Register SIP Phone

Hello;
Can u help me how to register Cisco SIP Phone on CCME 4.0(2)
The doc on cisco.com is not available
Thanks

Hi, can you try:
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_configuration_guide_chapter09186a00807e13a7.html
procedures for boths\ SCCP and SIP are explained.
Hope this helps, please rare post if it does!

Similar Messages

  • 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#

  • What are the mandatory fields needed to setup/register the SIP phone manually in CUCM

    What are the mandatory fields needed to setup/register the SIP phone manually.Also, if someone can let me know the mandatory fields for Cisco based SIP phone and also the third party SIP hard phones like Avaya or any other Third party SIP phones both Soft phone and physical phone requirements...in CUCM
    Please suggest...I need to know if MAC address is mandatory for all Cisco SIP phone to setup 

    http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/7_1_2/ccmcfg/bccm-712-cm/b09sip3p.html
    http://www.cisco.com/c/en/us/support/docs/voice-unified-communications/unified-communications-manager-version-70/112110-phone-add-00.html

  • 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.

  • SIP phone registering on SIP trunk

    Hi,
    i have a UC 500 connected to our phone provider using a SIP trunk.
    All the phones are SPA508 G
    All is working fine !
    Then, some days ago i added a SIP phone (extention 350) on the UC500, that also worked fine, and then after some minutes all our incoming/outgoing calls were blocked.
    I called my provider that told me that our IP was banned because they have seen to much registration attempt from a bad user that was "350"
    I can confirm with a "sh sip-ua register status" command that i had two sip registration : my SIP trunk and the SIP phone
    Then it seems that the UC 500 is trying to register the SIP phone on the SIP trunk ?
    What am i doing wrong ?
    Is there a command to avoid that ?
    Bellow is how the SIP phone and the SIP trunk are configured
    Many thanks for your help, i was unable to find anything about that, but i guess somebody already had this problem !
    The SIP phone -------------------------------------------------------------------------
    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 none
     modem passthrough nse codec g711ulaw
     sip
      registrar server expires max 3600 min 120
      no update-callerid
    voice class codec 1
     codec preference 1 g711ulaw
     codec preference 2 g729r8
    voice register global
     mode cme
     source-address 10.1.1.1 port 5060
     max-dn 20
     max-pool 20
     load 9971 sip9971.9-2-2
     load 9951 sip9951.9-2-2
     load 8961 sip8961.9-2-2
     load 7971 term71.default
     authenticate register
     authenticate realm xxxxxx.com
     timezone 13
     hold-alert
     mwi stutter
     mwi reg-e164
     create profile sync 0636240803635305
    voice register dn  1
     number 350
     name Conference
     label Conference
    voice register pool  1
     id mac 1234.1234.1234
     number 1 dn 1
     username 350 password 1234
     codec g711ulaw
    The SIP trunk ----------------------------------------------------------------------
    sip-ua
     credentials username user1234 password 1234 realm sipgw9.provider.com
     authentication username user1234 password 1234 no remote-party-id
     retry invite 2
     retry register 10
     timers connect 100
     registrar dns:sipgw9.provider.com expires 3600
     sip-server dns:sipgw9.provider.com

    I'm still searching on the forum, and maybe i found somthing related to my problem, not sure... any advice ?
    Disable outbound proxy on voice register global as by default it will use the outbound proxy configured on the system which would not make sense
    voice register global
      no outbound-proxy
    found there : https://supportforums.cisco.com/discussion/10760741/uc500-sip-server-and-sip-trunk

  • Cisco UC5xx 8.6 Support for 99xx SIP phones using CCA 3.2.1

    Hi Friends,
    I was looking through posts here in the SMB commuity, the SWP 8.6 for UC5xx, the RN for CCA 3.2.1, and the OLH for CCA 3.2.1, and found a nice thread that will help anyone wanting to get a 9951 or 9971 SIP phone to operate on the UC5xx after upgrade to the Cisco IOS {15.1(4)M4} bringing CME 8.6 to Telephony Services and selection of the SIP 9-2-2 phone loads (included in the SWP) for 99xx.
    My only inquiry for Cisco to check, would be why isnt this documented in the release notes,
    https://supportforums.cisco.com/servlet/JiveServlet/previewBody/26979-102-1-67567/cca_3_2_2_relnotes.pdf
    since CCA doesnt seem to add the 'load 99xx sip99xx.9-2-2' statement, the 'tftp-path flash:', followed by the 'create profile' under VOICE REGISTER GLOBAL?
    If it is supposed to work, then I would alert you that it did not.  After Upgrading CCA to 3.2.1 and then upgrading the UC5xx to SWP 8.6, the adminisrtrator manually adds the 99xx phone by entering MAC and Type under Configure> Telephony> Users/Extensions> Users and Phones: ADD button.  Nothing special, just a normal extension on one button, Video Enabled, and a VM box created.  This allows the phone to register just fine, but it doesnt automatically upgrade to 9-2-2 due to the missing bold commands above.
    I think if this were a known defect, it would have been documented in the RNs, so I raise it to your attention.
    Which operation should have added these commands?
    Can you let us know if this is an anomaly or if everyone will encounter this?
    Thanks kindly,
    Steve

    Yeah uh beleive me Steven, I have tried everything and every location to get these phones working and nothing did. Other people have the same issue (thread here somewhere) luckily mine are only out 1 hour others are out many hours.
    Thanks,
    Bob James

  • Cisco CP-78XX SIP Phone Pickup Not Work on CME

    Hi,
    I configured some SIP phones (CP-7821, CP-7841) with pickup function. Is it the Pickup / GPickup soft keys not function as the SIP phone? If yes, then I can use the FAC to access that? And I tried the FAC std. / custom as the pickup / gpickup  .. both not work ... I don't know how to use the FAC on CME? As the FAC std., if I pickup local, that I should press (**3) > call?
    Ref.:
    http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucme/admin/configuration/guide/cmeadm/cmecover.html#45535
    http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucme/admin/configuration/guide/cmeadm/cmefacs.html#30064
    This is the configuration:
    CME-SIP-Phone#sh run
    Building configuration...
    Current configuration : 5413 bytes
    ! Last configuration change at 11:06:12 UTC Fri Nov 28 2014 by mtlops
    version 15.4
    no service pad
    service tcp-keepalives-in
    service tcp-keepalives-out
    service timestamps debug datetime msec localtime show-timezone
    service timestamps log datetime msec localtime show-timezone
    service password-encryption
    service sequence-numbers
    hostname CME-SIP-Phone
    boot-start-marker
    boot system flash:c2900-universalk9-mz.SPA.154-2.T1.bin
    boot-end-marker
    ! card type command needed for slot/vwic-slot 0/0
    enable secret 5 $XXXXXXXXXXXXXXXXXXXXXXXX
    aaa new-model
    aaa authentication login default local
    aaa authorization console
    aaa authorization exec default local
    aaa session-id common
    ip cef
    no ipv6 cef
    multilink bundle-name authenticated
    stcapp feature access-code
    voice-card 0
     dspfarm
     dsp services dspfarm
    voice service pots
    voice service voip
     ip address trusted list
      ipv4 10.118.0.0 255.255.255.0
     allow-connections h323 to h323
     allow-connections h323 to sip
     allow-connections sip to h323
     allow-connections sip to sip
     supplementary-service h450.12
     no supplementary-service h225-notify cid-update
     redirect ip2ip
     fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
     h323
      no h225 timeout keepalive
      call preserve
     sip
      bind control source-interface GigabitEthernet0/0
      bind media source-interface GigabitEthernet0/0
      registrar server expires max 600 min 60
    voice class codec 1
     codec preference 1 g711ulaw
     codec preference 2 g711alaw
     codec preference 3 g729r8
    voice class h323 1
      h225 timeout tcp establish 3
      call preserve
    voice class custom-cptone ABC-Company
     dualtone disconnect
      frequency 425
      cadence 500 500
    voice register pool-type  7821
     description Cisco IP Phone 7821
     reference-pooltype 6921
    voice register pool-type  7841
     description Cisco IP Phone 7841
     reference-pooltype 6941
    voice register global
     mode  cme
     source-address 10.118.0.10 port 5060
     timeouts interdigit 2
     max-dn 200
     max-pool 100
     authenticate register
     authenticate realm all
     timezone 42
     time-format 24
     date-format D/M/Y
     mwi stutter
     mwi reg-e164
     voicemail 5000
     call-feature-uri pickup http://10.118.0.10/pickup
     call-feature-uri gpickup http://10.118.0.10/gpickup
     tftp-path flash:
     file text
     create profile sync 0001170446349417
     ntp-server 10.118.0.10 mode unicast
     ip qos dscp af11 media
     ip qos dscp cs2 signal
     ip qos dscp af43 video
     ip qos dscp 25 service
     camera
     video
    voice register dn  2
     number 1000
     pickup-call any-group
     pickup-group 1
     name BB Leung
     label BB Leung
    voice register dn  3
     number 1001
     pickup-call any-group
     pickup-group 1
     name CC Chan
     label CC Chan
    voice register dn  4
     number 1002
     pickup-call any-group
     pickup-group 1
     name DD Leung
     label DD Leung
    voice register dn  50
     mwi
    voice register template  1
     softkeys hold  Newcall Resume
     softkeys idle  Newcall Redial Gpickup Pickup Cfwdall DND
     softkeys seized  Cfwdall Endcall Redial
     softkeys connected  Confrn Endcall Hold Trnsfer
    voice register pool  1
     busy-trigger-per-button 1
     id mac A8XX.XXXX.XXXX
     type 7841
     number 1 dn 2
     template 1
     dtmf-relay sip-notify
     username 1001 password 112233
     codec g711ulaw
     no vad
    voice register pool  2
     busy-trigger-per-button 1
     id mac 50XX.XXXX.XXXX
     type 7841
     number 1 dn 3
     template 1
     dtmf-relay sip-notify
     username 1002 password 112233
     codec g711ulaw
     no vad
    voice register pool  3
     busy-trigger-per-button 1
     id mac 00XX.XXXX.XXXX
     type 7821
     number 1 dn 4
     template 1
     dtmf-relay sip-notify
     username 1003 password 112233
     codec g711ulaw
     no vad
    license udi pid CISCO2921/K9 sn FHK1407F25D
    license accept end user agreement
    license boot c2900 technology-package uck9
    hw-module pvdm 0/0
    hw-module sm 1
    username mtlops privilege 15 secret 5 $1$0qqx$1WGdfRW.flJrwmY7k8eUy0
    redundancy
    interface Embedded-Service-Engine0/0
     no ip address
     shutdown
    interface GigabitEthernet0/0
     ip address 10.118.0.10 255.255.255.0
     duplex auto
     speed auto
    interface GigabitEthernet0/1
     no ip address
     shutdown
     duplex auto
     speed auto
    interface GigabitEthernet0/2
     no ip address
     shutdown
     duplex auto
     speed auto
    interface SM1/0
     no ip address
     shutdown
     service-module fail-open
    interface SM1/1
     no ip address
    interface Vlan1
     no ip address
    ip forward-protocol nd
    no ip http server
    no ip http secure-server
    ip route 0.0.0.0 0.0.0.0 10.118.0.1
    control-plane
    mgcp behavior rsip-range tgcp-only
    mgcp behavior comedia-role none
    mgcp behavior comedia-check-media-src disable
    mgcp behavior comedia-sdp-force disable
    mgcp profile default
    dspfarm profile 1 conference
     codec g711ulaw
     codec g711alaw
     codec g729ar8
     codec g729abr8
     codec g729r8
     codec g729br8
     maximum sessions 7
     associate application SCCP
     shutdown
    gatekeeper
     shutdown
    telephony-service
     max-conferences 8 gain -6
     transfer-system full-consult
     fac standard
    line con 0
    line aux 0
    line 2
     no activation-character
     no exec
     transport preferred none
     transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
     stopbits 1
    line 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
     transport input all
    scheduler allocate 20000 1000
    end
    CME-SIP-Phone#sh telephony-service fac
      telephony-service fac standard
        callfwd all **1
        callfwd cancel **2
        pickup local **3
        pickup group **4
        pickup direct **5
        park **6
        dnd **7
        redial **8
        voicemail **9
        ephone-hunt join *3
        ephone-hunt cancel #3
        ephone-hunt hlog *4
        ephone-hunt hlog-phone *5
        trnsfvm *6
        dpark-retrieval *0
        cancel call waiting *1

    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.

  • CME SIP Phone Calls in one-way (inside local network)

    Hello everyone, first time here, need a little help.
    I'm having some trouble to find a solution to the following problem.
    Recently I've installed CME 9.1 using the router 2921. Most of the phones are SIPs, model 3905 (around 20 of them), with the last firmware updated.
    Some users are complaining one way audio issue in internal calls, from a extension to another (only in sip phones)
    With Wireshark capture I could see that RTP packets are being sent and receive by the router and not directly trough the phones. Is this normal in CME? When a call with problems occours (one way audio) there is no audio in one way, but router still sends confort noise packets.
    Here is my config.
    Thanks for any help.
    Martin
    ##################################################################################33
    System returned to ROM by power-on
    System restarted at 11:29:23 BR Tue Jan 29 2013
    System image file is "flash0:c2900-universalk9-mz.SPA.152-4.M2.bin"
    Last reload type: Normal Reload
    Last reload reason: power-on
    voice service voip
    no ip address trusted authenticate
    allow-connections h323 to h323
    allow-connections h323 to sip
    allow-connections sip to h323
    allow-connections sip to sip
    fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
    sip
      registrar server expires max 3600 min 120
    voice register global
    mode cme
    source-address 10.3.245.1 port 5060
    max-dn 60
    max-pool 70
    load ATA-187 ATA187.9-2-3-1
    load 3905 CP3905.9-2-1-0
    authenticate realm all
    timezone 17
    time-format 24
    date-format D/M/Y
    tftp-path flash:
    file text
    create profile sync 0094230880392697
    network-locale U1
    user-locale U1 load /CME-locale-pt_BR-Portuguese-8.8.2.5.tar
    ntp-server 10.3.244.7 mode directedbroadcast
    voice register dn  1
    number 9006
    name Sala_Reuniao_02
    label Sala de Reuniao 2
    voice register dn  2
    number 9007
    name Sala_Reuniao_03
    voice register dn  3
    number 9008
    name Sala Reuniao 04
    voice register pool  1
    id mac 8478.ACE6.09A2
    type 3905
    number 1 dn 1
    template 1
    codec g711ulaw
    voice register pool  2
    id mac 8478.ACE6.0573
    type 3905
    number 1 dn 2
    codec g711ulaw
    voice register pool  3
    id mac 5897.1ECD.8F8D
    type 3905
    number 1 dn 3
    codec g711ulaw
    interface GigabitEthernet0/0
    no ip address
    duplex auto
    speed auto
    interface GigabitEthernet0/0.220
    encapsulation dot1Q 220
    ip address 10.3.245.1 255.255.255.0
    ip helper-address 10.3.244.71
    h323-gateway voip bind srcaddr 10.3.245.1
    telephony-service
    max-ephones 5
    max-dn 5 no-reg both
    ip source-address 10.3.245.1 port 2000
    timeouts interdigit 5
    timeouts busy 12
    system message  XXXXXXXX
    cnf-file location flash:
    cnf-file perphone
    user-locale U2 load CME-locale-pt_BR-Portuguese-8.8.2.5.tar
    user-locale 2 PT
    network-locale U2
    load 7925 CP7925G-1.4.1SR1.LOADS
    load 6941 SCCP69xx.9-2-1-0.loads
    time-zone 17
    time-format 24
    date-format dd-mm-yy
    max-conferences 8 gain -6
    dn-webedit
    time-webedit
    transfer-system full-consult
    transfer-pattern .T
    create cnf-files version-stamp Jan 01 2002 00:00:00
    ephone-dn  1  dual-line
    number 9001
    ephone  1
    mac-address D867.D9E6.F57F
    ephone-template 1
    type 6941
    button  1:1

    Hi ,
    We have upgarded the the firmware to the  3905.9-2-2ES2 , but show voice register pool phone-load still shows the old firmware, but the phoen itself is showing the new upgraded version on the dsiplay ...any advice is highly appricated,
    ADM-CME9#show voice register pool phone-load
    Pool Device Name     Current-Version             Previous-Version
    ==== =============== =========================== ===========================
    1    SEP7081053DE72F Cisco/SPA502G-7.4.8a                                  
    3    SEP34BDC8C6C412 Cisco-CP3905/9.2.1                                    
    4    SEP34BDC8C64561 Cisco-CP3905/9.2.1                                    
    5    SEP54781AE1F531 Cisco-CP3905/9.2.1                                    
    6    SEP54781AE171D2 Cisco-CP3905/9.2.1                                    
    10   SEP54781AE1F544 Cisco-CP3905/9.2.1                                    
    15   SEP1CE6C77323CD Cisco-CP3905/9.2.1                                    
    16   SEP58971E282A23 Cisco-CP3905/9.2.1                                    
    17   SEP58971E2822A8 Cisco-CP3905/9.2.1                                    
    19   SEP1CE6C77321F3 Cisco-CP3905/9.2.1                                    
    30   SEP54781AE171E2 Cisco-CP3905/9.2.1                                    
    31   SEP54781AE16FD4 Cisco-CP3905/9.2.1                                    
    32   SEP54781AE16F2F Cisco-CP3905/9.2.1                                    
    33   SEP54781A1C77FD Cisco-CP3905/9.2.1                                    
    34   SEP54781A1C77DC Cisco-CP3905/9.2.1                                    
    35   SEP54781AE17527 Cisco-CP3905/9.2.1                                    
    36   SEP54781AE17766 Cisco-CP3905/9.2.1                                    
    37   SEP54781AE1731A Cisco-CP3905/9.2.1                                    
    38   SEP54781AE08B8D Cisco-CP3905/9.2.1                                    
    39   SEP54781AE123B1 Cisco-CP3905/9.2.1                                    

  • CME SIP phone outside call issue

    Dear all,
    i have cme version 9.1 on router 2921 with 7962 sccp phones and 3905 sip phone.
    when i place outside call ( to pstn) using the below dial peer, call is processed. 
    when the call is answered by the autoattendent of the called company ( assume i called x company)  , i cant press any other numbers using the sip phones.
    i mean if i want to press zero for help or internal extension of the x company, these pressed numbered are not recognized by the analog panasonic PBX of the x company.
    Sccp phones works well.
    Any help please and below is the dial-peer.
    dial-peer voice 1003 pots
     trunkgroup 1
     corlist outgoing CITIES
     description CALLING CITIES
     destination-pattern 90[1-9]......
     forward-digits 8
    voice service voip
     allow-connections h323 to h323
     allow-connections h323 to sip
     allow-connections sip to h323
     allow-connections sip to sip
     no supplementary-service sip handle-replaces
     fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
     sip
      bind control source-interface GigabitEthernet0/2.10
      bind media source-interface GigabitEthernet0/2.10
      registrar server expires max 36000 min 600
    voice class codec 5
     codec preference 1 g729r8
     codec preference 2 g711ulaw
    voice register global
     mode cme
     source-address 10.100.4.20 port 5060
     max-dn 200
     max-pool 100
     load 3905 CP3905.9-2-1-0.loads
     authenticate register
     timezone 31
     date-format D/M/Y
     voicemail 177
     tftp-path flash:
     create profile sync 000473524028932A
     conference hardware
    voice register dn  1
     number 109
     allow watch
     pickup-call any-group
     pickup-group 170
     shared-line max-calls 3
    voice register pool  1
     id mac 6C99.8984.9678
     type 3905
     number 1 dn 1
     template 1
     dtmf-relay sip-notify
     voice-class codec 5
     username SFD1 password SFD1
    thanks

    Hi Yahsiel,
    firstly thanks for help, secondly if you don't mind i want to ask you the below if possible:
    1- in my cme, is there a way when i call an internal extension (e.g 110) from an internal phone it rings normally but when i call from outside-->autoattendent answers-->when i press 110 it get transferred to another phone (e.g 111)....????
    2- when i call from outside(pstn) to the cme -->when the plar command is directly to the internal extension the caller id appears but when the autoattendent answers and then transfer to the operator (by pressing zero) the caller id appears as unknown number ??????
    3- is the 3905 sip phone support 1Gbps when connected to the PC, as after connecting the phones to the PCs the speed decreased up to 100Mbps?? or it is another matter?
    (poe switches is cisco SG200)
    regards,

  • Incoming calls issue in Third Party SIP Phone

    Hi,
    Yesterday I configured my third party sip phone which is yealink in this case on cucm and successfully registered it with cucm, despite of registration i have some calling issue in this phone. I am able to make outbound calls from this phone to any other phone however issue is related to inbound calls.I tried calling its DN from anywhere but call disconnect after sometime. Also didnt get any proper sip session trace in RTMT. Kindly suggest some step to sortout this issue.
    Thanks

    Dear Manish,
    Call normally dicsonnected after 30-40 sec with termination code 102 in session trace. PFB SDI  trace with 5030 is Thirdparty sip phone and 5033 is c7945. Looking forward for your suggestion.
    CallingPartyNumber=5033
    |DialingPartition=
    |DialingPattern=5030
    |FullyQualifiedCalledPartyNumber=5030
    |DialingPatternRegularExpression=(5030)
    |DialingWhere=
    |PatternType=Enterprise
    |PotentialMatches=NoPotentialMatchesExist
    |DialingSdlProcessId=(0,0,0)
    |PretransformDigitString=5030
    |PretransformTagsList=SUBSCRIBER
    |PretransformPositionalMatchList=5030
    |CollectedDigits=5030
    |UnconsumedDigits=
    |TagsList=SUBSCRIBER
    |PositionalMatchList=5030
    |VoiceMailbox=
    |VoiceMailCallingSearchSpace=PT-LHR-LOCAL:PT-Local:Unityvmpt:PT-F6-Local:PT-ISL-LOCAL:PT-KHI-LOCAL:PT_Operator_LHR:PT_Operator_KHI:PT_Operator_ISL
    |VoiceMailPilotNumber=7103
    |RouteBlockFlag=RouteThisPattern
    |RouteBlockCause=0
    |AlertingName=Syed Ahmer
    |UnicodeDisplayName=Syed Ahmer
    |DisplayNameLocale=1
    |OverlapSendingFlagEnabled=0
    12:17:38.028 |//SIP/SIPUdp/wait_SdlSPISignal: Outgoing SIP UDP message to 172.16.200.21:[5062]:
    [23928282,NET]
    INVITE sip:[email protected]:5062 SIP/2.0
    Via: SIP/2.0/UDP 10.100.200.11:5060;branch=z9hG4bK1ca0cc6e317649
    From: "Syed Ahmer" ;tag=8787406~039e2a80-8561-4586-8954-d01ed2aa12c8-246211918
    To:
    Date: Thu, 30 Jan 2014 07:17:38 GMT
    Call-ID: [email protected]
    Supported: timer,resource-priority,replaces
    Min-SE:  1800
    User-Agent: Cisco-CUCM8.5
    Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
    CSeq: 101 INVITE
    Expires: 180
    Allow-Events: presence
    Send-Info: conference, x-cisco-conference
    Alert-Info:
    Contact:
    Remote-Party-ID: "Syed Ahmer" ;party=calling;screen=yes;privacy=off
    Max-Forwards: 70
    Content-Length: 0
    |14,100,50,1.14103336^10.163.14.4^SEP00230432C828
    12:17:38.028 |EnvProcessUdpPort - EnvProcessUdpHandler::fireSignal() varId = 0|14,100,50,1.14103336^10.163.14.4^SEP00230432C828
    12:17:38.028 |EnvProcessUdpHandler::fireSignal - SEND: index = 0, handler = 0xaf299320|*^*^*
    12:17:38.028 |EnvProcessUdpPort::fireSignal - SEND, destination = 172.16.200.21:5062|*^*^*
    12:17:38.028 |EnvProcessUdpPort - EnvProcessUdpHandler::send(buff, 850, 172.16.200.21:5062)|*^*^*

  • SIP- h323 in a AS5850 - Not able to send h323 calls coming from a SIP Phone

    Dear All!
    I have an AS5850 configured as a SIP Gateway and as a H323 Gateway. I'm planning to use this equipment as an interconnection point between PSTN,SIP and H323.
    I already have a functional H323 Network with ISDN trunks to the pstn and it is working fine. I added SIP configuration to the AS5850 in order to be able to route calls out to the PSTN or H323 remote ends coming from a SIP Phone registered with a third-party SIP Proxy.
    When the calls coming from the SIP Phone goes to a PSTN destination the calls completes properly, but i am having problems trying to send calls coming from the SIP phone to a remote h323 gateway(also cisco)
    Attached is my configuration and the error i'm getting in my cdr. It seems that the "ext" number of the phone is being used as destination string in the last call leg, but i'm not sure.
    Please Help!
    dial-peer voice 100 pots
    application session
    destination-pattern 5T
    port 2/6:D
    forward-digits all
    dial-peer voice 102 pots
    application session
    destination-pattern 044T
    port 2/6:D
    forward-digits all
    dial-peer voice 103 voip
    application session
    incoming called-number 001T
    destination-pattern 001T
    session protocol sipv2
    session target ipv4:20X.21X.17X.1X
    tech-prefix 10511
    sip-ua
    sip-server ipv4:20X.6X.14X.18X
    CDR ERROR:
    .Mar 24 2004 18:31:42.620 GMT: %VOIPAAA-5-VOIP_CALL_HISTORY: CallLegType 2, ConnectionId 9F74CE17 7D2A11D8 82A09B41 D2C3D418, SetupTime .18:31:42.470 GMT Wed Mar 24 2004, ***PeerAddress 2006***, PeerSubAddress , DisconnectCause 3 , DisconnectText no route to destination (3), ConnectTime .18:31:42.620 GMT Wed Mar 24 2004, DisconnectTime .18:31:42.620 GMT Wed Mar 24 2004, CallOrigin 2, ChargedUnits 0, InfoType 2, TransmitPackets 0, TransmitBytes 0, ReceivePackets 0, ReceiveBytes 0
    Thanks.
    Attached you can find the debug ccsip messages output.

    There are 2 solutions here.
    1. Use of SIP/H.323 Signalling Gateway as the protocol convertor. Search google will yield heaps of hits on this subject. Product available both commercial and open source, trial, etc. Using this method means that the SIP End Point will communicate with H.323 End Point without going out the PSTN. I believe this is what you want to achieve in the long term. You are trying the AS5xxx as the protocol convertor for you, which it will not work. A call flow will be something like SIP IP Phone->SIP Server->SIP-to-H.323 Gateway->H.323 Gatekeeper->H.323 End Point. Of couse there is a SIP server that do the protocol convertor in the same box but the functionality is the still the same. Performance and concurrent call setup differ from products to products. Going for this solution would require you to find such products and test it on the your network.
    2. If you do not wish to try on Soluton 1, this solution is a workaround way by not getting device but using the existing equipment that you have right now. Onto whether this good long term solution for depends on what you want to achieve both in term of commercially and technically. A call flow will be SIP End Point->SIP Server->Voice Gateway (AS5xxx)->PSTN Switch(ISDN/PRI)->Voice Gateway->H.323 Gatekeeper>H.323 End Point. The key is the Voice session must traverse the ISDN link. In other words your dial pattern must be setup is such as way that will go out thru the dial peer pots to pstn switch then come back to another dial-peer pots. I am not saying this is the most efficient way of doing it, I merely suggesting a workable way to achieve your desired goal without soluton 1.
    Hopes you get better understanding now.
    Thanks
    SSng

  • 3rd Party SIP phone to CUCM via SIP Proxy

    Hi all,
    This is the scenario i'm currently working on :
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    The SIP proxy basically terminates everything (REGISTER, INVITE, etc), including the RTP stream.
    I can register the 3rd party SIP phone to CUCM and in CUCM and  i can see SIP Proxy IP Address as the registered address of the phone.
    Calls from the 3rd party SIP phone to internal Cisco or internal 3rd party SIP phone and vice versa work like charm.
    The only (fatal) problem is i can only register 1 3rd party SIP phone to CUCM via this SIP proxy.
    Since this SIP Proxy always use its internal IP Address and port 5060 (TCP) as its source of registration, CUCM sees multiple registrations for multiple extensions (users) come from a single IP and port, and rejects the second registration request.
    It seems that CUCM binds a digest user to an IP address and port, therefore cannot accept multiple registrations from a single IP and port.
    Can anyone clarify this?  Or is there any way around this?
    I'm using CUCM 8.6.2 and CUCM 9.X (both do not work).
    Regards,
    Christian

    This is most likely because of the following...
    Because third-party SIP phones do not send a MAC address, they must identify themselves by using digest authentication.
    The REGISTER message includes the following header:
    Authorization: Digest username="xxxxxxxxxx",realm="ccmsipline",nonce="GBauADss2qoWr6k9y3hGGVDAqnLfoLk5",uri="sip:172.18.197.224",algorithm=MD5,response="126c0643a4923359ab59d4f53494552e"
    The username, xxxxxxxxxxx, must match an end user that is configured in the End User Configuration window of Cisco Unified CallManager Administration. The administrator configures the SIP third-party phone with the user; for example, swhite, in the Digest User field of Phone Configuration window.
    See the following document.
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  • Add third party SIP Phone to CCM 5

    'm not able to register this SIP Phone to the CCM5.0. I have device license that cater all IP Phone models.(LIC-CM-DL-100=)
    I got error message " Login Forbidden" "timeout" in the IP Phone.
    In the CCM, I got this message in Phone COnfig Window
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    Can you explain on how to register this 3rd party IP phone to CCM?
    Is it CCM able to support SIP Phone?

    Hi,
    This is most likely because of the following...
    Because third-party SIP phones do not send a MAC address, they must identify themselves by using digest authentication.
    The REGISTER message includes the following header:
    Authorization: Digest username="swhite",realm="ccmsipline",nonce="GBauADss2qoWr6k9y3hGGVDAqnLfoLk5",uri="sip:172.18.197.224",algorithm=MD5,response="126c0643a4923359ab59d4f53494552e"
    The username, swhite, must match an end user that is configured in the End User Configuration window of Cisco Unified CallManager Administration. The administrator configures the SIP third-party phone with the user; for example, swhite, in the Digest User field of Phone Configuration window.
    See the following document.
    http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/5_1_3/ccmcfg/b09sip3p.html
    Hope this helps, if so please rate.
    Regards,
    Dave

  • CUCM Third Party SIP Phone "Time" issue

    Hi Team,
    we have setup with Avaya 1230 SIP Phone,
    and this phone we added to CUCM using "Third Party Basic SIP Phone" option.
    Once registered with Call Manager "Date and time" in SIP Phones was showing fine.
    we have reset the entire device pool, after that all the Avaya 1230 SIP Phone "Time" is showing +1 hour from the normal time.
    How we can reslove this issue.
    CUCM Version: 9.1(2a).
    SIP Phone Model: Avaya 1230

    Thanks for the Suggestion Manish,
    I have tried the same But its not working,
    In the phone level we have the option to change the "Time Zone", The same we have changed to GMT+5:30 Indian Standard Time.
    Any other suggestion....

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