extension-assigner tag-type provision-tag
max-ephones 104
max-dn 299
ip source-address 10.245.226.132 port 2000
auto assign 101 to 105
no service directed-pickup
timeouts interdigit 5
system message CFGS
url services http://10.245.226.131/voiceview/common/login.do
url authentication http://10.245.226.132/CCMCIP/authenticate.asp
cnf-file location flash:
cnf-file perphone
load 7931 SCCP31.9-2-1S
load 6921 SCCP69xx.9-2-1-0
time-zone 22
date-format dd-mm-yy
voicemail 590
max-conferences 8 gain -6
call-forward pattern .T
moh enable-g711 "music-on-hold.au"
web admin system name cmeadmin secret 5 $1$QmIK$46fDKVSudMxzI2bRp/Ef7/
time-webedit
transfer-system full-consult
transfer-pattern .T
secondary-dialtone 9
create cnf-files version-stamp Jan 01 2002 00:00:00
ephone-dn 298
number 598...
mwi on
ephone-dn 299
number 599...
mwi off

Page 7 of the following link recommends that you use option 150 with the Cisco 7800 series phones and use option 66 if you cannot use option 150
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cuipph/7821_7841_7861/10_1/english/admin_guide/PA2D_BK_AB3F74DA_00_admin-7821-7841-7861-10_0/PA2D_BK_AB3F74DA_00_admin-7821-7841-7861-10_0_chapter_01.pdf
Dynamic Host Configuration Protocol (DHCP)
DHCP dynamically allocates and assigns an IP address to network devices.
DHCP enables you to connect an IP phone into the network and have the phone become operational without your needing to manually assign an IP address or to configure additional network parameters.
DHCP is enabled by default. If disabled, you must manually configure the IP address, subnet mask, gateway, and a TFTP server on each phone locally.
Cisco recommends that you use DHCP custom option 150. With this method, you configure the TFTP server IP address as the option value. For additional supported DHCP configurations, go to the "Dynamic Host Configuration Protocol" chapter and the "Cisco TFTP" chapter in the Cisco Unified Communications Manager System Guide.
Note
If you cannot use option 150, you may try using DHCP option 66.

Cisco SIP Phone 9971 won't register on CME 8.6

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

Thank you for your reply.
I did some debugs and the results are very strange!
This is what I got:
Feb 24 18:01:12.219: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 400 Bad Request
Via: SIP/2.0/UDP 172.35.140.12:5060;branch=z9hG4bK08011844
From: ;tag=189c5db6bd09000260cf3daf-289a76d1
To: ;tag=52488-160A
Date: Mon, 24 Feb 2014 18:01:12 GMT
Call-ID: [email protected]
CSeq: 1000 REFER
Content-Length: 0
Contact:
Feb 24 18:01:12.291: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
REGISTER sip:172.25.140.1 SIP/2.0
Via: SIP/2.0/UDP 172.35.140.12:5060;branch=z9hG4bK1e9ad079
From: ;tag=189c5db6bd0900032df02e9c-25d79707
To:
Call-ID: [email protected]
Max-Forwards: 70
Date: Fri, 01 Jan 1982 00:02:41 GMT
CSeq: 101 REGISTER
User-Agent: Cisco-CP9971/9.4.1
Contact: ;+sip.instance="
000000-0000-0000-0000-189c5db6bd09>";+u.sip!devicename.ccm.cisco.com="SEP189C5DB
6BD09";+u.sip!model.ccm.cisco.com="493";video
Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-
cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-
cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-6.0.2,X-cisco-xsi-
8.0.1
Content-Length: 0
Reason: SIP;cause=200;text="cisco-alarm:22 Name=SEP189C5DB6BD09 ActiveLoad=sip99
71.9-4-1-9.loads InactiveLoad=sip9971.9-3-2SR1-1.loads Last=reset-reset"
Expires: 3600
Feb 24 18:01:12.395: voice_reg_get_reg_expires_timer: no voice register pool found
Feb 24 18:01:12.395: VOICE_REG_POOL: Register request for (1010) from (172.35.140.12)
Feb 24 18:01:12.395: VOICE_REG_POOL: Contact matches pool 1 number list 1
Feb 24 18:01:12.395: VOICE_REG_POOL: No entry for (172.35.140.12) found in srst contact table
Feb 24 18:01:12.395: VOICE_REG_POOL: key(1010) contact(172.35.140.12:5060) add to contact table
Feb 24 18:01:12.395: VOICE_REG_POOL: No entry for (1010) found in contact table
Feb 24 18:01:12.399: VOICE_REG_POOL: key(1010) contact(172.35.140.12) added to contact table
Feb 24 18:01:12.399: VOICE_REG_POOL: key(172.35.140.12) contact(1010) add to srst contact table
Feb 24 18:01:12.399: VOICE_REG_POOL: No entry for (172.35.140.12) found in srst contact table
Feb 24 18:01:12.399: VOICE_REG_POOL: key(172.35.140.12) contact(1010) added to srst contact table
Feb 24 18:01:12.399: VOICE_REG_POOL pool->tag(1), dn->tag(1), submask(1)
But right after these errors, I get the following:
Feb 24 18:01:12.399: VOICE_REG_POOL: Creating param container for dial-peer 4000
1.VOICE_REG_POOL pool->tag(1), dn->tag(1), submask(1)
VOICE_REG_POOL pool_tag(1), dn_tag(1)
Feb 24 18:01:12.399: VOICE_REG_POOL: Created dial-peer entry of type 0
Feb 24 18:01:12.399: VOICE_REG_POOL: Registration successful for 1010, registration id is 1
Feb 24 18:01:12.411: VOICE_REG_POOL: Contact matches pool 1 number list 1
Feb 24 18:01:12.411: VOICE_REG_POOL: GW SIS: X-cisco-cme-sis-1.0.0
Feb 24 18:01:12.411: VOICE REGISTER POOL-1 has registered.
                               Name:SEP189C5DB6BD09 IP:172.35.140.12 DeviceType:Phone
Feb 24 18:01:12.411: VOICE_REG_POOL: Pool[1]: service-control (reset type: 2) message sent to sip:[email protected]
Feb 24 18:01:12.411: voice_reg_privacy_update_to_phone: delay sending privacy update during bulk registration
Feb 24 18:01:12.415: //1/7B0070C28003/SIP/Msg/ccsipDisplayMsg:
====================
And when I do a sh voice register pool, I get the following:
C2811#sh voice register pool 1
Pool Tag 1
Config:
Mac address is 189C.5DB6.BD09
Type is 9971
Number list 1 : DN 1
Proxy Ip address is 0.0.0.0
Current Phone load version is Cisco-CP9971/9.4.1
DTMF Relay is enabled, rtp-nte
Call Waiting is enabled
DnD is disabled
Video is enabled
Camera is enabled
Busy trigger per button value is 0
call-forward b2bua busy 68600
keep-conference is enabled
registration expires timer max is 3600 and min is 120
username adm password adm
kpml signal is enabled
Lpcor Type is none
blf call list is enabled
Transport type is udp
service-control mechanism is supported
registration Call ID is [email protected]
Registration method: per line
Privacy feature is not configured.
Privacy button is disabled
active primary line is: 1010
contact IP address: 172.35.140.12 port 5060
Phone SIS Version: 6.0.2
GW SIS Version: 1.0.0
Dialpeers created:
Dial-peers for Pool 1:
dial-peer voice 40001 voip
destination-pattern 1010
session target ipv4:172.35.140.12:5060
session protocol sipv2
dtmf-relay rtp-nte
digit collect kpml
codec g711ulaw bytes 160
no vad
call-fwd-busy        68600
after-hours-exempt   FALSE
Statistics:
Active registrations : 4
Total SIP phones registered: 1
Total Registration Statistics
    Registration requests : 4
    Registration success   : 4
    Registration failed    : 0
    unRegister requests    : 0
    unRegister success     : 0
    unRegister failed      : 0
    Attempts to register
           after last unregister : 0
    Last register request time   : 18:11:43.551 UTC Mon Feb 24 2014
    Last unregister request time :
    Register success time        : 18:11:43.551 UTC Mon Feb 24 2014
    Unregister success time      :
C2811#
So apparently the Phone is actually registered!
However, the Phone screens still shows this message: Phone Not Registered.
So frankly I don't understand what's going on!
I really hope somebody can help. Thanks!

Cisco SIP Phone 9971 will not register on CME 8.6

Hello,
I'm trying to configure a Cisco SIP Phone 9971,
but it won't register on CME 8.6, which is running on a 2811
The Phone shows this error message: Phone Not Registered.
And when I check the the Status Messages in the Phone, I see the following:
VPN Error: vpn is not configured
Actually, it shows all these 4 messages in a constant Loop:
12:01:59a SEP189C5DB6BD09.cnf.xml (TFTP)
12:01:59a No Trust List instaled
12:01:59a Updating Trust list
12:02:00a VPN Error: VPN is not Configured
It seems that this VPN Error is keeping the Phone from registering.
This is repeated for ever and the Phone never registers; at least that's what it appears.
However, when I do a sh voice register pool, I get the following:
C2811#sh voice register pool 1
Pool Tag 1
Config:
Mac address is 189C.5DB6.BD09
Type is 9971
Number list 1 : DN 1
Proxy Ip address is 0.0.0.0
Current Phone load version is Cisco-CP9971/9.4.1
DTMF Relay is enabled, rtp-nte
Call Waiting is enabled
DnD is disabled
Video is enabled
Camera is enabled
Busy trigger per button value is 0
call-forward b2bua busy 68600
keep-conference is enabled
registration expires timer max is 3600 and min is 120
username adm password adm
kpml signal is enabled
Lpcor Type is none
blf call list is enabled
Transport type is udp
service-control mechanism is supported
registration Call ID is [email protected]
Registration method: per line
Privacy feature is not configured.
Privacy button is disabled
active primary line is: 1010
contact IP address: 172.35.140.12 port 5060
Phone SIS Version: 6.0.2
GW SIS Version: 1.0.0
Dialpeers created:
Dial-peers for Pool 1:
dial-peer voice 40001 voip
destination-pattern 1010
session target ipv4:172.35.140.12:5060
session protocol sipv2
dtmf-relay rtp-nte
digit collect kpml
codec g711ulaw bytes 160
no vad
call-fwd-busy        68600
after-hours-exempt   FALSE
Statistics:
Active registrations : 4
Total SIP phones registered: 1
Total Registration Statistics
    Registration requests : 4
    Registration success   : 4
    Registration failed    : 0
    unRegister requests    : 0
    unRegister success     : 0
    unRegister failed      : 0
    Attempts to register
           after last unregister : 0
    Last register request time   : 18:11:43.551 UTC Mon Feb 24 2014
    Last unregister request time :
    Register success time        : 18:11:43.551 UTC Mon Feb 24 2014
    Unregister success time      :
C2811#
This sh voice register pool seems to indicate that the Phone has actually registered.
But I still get the Phone Not Registered message on the screen!
I did some Debugs and they also seem to indicate that the Phone has indeed registered:
Feb 24 18:01:12.399: VOICE_REG_POOL: Creating param container for dial-peer 4000
1.VOICE_REG_POOL pool->tag(1), dn->tag(1), submask(1)
VOICE_REG_POOL pool_tag(1), dn_tag(1)
Feb 24 18:01:12.399: VOICE_REG_POOL: Created dial-peer entry of type 0
Feb 24 18:01:12.399: VOICE_REG_POOL: Registration successful for 1010, registration id is 1
Feb 24 18:01:12.411: VOICE_REG_POOL: Contact matches pool 1 number list 1
Feb 24 18:01:12.411: VOICE_REG_POOL: GW SIS: X-cisco-cme-sis-1.0.0
Feb 24 18:01:12.411: VOICE REGISTER POOL-1 has registered.
                               Name:SEP189C5DB6BD09 IP:172.35.140.12 DeviceType:Phone
So frankly, I have no idea why the Phone keeps showing the Phone Not Registered message.
I'm pasting the Router's config.
I hope somebody is able to point me in the right direction.
Here is the config. Thank you!
C2811#sh run
Building configuration...
version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname C2811
no aaa new-model
dot11 syslog
ip source-route
ip cef
ip dhcp excluded-address 172.25.140.1 172.25.140.10
ip dhcp excluded-address 172.35.140.1 172.35.140.10
ip dhcp pool Data
network 172.25.140.0 255.255.255.0
default-router 172.25.140.1
option 150 ip 172.25.140.1
dns-server 172.25.140.1
ip dhcp pool Voice
network 172.35.140.0 255.255.255.0
default-router 172.35.140.1
option 150 ip 172.35.140.1
dns-server 172.35.140.1
no ip domain lookup
no ipv6 cef
multilink bundle-name authenticated
voice service voip
allow-connections sip to sip
sip
registrar server expires max 3600 min 120
voice register global
mode cme
source-address 172.25.140.1 port 5060
max-dn 40
max-pool 42
load 9971 sip9971.9-4-1-9.loads
authenticate register
authenticate realm cisco
tftp-path flash:
create profile sync 0004820400584603
voice register dn 1
number 1010
allow watch
name Phone10
label Phone10
mwi
voice register pool 1
id mac 189C.5DB6.BD09
type 9971
number 1 dn 1
presence call-list
dtmf-relay rtp-nte
username adm password adm
call-forward b2bua busy 68600
codec g711ulaw
no vad
camera
video
voice-card 0
crypto pki token default removal timeout 0
crypto pki trustpoint TP-self-signed-1879153754
enrollment selfsigned
subject-name cn=IOS-Self-Signed-Certificate-1879153754
revocation-check none
rsakeypair TP-self-signed-1879153754
crypto pki certificate chain TP-self-signed-1879153754
certificate self-signed 01
(details ommited)
license udi pid CISCO2811 sn FTX1146A44H
username admin privilege 15 password 0 admin
redundancy
interface FastEthernet0/0
no ip address
duplex auto
speed auto
interface FastEthernet0/0.25
description Data VLAN
encapsulation dot1Q 25
ip address 172.25.140.1 255.255.255.0
interface FastEthernet0/0.35
description Voice VLAN
encapsulation dot1Q 35
ip address 172.35.140.1 255.255.255.0
interface FastEthernet0/1
no ip address
shutdown
duplex auto
speed auto
ip forward-protocol nd
ip http server
ip http authentication local
ip http secure-server
ip http timeout-policy idle 600 life 86400 requests 10000
tftp-server flash:P00308010200.bin
tftp-server flash:P00308010200.sbn
tftp-server flash:P00308010200.sb2
tftp-server flash:P00308010200.loads
tftp-server flash:SCCP42.9-3-1SR3-1S.loads
tftp-server flash:apps42.9-3-1ES19.sbn
tftp-server flash:cnu42.9-3-1ES19.sbn
tftp-server flash:cvm42sccp.9-3-1ES19.sbn
tftp-server flash:dsp42.9-3-1ES19.sbn
tftp-server flash:jar42sccp.9-3-1ES19.sbn
tftp-server flash:term42.default.loads
tftp-server flash:term62.default.loads
tftp-server flash:SCCP45.9-3-1SR3-1S.loads
tftp-server flash:apps45.9-3-1ES19.sbn
tftp-server flash:cnu45.9-3-1ES19.sbn
tftp-server flash:cvm45sccp.9-3-1ES19.sbn
tftp-server flash:dsp45.9-3-1ES19.sbn
tftp-server flash:jar45sccp.9-3-1ES19.sbn
tftp-server flash:term45.default.loads
tftp-server flash:term65.default.loads
tftp-server flash:/Ringtones/Ringlist.xml alias Ringlist.xml
tftp-server flash:/Ringtones/DistinctiveRingList.xml alias DistinctiveRingList.x
ml
tftp-server flash:sip9971.9-4-1-9.loads
tftp-server flash:kern9971.9-4-1-9.sebn
tftp-server flash:rootfs9971.9-4-1-9.sebn
tftp-server flash:dkern9971.100609R2-9-4-1-9.sebn
tftp-server flash:sboot9971.031610R1-9-4-1-9.sebn
tftp-server flash:skern9971.022809R2-9-4-1-9.sebn
tftp-server flash:/g4-tones.xml alias United_States/g4-tones.xml
tftp-server flash:/gd-sip.jar alias English_United_States/gd-sip.jar
control-plane
mgcp profile default
telephony-service
max-ephones 24
max-dn 48
ip source-address 172.25.140.1 port 2000
cnf-file location flash:
load 7960-7940 P00308010200
load 7942 SCCP42.9-3-1SR3-1S.loads
load 7945 SCCP45.9-3-1SR3-1S.loads
load 7962 SCCP42.9-3-1SR3-1S.loads
load 7965 SCCP45.9-3-1SR3-1S.loads
max-conferences 8 gain -6
dn-webedit
transfer-system full-consult
create cnf-files version-stamp 7960 Feb 11 2014 07:18:32
ephone-dn 1
number 1001
description Phone 1
name Phone 1
hold-alert 30 originator
ephone-dn 2
number 1002
description Phone 2
name Phone 2
hold-alert 30 originator
ephone-dn 3
number 1003
description Phone 3
name Phone 3
hold-alert 30 originator
ephone 1
device-security-mode none
mac-address 001C.58FB.6E0F
button 1:1
ephone 2
device-security-mode none
mac-address 0014.A981.7F8A
button 1:2
ephone 3
device-security-mode none
mac-address 0006.5356.A4B8
button 1:3
alias exec con conf t
alias exec sib show ip int brief
alias exec srb show run | b
alias exec sri show run int
line con 0
exec-timeout 0 0
logging synchronous
line aux 0
line vty 0 4
privilege level 15
login local
transport input telnet ssh
transport output telnet ssh
line vty 5 15
privilege level 15
login local
transport input telnet ssh
transport output telnet ssh
scheduler allocate 20000 1000
ntp master 1
end
C2811#

VPN is not Configured prints on all phones now with the built-in VPN client if VPN isn't configured. That's normal and is just cosmetic. That should not be causing your registration issues.

Cisco 7941G phones will not boot up after factory reset.

Cisco 7941G phones will not boot up after factory reset.
The following happens.
1. Speaker light lit (Green)
2. Phone displays a firmware upgrade page (see attachment) for a few minutes. (Doesn't seems to be upgrading, but it "saids" upgrading)
3. Saw Cisco logo with a Java Powered page.
4. The whole thing loops again and again...
I think the reason why the phone isn't booting up is because the phone needs to know where to find the configuration/firmware files from a tftp server. I do have a TFTP server, but normally I need to go in settings to change the alternative tftp server, but the thing is that now I can't even press the settings button...
Please help.

How many phones were reset and upgrading at one time? What is the setting of the TFTP Maximum Serving Count (Advanced TFTP Service Parm) on each of the TFTP servers?
Sounds like too many devices trying to use TFTP at once.

Can I configure a 79xx Phone with two busy trigger using UCCE 8.5?

Hi, my name is Eric and I've some doubts to configure my Phones. First of all, I will explain my environment below:
- UCCE 8.5
- CM 8.5
- CVP 8.0
I would like to know if it's supported if I configure a 79xx Phone with 2 busy triggers configuration, because the customer would like to the agent's phone receive two calls per line.
Thank you very much.

Eric,
No, this is not supported and stated as such in cisco docs.
Customer will need to designate separate DN for ACD calls and separate DN for DID/personal calls.
HTH,
Chris

No ringback from cisco ip phones to alcatel Phones

I have a CUCM 7.1.3 integrated with a Alcatel OXO PBX and when a try to place a call from cisco IP-Phones registered on CUCM to phones on PBX, I have no ringback,but when a place a call from cisco IP-Phones registered on CUCM to PSTN using PBX as it has a PRI line to the local PSTN, I dont have that issue.
The voice gateway connects to PBX using a PRI(E1) running MGCP and QSIG.
interface Serial 3/1:15
no ip address
encapsulation hdlc
isdn switch-type primary-qsig
isdn timer T310 120000
isdn protocol-emulate network
isdn incoming-voice voice
isdn bind-l3 ccm-manager
no cdp enable

You can configure COS values for the Cisco IP Phones so that the voice packets from Cisco IP phone are given more priority over the data packets from the PCs/Hosts. But the same might not be configurable for non Cisco IP Phone. I am not very sure of this. Other than that I guess, you can have all QoS features.

RTP for Cisco IP Phones

Hello friends!
I'm developing an application for Cisco IP Phones that whorks with JMF. Since there are some programmers in this forum that knows Cisco IP phones, I send my question here.
I was send succefully the XML message to phone (CiscoIPPhoneExecute), I was customized the paket's size to 160, etc. All works aparently. But audio stream does not reach to phone. Can anyone tell me why??
Below I write my source code.
Tanks!
Max.
import java.io.*;
import java.net.*;
import java.util.*;
import org.apache.xerces.impl.dv.util.Base64;
import javax.media.*;
import javax.media.control.*;
import javax.media.format.*;
import javax.media.protocol.*;
* @author Administrator
public class MMHTTPPost {
    /** Creates a new instance of MMHTTPPost */
    public MMHTTPPost() {
        try {
             * Envia el CiscoIPPhoneExecute al telefono
            String xml = new String("<CiscoIPPhoneExecute>"+
                                        "<ExecuteItem Priority=\"0\" URL=\"RTPRx:Stop\"/>"+"<ExecuteItem Priority=\"1\" URL=\"RTPRx:10.1.15.10:23480\"/>"+
                                        "</CiscoIPPhoneExecute>");
            String userId      = "Max",
                    password    = "12345",
                    basicAuth   = "Basic ",
                    params      = "XML=" + URLEncoder.encode(xml, "ISO-8859-1" );
            byte[] bytes       = params.getBytes();
            // Create a URL pointing to the servlet or CGI script and open an HttpURLConnection on that URL
            URL url = new URL( "http://10.1.15.56/CGI/Execute" );
            HttpURLConnection con = ( HttpURLConnection ) url.openConnection();
            // Indicate that you will be doing input and output, that the method is POST, and that the content
            // length is the length of the byte array
            con.setDoOutput( true );
            con.setDoInput( true );
            con.setRequestMethod( "POST" );
            con.setRequestProperty( "Content-length", String.valueOf( bytes.length ) );
            // Create the Basic Auth Header
            Base64 encoder = new Base64();
            basicAuth = (String)encoder.encode( ((String)(userId + ":" + password)).getBytes() );
            System.out.println("Codificado: " + basicAuth);
            con.setRequestProperty( "Authorization", "Basic " + basicAuth.trim() );
            // Write the parameters to the URL output stream
            OutputStream output = con.getOutputStream();
            output.write( bytes );
            output.flush();
            // Read the response
            BufferedReader input = new BufferedReader( new InputStreamReader( con.getInputStream() ) );
            while ( true ) {
                String line = input.readLine();
                if ( line == null )
                    break;
                System.out.println( line );
            input.close();
            output.close();
            con.disconnect();
             * Aca empieza la parte RTP/JMF
            // Create a Processor for the selected file. Exit if the
            // Processor cannot be created.
            Processor processor = null;
            try {
                String mediaUrl = "file:\\C:\\pruebas execute\\spacemusic.au";
                processor = Manager.createProcessor( new MediaLocator(mediaUrl));
            } catch (IOException e){
                System.out.println("Exception occured (1a): " + e);
            } catch (NoProcessorException e){
                System.out.println("Exception occured (1b): " + e);
            // configure the processor
            //processor.configure();
            //Espera el estado Processor.Configured
            boolean result = waitForState(processor, Processor.Configured);
            if (result == false)
                System.out.println("No se pudo configurar el procesador...");
            // Block until the Processor has been configured
            TrackControl track[] = processor.getTrackControls();
            boolean encodingOk = false;
            // Go through the tracks and try to program one of them to
            // output ulaw data.
            for (int i = 0; i < track.length; i++)
                if (!encodingOk && track[i] instanceof FormatControl)
                    if (((FormatControl)track).setFormat( new AudioFormat(AudioFormat.ULAW_RTP,8000,8,1)) == null)
track[i].setEnabled(false);
else
encodingOk = true;
else
// we could not set this track to ulaw, so disable it
track[i].setEnabled(false);
*Aqu? se cambia el tama?o del paquete a 160
if (((TrackControl)track[i]).isEnabled()) {
try {
System.out.println("Cambiando la lista de codecs...");
Codec codec[] = new Codec[3];
codec[0] = new com.ibm.media.codec.audio.rc.RCModule();
codec[1] = new com.ibm.media.codec.audio.ulaw.JavaEncoder();
//codec[2] = new com.ibm.media.codec.audio.ulaw.Packetizer();
codec[2] = new com.sun.media.codec.audio.ulaw.Packetizer();
((com.sun.media.codec.audio.ulaw.Packetizer)codec[2]).setPacketSize(160);
((TrackControl)track[i]).setCodecChain(codec);
catch (Exception e){
System.out.println("Error al cambiar el tamano del paquete: " + e);
System.out.println("Encoding ok?: " + encodingOk );
// At this point, we have determined where we can send out
// ulaw data or not.
// realize the processor
if (encodingOk)
//processor.realize();
// block until realized.
// get the output datasource of the processor and exit
// if we fail
//Espera el estado Processor.Realized
result = waitForState(processor, Processor.Realized);
if (result == false)
System.out.println("No se pudo realizar el procesador...");
DataSource ds = null;
try
ds = processor.getDataOutput();
catch (NotRealizedError e)
System.out.println("Exception occured(2): "+e);
// hand this datasource to manager for creating an RTP
// datasink.
// our RTP datasink will multicast the audio
try
String mediaUrl= "rtp://10.1.15.56:23480/audio/1";
MediaLocator m = new MediaLocator(mediaUrl);
DataSink d = Manager.createDataSink(ds, m);
d.open();
d.start();
catch (Exception e)
System.out.println("Exception occured(3): "+e);
catch ( MalformedURLException murlex )
System.out.println( murlex );
catch ( IOException ioex )
System.out.println( ioex );
* @param args the command line arguments
public static void main(String[] args) {
new MMHTTPPost();
* Convenience methods to handle processor's state changes.
private Integer stateLock = new Integer(0);
private boolean failed = false;
Integer getStateLock() {
     return stateLock;
void setFailed() {
     failed = true;
private synchronized boolean waitForState(Processor p, int state) {
     p.addControllerListener(new StateListener());
     failed = false;
     // Call the required method on the processor
     if (state == Processor.Configured) {
     p.configure();
     } else if (state == Processor.Realized) {
     p.realize();
     // Wait until we get an event that confirms the
     // success of the method, or a failure event.
     // See StateListener inner class
     while (p.getState() < state && !failed) {
     synchronized (getStateLock()) {
          try {
          getStateLock().wait();
          } catch (InterruptedException ie) {
          return false;
     if (failed)
     return false;
     else
     return true;
* Inner Classes
class StateListener implements ControllerListener {
     public void controllerUpdate(ControllerEvent ce) {
     // If there was an error during configure or
     // realize, the processor will be closed
     if (ce instanceof ControllerClosedEvent)
          setFailed();
     // All controller events, send a notification
     // to the waiting thread in waitForState method.
     if (ce instanceof ControllerEvent) {
          synchronized (getStateLock()) {
          getStateLock().notifyAll();

ignore my last post.. I didn't properly read the source code.
Anyway, I got g.711 streaming to work. I started out using the AVTransmit2 sample, but set the audio output format to what is used in the source code posted here. I've also applied the same codec chain described above, and that was it.
I could never get streaming to work using the first RTP streaming method described in the JMF Programmers guide (and this is the method used in the source posted above), but using the RTPManager, things work just fine.

Cisco IP Phone 7962 not registering with CME 9

Dear Experts,
I have CME router 2811 with 15 - 6921 phones and added 1 new Cisco 7962 phone. All the 6921 phones are registered and working fine.
7962 phone does not register and the screen goes blank after the phone boot. Software version the phone is running is 9.3.1 SR2-1S
Verified the CNF File is created
tftp-server system:/its/vrf1/XMLDefault7962.cnf.xml alias SEP501CBFFC8735.cnf.xml
Here is the configuration on the router.
ip dhcp pool VOICE
   network 192.168.10.0 255.255.255.0
   default-router 192.168.10.1
   option 150 ip 192.168.10.1
ephone-dn 11 octo-line
number 2211
label ABC 2221
name ABC
ephone 11
device-security-mode none
mac-address 501C.BFFC.8735
type 7962
button 1:11
The results of the debug tftp events are as below -
Oct 26 17:52:06.491: TFTP: Looking for CTLSEP501CBFFC8735.tlv
Oct 26 17:52:06.595: TFTP: Looking for ITLSEP501CBFFC8735.tlv
Oct 26 17:52:06.699: TFTP: Looking for ITLFile.tlv
Oct 26 17:52:06.931: TFTP: Looking for SEP501CBFFC8735.cnf.xml
Oct 26 17:52:07.487: TFTP: Opened system:/its/vrf1/XMLDefault7962.cnf.xml, fd 10, size 1278 for process 366
Oct 26 17:52:07.495: TFTP: Finished system:/its/vrf1/XMLDefault7962.cnf.xml, time 00:00:00 for process 366
Oct 26 17:52:09.799: TFTP: Looking for English_United_States/mk-sccp.jar
Oct 26 17:52:10.119: TFTP: Looking for United_States/g3-tones.xml
Oct 26 17:52:11.067: New Skinny socket accepted [2] from 0, sub 1 (15 active)
Oct 26 17:52:11.067: sin_family 2, sin_port 49152, in_addr 192.168.110.30
Oct 26 17:52:11.067: skinny_add_socket 2 192.168.110.30 49152
Oct 26 17:52:11.799: Cannot find device entry on socket fd 7 for message 346
Oct 26 17:52:11.799: Got wrong skinny message size 1836597052 on socket fd 7
Oct 26 17:52:11.799: Got wrong skinny message size 824327534 on socket fd 7
Oct 26 17:52:11.799: Got wrong skinny message size 1735289188 on socket fd 7
Oct 26 17:52:11.799: Got wrong skinny message size 1007304255 on socket fd 7
Oct 26 17:52:11.799: Got wrong skinny message size 1918987361 on socket fd 7
Oct 26 17:52:11.799: Got wrong skinny message size 1632510061 on socket fd 7
Oct 26 17:52:11.799: Got wrong skinny message size 1333032271 on socket fd 7 ... .so on
Oct 26 17:52:11.815: Got wrong skinny message size 2622 on socket fd 7
Oct 26 17:52:11.815: Got wrong skinny message size 0 on socket fd 7
Oct 26 17:52:11.815: Got wrong skinny message size 0 on socket fd 7
Oct 26 17:52:11.815: Got wrong skinny message size 0 on socket fd 7
Oct 26 17:52:11.815: Got wrong skinny message size 0 on socket fd 7
Oct 26 17:52:11.815: Got wrong skinny message size 0 on socket fd 7
Oct 26 17:52:11.815: Got wrong skinny message size 0 on socket fd 7
Oct 26 17:52:11.815: Got wrong skinny message size 0 on socket fd 7
Oct 26 17:52:11.815: Got wrong skinny message size 0 on socket fd 7
Oct 26 17:52:11.815: Got wrong skinny message size 0 on socket fd 7
Oct 26 17:52:11.815: Got wrong skinny message size 0 on socket fd 7
Oct 26 17:52:11.815: Got wrong skinny message size 0 on socket fd 7
Oct 26 17:52:11.815: Got wrong skinny message size 0 on socket fd 7
Oct 26 17:52:11.815: Got wrong skinny message size 0 on socket fd 7
Oct 26 17:52:11.815: Got wrong skinny message size 0 on socket fd 7
Oct 26 17:52:11.815: Got wrong skinny message size 0 on socket fd 7.. so on
Oct 26 17:52:11.883: Cannot find device entry on socket fd 7 for message 0
Oct 26 17:52:11.883: Got wrong skinny message size -2056126442 on socket fd 7
Oct 26 17:52:11.883: Got wrong skinny message size -54584240 on socket fd 7
Oct 26 17:52:11.883: Got wrong skinny message size 3 on socket fd 7
Oct 26 17:52:11.883: Got wrong skinny message size 0 on socket fd 7
Oct 26 17:52:11.883: Got wrong skinny message size 0 on socket fd 7
Oct 26 17:52:11.883: Got wrong skinny message size 825045805 on socket fd 7
Oct 26 17:52:11.883: Got wrong skinny message size 0 on socket fd 7
Oct 26 17:52:21.915: Cannot find device entry on socket fd 7 for message 0
Oct 26 17:52:41.995: Cannot find device entry on socket fd 7 for message 0
Oct 26 17:53:02.064: Cannot find device entry on socket fd 7 for message 0
ADVILLA-2811#
Oct 26 17:54:24.556: socket 3 fatal error 260! can't read msg header with size -1, fd 3
Oct 26 17:54:24.556: it's a stale socket! delete it!!
Please advise the issue.. thanks..

This could be a compatibility issue. Looking at the feature matrix, 15.1 is CME8.8 and only has support for SCCP42.9-2-1S.loads. Even the latest CME (10.5) only has listed support for 9.2.1 on 7962.
I would try downgrading the phone firmware to 9.2.1 and see if you continue to have the issue.
Also, make sure you are advertising all the following files on TFTP:
SCCP42.9-2-1S.loads
apps42.9-2-1TH1-13.sbn
cnu42.9-2-1TH1-13.sbn
cvm42sccp.9-2-1TH1-13.sbn
dsp42.9-2-1TH1-13.sbn
jar42sccp.9-2-1TH1-13.sbn
term42.default.loads
term62.default.loads

Using 79xx Phone PC access port to connect a Netgear 5 port switch

Cisco documentation indicates that on 79xx phones you can "Use the PC access port to connect a network device, such as a computer, to the phone". What types of problems might be encountered if that device is a Netgear Fast Ethernet Unmanaged FS105 five port switch? Is there any documentation indicating whether this "supported" or "not supported"?
I know that the speed on the devices connected to the Netgear will be cut to 10/100. The devices connected to the Netgear would include - at a minimum - a computer and a Cisco wireless access point. Other devices which might be connected could include additional computers and/or a printer.
The computer(s) would also need to access a distributed video stream daily.
We are running CallManager 8.6.2.21900-5 with phone types: 7940, 7941, 7945, 7965
Thanks for any insight you might be able to provide.

Will Cisco support you doing this? Not likely.
Will it work? I suppose, in theory, it should. Just don't complain if it doesn't.
GTG

Integration with Cisco IP Phones

Possible to integrate Lync to Cisco IP Phones?

Depends on what you mean by integrate. You cannot directly register Cisco phones with Lync, but you have the following options:
Create a SIP trunk (or trunk through a gateway) and connect CUCM to Lync and then you would be able to call back and forth
If you want to repurpose Cisco phones to use with Lync then you can use a third party product from AudioCodes (SPS -
http://www.audiocodes.com/sps ) or NET (SmartSIP -
http://www.net.com/Pages/Product.aspx?pgid=229 )
Tim Harrington | MVP: Lync | MCITP: EMA 2007/2010, Lync 2010, MCTS: OCS | Blog:
http://HowDoUC.blogspot.com | Twitter:
@twharrington

Cisco 79XX phones time question

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