Cisco 6921 phone stuck on start up

Cisco 6921 is plugged on PoE switch, patch cord & routing is tested with another phone. But one 6921 phone is stuck on start up process. All Receiver, mute, speaker, headset buttons lights are on but there is no led display.
Kindly advise.
Regards,
Humza Khan

Hi Humza,
If other 6921 phone are registering using the same switch port then it seems to be a hardware problem. Moreover you can try to delete this phone from CUCM and try to recreate it using different protocol (using SIP if it was configured as sccp or vice versa) after that you recreate it using the protocol which you want.
I hope you already did the factory reset.
Suresh

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Cisco 6921 Phones and CUCM 6.1

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Are there instuctions?
The phone does not appear in the Device Default list.
Thank You
Matt

Hello,
Already replied in another thread. Anyway, CallManager and CallManager Express (e.g. IOS gateways) support CDRs.
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Kind regards,
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Cisco 6921 phone SRST msg

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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 6921 - Inline Power Capable?

I am trying to determine if Cisco 6921 phones are actually compatible with Inline Power (Cisco pre-standard).
I have seen multiple data sheets from Cisco that say that they are, and then others that say that they only support 802.3af. They are 6.3w phones, so I'd assume they support pre-standard, but I am not 100% sure without having one in front of me. Does anybody know?
Thanks!

Wrong forum, should be "IP Telephony". You can move the posts using the control panel on the right".
Anwyay, that may also depend by the specific switch you have. Check:
https://supportforums.cisco.com/message/3581321

Unable to start cisco ip phone 7942

Hello Guys,
A friend of mine has given cisco ip phone 7942 to us (which was working good at his work desk)
We are trying to connect this phone at home but facing problem.
First thing we want to confirm is, does this phone require AC-DC power supply?
Reason for asking is we have connected all other things (cables) properly to get it working but we do not see anything on the display / phone screen.
Please check below picture for our connections ( Laptop - Phone - Modem and ADSL Phone Filter)
Friend has confirmed that they do not have any power connected to phone in office/work, so not sure if AC-DC power supply is really required?
Appreciate if you guys can help to resolve this.
Merry Christmas and Happy New Year!
- Mandy

Hi Guys, thanks for your response.
I am thinking to get/buy this Cisco CP-PWR-CUBE-3 VoIP Phone Power Supply
http://www.ebay.com.au/itm/Cisco-CP-PWR-CUBE-3-VoIP-Phone-Power-Supply-CP-7941G-7945G-7975G-7914-3MthWty-/281030628576?pt=LH_DefaultDomain_15&hash=item416ebb14e0
But I am confused with above response from Chris.
Will you be registering it with a specific SIP server or will you be installing a PBX such as Cisco Communications Manager, etc?
>>> I have no idea about this....I thought if I just simply get the power supply it should start working?
what is your intent for the phone?
>>> I just want to use this as my home phone. This phone was working at my friend's office desk.
Are you saying this phone is locked with something code etc? which won't allow me to use at home if I just get the power supply?
Please help me to understand this guys....
- Mandy

My phone stuck after it started to install IOS 8. I can't power it off , can't do anything.

my phone stuck after it started to install IOS 8. I can't power it off , can't do anything.

If your iPhone suddenly freezes or is just not working right, you can do a hard reset by following these simple instructions:
Press and hold the Home button (big circle below the screen) and the Sleep/Wake button (on top of the iPhone) simultaneously.
Hold both buttons until the iPhone shuts off and begins to restart.

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#

Cisco 877 router - Cisco IP phone won't register with SIP provider

Hi all,
I'm having a problem with a Cisco SPA504G phone not registering with the SIP carrier over the Internet. We've recently rolled out a Cisco 877 router onto a new NBN business connection and can't get the pre-configured IP phone to register.
When we tested the phone with the NBN-provided Netgear router, it worked fine, as it did with the previous Cisco 1841 router we were using on a different link.
The way it's setup is using VLANs to define the internal subnets, which are then assigned to the physical interfaces (since the 887 doesn't allow IP assignments to the interfaces directly).
VLAN 100 is the internal network and has a SBS2011 server – assigned to F0 – IP range is 192.168.1.0
VLAN 200 is the guest network and has Internet access only – assigned to F1 – IP range is 10.1.1.0
VLAN 500 is the WAN network and connects to the NBN upstream box – assigned to F3 – external IP address assigned by DHCP
I've been playing around with access lists, nat rules, basically everything in my limited Cisco knowledge to try and figure this out, but to no avail. I have even configured what I believe is unrestricted access to IP, UDP and TCP outbound and inbound to all VLANs and still can't get it to register.
Tried isolating the issue by creating a new VLAN and assigning it to the spare interface and basically allowing everything in and out, but still no luck.
The problem has to be something on the router – probably some small line of config I haven’t removed or added.
I am going to pull my hair out soon, so would really appreciate some assistance from the Cisco gurus out there.
My client has just purchased about 10 of these handsets from their provider so I need to fix this ASAP. The guy who provided them wasn't very helpful, and basically said I'm on my own once we tested using the NBN-provided Netgear router.
Happy to post my config as well.
Please help!!!!

Current configuration : 4912 bytes
version 15.1
no service pad
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname Router1
boot-start-marker
boot-end-marker
no aaa new-model
memory-size iomem 10
crypto pki token default removal timeout 0
no ip source-route
ip dhcp excluded-address 10.1.1.1
ip dhcp pool GUEST
network 10.1.1.0 255.255.255.0
dns-server 10.1.1.1 203.50.2.71 139.130.4.4
default-router 10.1.1.1
ip cef
no ip domain lookup
ip domain name network.local
ip name-server 192.168.1.123
ip name-server 203.23.53.12
ip name-server 197.12.32.86
ip name-server 8.8.8.8
no ipv6 cef
license udi pid CISCO887VA-K9 sn FGL171220XY
username admin privilege 15 secret 5 $1$aNsm$N1BCQYkoi8gnURyvloYEX/
controller VDSL 0
interface Ethernet0
no ip address
shutdown
interface ATM0
no ip address
no atm ilmi-keepalive
bridge-group 10
pvc 8/35
interface FastEthernet0
description NAC - Internal network
switchport access vlan 100
no ip address
interface FastEthernet1
description NAC - Guest network
switchport access vlan 200
no ip address
interface FastEthernet2
no ip address
shutdown
interface FastEthernet3
description **** WAN Port ****
switchport access vlan 500
no ip address
interface Vlan1
no ip address
bridge-group 10
hold-queue 100 out
interface Vlan100
description NAC - Internal Vlan
ip address 192.168.1.1 255.255.255.0
ip access-group IN-100 in
ip access-group OUT-100 out
ip nat inside
ip virtual-reassembly in
interface Vlan200
description NAC - Guest Vlan
ip address 10.1.1.1 255.255.255.0
ip access-group IN-200 in
ip access-group OUT-200 out
ip nat inside
ip virtual-reassembly in
interface Vlan500
description **** WAN Vlan ****
ip address dhcp
ip nat outside
no ip virtual-reassembly in
no ip forward-protocol nd
ip http server
ip http access-class 23
ip http secure-server
ip dns server
ip nat inside source list NAT-100 interface Vlan500 overload
ip nat inside source list NAT-200 interface Vlan500 overload
ip nat inside source static tcp 192.168.1.123 25 interface Vlan500 25
ip nat inside source static tcp 192.168.1.123 443 interface Vlan500 443
ip nat inside source static tcp 192.168.1.123 3389 interface Vlan500 3399
ip nat inside source static tcp 192.168.1.123 80 interface Vlan500 80
ip nat inside source static tcp 192.168.1.123 4125 interface Vlan500 4125
ip nat inside source static tcp 192.168.1.124 3389 interface Vlan500 3390
ip nat inside source static tcp 192.168.1.123 987 interface Vlan500 987
ip nat inside source static tcp 192.168.1.123 1723 interface Vlan500 1723
ip route 0.0.0.0 0.0.0.0 55.234.52.43
ip access-list extended IN-100
permit udp any any range bootps bootpc
deny ip 10.1.1.0 0.0.0.255 any
permit ip 192.168.1.0 0.0.0.255 any
ip access-list extended IN-200
permit udp any any range bootps bootpc
permit ip 10.1.1.0 0.0.0.255 any
ip access-list extended NAT-100
deny ip 192.168.0.0 0.0.255.255 192.168.0.0 0.0.255.255
permit ip 192.168.1.0 0.0.0.255 any
ip access-list extended NAT-200
deny ip 10.1.0.0 0.0.255.255 10.1.0.0 0.0.255.255
permit ip 10.1.1.0 0.0.0.255 any
ip access-list extended OUT-100
permit udp any range bootps bootpc any
deny ip 10.1.1.0 0.0.0.255 any
permit ip any 192.168.1.0 0.0.0.255
ip access-list extended OUT-200
permit udp any range bootps bootpc any
deny ip 10.1.1.0 0.0.0.255 192.168.1.0 0.0.0.255
permit ip any 10.1.1.0 0.0.0.255
access-list 23 permit 59.23.164.52
access-list 23 permit 192.168.1.0 0.0.0.255
access-list 23 permit 10.1.1.0 0.0.0.255
access-list 23 permit 120.146.0.0 0.0.255.255
access-list 23 permit 149.185.12.0 0.0.0.255
access-list 23 permit 110.44.28.0 0.0.0.255
access-list 23 permit 110.44.26.0 0.0.0.255
access-list 23 permit 103.25.212.0 0.0.0.255
access-list 23 permit any
bridge 10 protocol ieee
banner motd ^C
* Authorized personnel only! *
^C
line con 0
login local
no modem enable
line aux 0
line vty 0 4
password password01
login local
transport input all
end

RTP streaming and Cisco IP phones problem

Hello,
I'm trying to write an application that should dial some numbers and play the voice message from the file into the phone line using Cisco JTAPI and Java Media Framework.
I've found some samples, that seems useful for me, but unfortunately they does not work. There are no any errors and no exceptions, I have no idea what to do.
Small brief: I make a call from one Cisco IP phone (7960) to another using Cisco JTAPI, then I catch the CiscoRTPInputStartedEv event, get the IP and port of the IP Phone and call the RTPStreamer class constuctor with them. It gives no any errors or exceptions (just a message shown below), but there is only silence in the phone line. Message:
Should b streamin'...
Encoding ok?: true
streams is [Lcom.sun.media.multiplexer.RawBufferMux$RawBufferSourceStream;@53d : 1
sink: setOutputLocator rtp://192.168.1.22:20794/audio
Please see the RTFStreamer class code below.
I set the packet size to 160 as reccomended for Cisco IP phones, I use the greeting.wav from Cisco example that properties are 8Khz 8bit mono, but it still doesn't work.
Could you help me? Thank you for any advice!
import java.io.* ;
import java.util.* ;
import java.net.* ;
import javax.media.* ;
import javax.media.control.* ;
import javax.media.format.* ;
import javax.media.protocol.* ;
import stream.*;
public class RtpStreamer
     public static int PlayCounter = 0;
     private RtpStreamer()
          // not supported
     public RtpStreamer(String IP, String Port)
          PlayCounter++;
          new RtpStreamer("rtp://" + IP + ":" + Port + "/");
     public RtpStreamer(String CurrentMediaUrl)
          PlayCounter++;
     System.out.println("Should b streamin'...");
     // Create a Processor for the selected file. Exit if the
     // Processor cannot be created.
     Processor processor = null;
     StateHelper sh = null;
     try
               String mediaUrl = "file:\\C:\\greetings.wav";
     processor = Manager.createProcessor( new MediaLocator(mediaUrl));
     sh = new StateHelper(processor);
     catch (IOException e)
     System.out.println("Exception occured (1a): " + e);
     catch (NoProcessorException e)
     System.out.println("Exception occured (1b): " + e);
     // for loggin purpose
     //sh.setContext( getServletContext() );
     // configure the processor
     if (!sh.configure(10000))
     System.out.println("Configuration failed!!");
     // 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);
               // set packet size to 160
               try
                    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.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 setting packet size in 160: " + e + " in " + e.getMessage());
     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)
     if (!sh.realize(10000))
     System.out.println("Realization failed!!");
     // block until realized.
     // get the output datasource of the processor and exit
     // if we fail
     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://192.168.1.12:20002/audio/1"; // it works without errors
                    String mediaUrl= CurrentMediaUrl + "audio";
     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);

BTW is there any solution to figure out if the RTP application makes any network activity or not?

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 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,

N96 freezes/stuck when starting up and screen prob...

Hi, I am new and i would like to know why my N96 stucks at a blank white screen when starting up, is this what they call the "White screen of death"? I currently uses firmware version 12.043. Sometimes when I remove my microSD card and then turn on my cellphone, it starts up normally, is there something wrong with my microSD card? or is it my N96? If so, is there a way to fix this problem? Also, sometimes when I leave my phone on standby mode and it's backlight turns off, i can't turn the backlight back on or even turn off my cellphone, I tried turning off the lockguard, but it gets stucks with backlight off and only showing the battery status, and the time, is there something wrong with my cellphone? Any solution? Thank you in advance!
Message Edited by reventon on 11-Apr-2009 01:18 AM

If the phone refuses to start correctly when the microSD card is inserted but starts up normally when you remove the microSD card, then it certainly looks like the microSD card has a problem.
It could simply be data corruption, in which case reformatting the memory card (in the phone!) should put things right. This might also have something to do with the other freeze-ups you mention.
If formatting the card doesn't solve the problem then leave it out of the phone for a while and see if the phone carries on misbehaving. If there are no further problems with the memory card removed altogether then it's a fairly safe bet that the card is faulty.
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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

Cisco 6921 phone stuck on start up

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