SIP Phone 9971

I have CUCM (9.1.2) and Sip phone 9971.
When user hangup phone, and try to dial phone number to PSTN, he waiting about 15 seconds before sip phone make a call. I would to make this time about 1 or 2 seconds. If he call to Polycom server, he does not waiting any time.
I changed any timer - no effects.
I apply "Sip dial rules" nothing changed.
Can anybody tell me how i can fix it?

Is only one user impacted. Normally delays in the dialing are due to below:
1) Overlap in dial plan or
2) Interdigit timeout  T302 (default is 15 sec) that comes into play specially if you are hitting a variable route pattern
So either you can check your dial plan and have specific route patterns or you can change the inter digit time out to something less 5 secs maybe according to your issue. (3 secs is the minimum value you can configure)
To change inter digit timeout T302 go to - Service Parameters, Call Manager service and search T302. Change it to the required value (its in msecs)
-Terry
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  • 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 SIP Phone 9971 won't register on CME 8.6

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

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

  • Cisco SIP Phone 9971 will not register on CME 8.6

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

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

  • Cisco ip phone 9971 registration failed with CME 8.6 on WAN

    Dear  Sir
    Subject: Cisco ip
    phone  9971 registration failed
    with CME 8.6 on WAN
    I have a WAN  simulation with one Cisco2811 router one Cisco1841 router (connected with fast  Ethernet interface) plus one layer 2  switch.
    I upgraded the 2811 IOS to 15.4M and installed CME8.6, when we tried  to add 9971 sip phone If the traffic goes throw LAN (Same subnet- directly from  switch to 2811) everything is fine and the SIP phone (9971) added with no  problem but when I change the subnet and traffic goes throw WAN
    ( from 1841  routed to 2811 with default route) my phone didn't register.
    It will be great  if anybody could help!!
    If you can solve my problem, I will be happy to  compensate.

    Try binding SIP to an interface on which Phone can reach (Use "voice service voip" "sip" and then bind)
    Udit

  • SIP IP Phone 9971 does'nt get dial tone and can't upgrade to the latest Fimware

    hi,
    I have Cisco CME 2901.
    The Ios vesion is 15.3(3)M2. 
    The CME version is 10.0.
    The IP phone 9971 Firmwares are:
    dkern9971.100609R2-9-2-2SR1-9.sebn
    kern9971.9-2-2SR1-9.sebn
    rootfs9971.9-2-2SR1-9.sebn
    sboot9971.031610R1-9-2-2SR1-9.sebn
    sip9971.9-2-2SR1-9.loads
    skern9971.022809R2-9-2-2SR1-9.sebn
    But the problem is the internal call between  the IP Phones don't run. for exemple ip phone with dn 100 unable to call the ip phone with the dn 102. when i issue the call, nothing happen.
    and more, the IP Phone are registered correctly.
    on the phone, theses status messages are displayed:
    File Not Found: United_States/g4-tones.xml
    File Not Found: Sip_English_States/gd-sip.jar
    the commands debugs , nothing happen
    Attached are the resultes of the show commands.
    Do i upgrade my IOS? 
    I've really need helps please

    hey,
    can u try this?
    voice class codec 1
     codec preference 1 g711ulaw
     codec preference 2 g711alaw
     codec preference 3 g729br8
     codec preference 4 g729r8
    then;
    voice register pool  1
     id mac "mac adress"
     session-transport tcp
     type 9971
     number 1 dn 1
     dtmf-relay rtp-nte
     username brk3 password 1234
     codec g711ulaw or voice class codec 1
     no vad
     camera
     video
    then please let me know if it is solved or no

  • 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 ip phone 9971 registration failed with CME 8.5

    Good day!
    I have voice bundle 2951 with 15.1.3(T) IOS version and  a few 9971 ip phones. In addition to it I've downloaded current SIP firmware for these phones (sip9971.9-2-1) and configured my CME as follows:
    voice register global mode cme source-address 192.168.2.254 port 5060 max-dn 10 max-pool 5 load 9971 sip9971.9-2-1 timezone 13 create profile sync 0069043299436028!voice register dn  1 number 104 name 9971 SIP Phone label 104 mwi!voice register pool  1 id mac 688D.ABA4.A88B type 9971 number 1 104 presence call-list dtmf-relay rtp-nte codec g711ulaw no vadvoice service voip sip
      registrar server expires max 3600 min 120
    All SIP firmware files were uploaded to tftp server on CME and shared then:
    tftp-server flash:kern9971.9-2-1.sebn tftp-server flash:rootfs9971.9-2-1.sebn tftp-server flash:sboot9971.031610R1-9-2-1.sebn tftp-server flash:sip9971.9-2-1.loads tftp-server flash:skern9971.022809R2-9-2-1.sebn tftp-server flash:dkern9971.100609R2-9-2-1.sebn tftp-server system:cme/sipphone
    During the boot process all ip settings including option 150 were successfully updated by the phone:
    ip dhcp pool voip_pool   network 192.168.2.0 255.255.255.0   default-router 192.168.2.254   option 150 ip 192.168.2.254
    But when the phone tries to get files on the CME it fails. The ouput of debug tftp events command shows that there is no such configuration file on tftp:
    Jul 28 16:36:14.446: TFTP: read request from host 192.168.2.8(49158) via GigabitEthernet0/1Jul 28 16:36:14.446: TFTP: Looking for SEP68BDABA4A88B.cnf.xmlJul 28 16:36:14.446: TFTP: Sending error 1 No such fileJul 28 16:36:15.522: TFTP: Server request for port 49158, socket_id 0xA2EFEE4 for process 324Jul 28 16:36:15.522: TFTP: read request from host 192.168.2.8(49158) via GigabitEthernet0/1Jul 28 16:36:15.522: TFTP: Looking for XMLDefault.cnf.xmlJul 28 16:36:15.522: TFTP: Opened system:/its/vrf1/XMLDefault.cnf.xml, fd 14, size 3210 for process 324Jul 28 16:36:15.522: TFTP: Sending block 1 (retry 0), socket_id 0xA2EFEE4Jul 28 16:36:15.522: TFTP: Received ACK for block 1, socket_id 0xA2EFEE4Jul 28 16:36:15.522: TFTP: Sending block 2 (retry 0), socket_id 0xA2EFEE4Jul 28 16:36:15.522: TFTP: Received ACK for block 2, socket_id 0xA2EFEE4Jul 28 16:36:15.522: TFTP: Sending block 3 (retry 0), socket_id 0xA2EFEE4Jul 28 16:36:15.526: TFTP: Received ACK for block 3, socket_id 0xA2EFEE4Jul 28 16:36:15.526: TFTP: Sending block 4 (retry 0), socket_id 0xA2EFEE4Jul 28 16:36:15.526: TFTP: Received ACK for block 4, socket_id 0xA2EFEE4Jul 28 16:36:15.526: TFTP: Sending block 5 (retry 0), socket_id 0xA2EFEE4Jul 28 16:36:15.526: TFTP: Received ACK for block 5, socket_id 0xA2EFEE4Jul 28 16:36:15.526: TFTP: Sending block 6 (retry 0), socket_id 0xA2EFEE4Jul 28 16:36:15.526: TFTP: Received ACK for block 6, socket_id 0xA2EFE
    So, i see this file on CME router in system/cme/sipphones direcotory:
    Directory of system:cme/sipphone/  747  -rw-         825  Jul 27 2011 16:10:04 +00:00  featurePolicyDefault.xml  744  -rw-           1  Jul 27 2011 15:59:49 +00:00  OS79XX.TXT  749  -rw-        3761  Jul 27 2011 16:10:04 +00:00  SEP688DABA4A88B.cnf.xml  742  -rw-        1948  Jul 27 2011 16:10:04 +00:00  SIPDefault.cnf  745  -rw-        4376  Jul 27 2011 16:10:04 +00:00  softkeyDefault.xml  746  -rw-        4342  Jul 27 2011 16:10:04 +00:00  softkeyDefault_kpml.xml  743  -rw-          69  Jul 27 2011 16:10:04 +00:00  syncinfo.xml
    So, i dont quiet understand why the registration process was failed. Could anyone tell me if there are restrictions on the SIP firmware and IOS version in this case?
    Thank you.

    linuxchild,
    thank you. i checked this out, but i think it doesnt matter whether tftp-path statement is set or not in this case.
    Configuration files for 9971 are located in system:cme/sipphone directory by default.
    While I have this statement in tftp-server configuration, it should get this file from the server.
    tftp-server system:cme/sipphone
    i tried to configure it accordiing to this example.
    Now i can see xml file in flash direcory:
    30        1948 Aug 1 2011 09:48:52 +00:00 SIPDefault.cnf
    31          69 Aug 1 2011 09:48:52 +00:00 syncinfo.xml
    32        4342 Aug 1 2011 09:48:54 +00:00 softkeyDefault_kpml.xml
    33        4376 Aug 1 2011 09:48:54 +00:00 softkeyDefault.xml
    34        3787 Aug 1 2011 09:48:54 +00:00 SEP68BDABA4A88B.cnf.xml
    35         825 Aug 1 2011 09:48:54 +00:00 featurePolicyDefault.xml
    But i cant still get this file from tftp and have the same error:
    *Aug  1 09:50:27.385: TFTP: read request from host 192.168.2.5(49157) via GigabitEthernet0/1
    *Aug  1 09:50:27.385: TFTP: Looking for SEP68BDABA4A88B.cnf.xml
    *Aug  1 09:50:27.385: TFTP: Sending error 1 No such file
    Has anyone had such problem?

  • Bandwidth required during registration of ip sip phone

    what is the bandwidth requirement of cisco 9971 model phone during registration?

    Thanks a lot Vivek... I went through it...but I think it talks about the bandwidth provisioning for voice traffic..for signalling and once the call is established....I neeed to know the bandwidth required just for registration ...i.e. in the following steps ( only registration without placing any call)   
    1. The phone contacts the TFTP server and requests the Certificate Trust List file .
    2. The phone contacts the TFTP server and requests its SEP<mac-address>.cnf.xml configuration file.
    3. The Phone downloads the default configuration XMLDefault.cnf.xml file from the TFTP server.
    4. The SIP phone requests a firmware upgrade (Load ID file) and upgrades the firmware image automatically when required for a new version of CUCM.
    5. The phone downloads the SIP dial rules configured for that phone.
    6. The phone Establish connection with the primary CUCM and the TFTP server end to end.
    7. The phone Registers with the primary CUCM server listed in its configuration file.
    8. The phone downloads the appropriate localization files from TFTP.
    9. The phone downloads the softkey configurations from TFTP.
    10. The phone downloads custom ringtones (if any) from TFTP.
    Also, I need t o know if the bandwidth required for this process is same for all phone models or different? Specifically, I need  this data for Cisco 9971 model.Please help...Thanks..

  • How do i monitor SIP Phone from a SCCP Phone? UC500

    Hello Community.
    This is my SIP Phone config:
    voice register dn  2
    number 214
    name xxx
    no-reg
    label xxx
    voice register pool  2
    id mac 44E4.D944.9B11
    session-transport tcp
    type 9971
    number 1 dn 2
    number 2 dn 3
    dtmf-relay rtp-nte
    username xxx password xxx
    codec g711ulaw
    And this is the SCCP Phone:
    ephone  12
    device-security-mode none
    mac-address 0011.2014.8248
    ephone-template 16
    username xxx password xxx
    type 7960 addon 1 7914
    button  1:12 2:51 8m11 9m10
    button  10m15 11m13 12m16 13m17
    button  14m18
    Hope you can help, thanks Patrick

    Yes you can.
    Sample config :
    sip-ua
              presence enable
         presence
              presence call-list
              max-subscription 64
              watcher all
              allow subscribe
         ephone-dn 12
              number 2010
              allow watch
              blf-speed-dial 1 214 label "Teset-User"
         voice register dn 2
              number 214
              allow watch
    CME admin guide:
    http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmepres.html#wp1011193
    Please rate replies and mark question as "answered" if applicable.

  • 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

  • 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 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.

  • CUCM 8.6 Dropped call transfers involving SIP phones

    Hi All,
    I am a developer who has been tasked with figuring out why call transfers are being dropped by Cisco CUCM when the original call comes from a SIP phone.  This scenario works:
    Cisco phone calls another Cisco phone, which transfers the original call to a SIP phone
    These scenarios do not work:
    SIP phone calls Cisco phone, which transfers the original call to another Cisco phone
    SIP phone calls Cisco phone, which transfers the original call to another SIP phone
    I have researched the Call Manager traces to the best of my ability, and I see some info in there that could potentially point to the source of the problem.  I am just unable to understand what the trace means:
    10:23:08.672 |//SIP/SIPCdpc(1,74,2342)/ci=24377698/ccbId=175645/scbId=0/active_CcDisconnReq: ccDisconnReq.onBehalfOf=Media : ccDisconnReq.s.sv=2 : ccDisconnReq.c.cv=47 |1,100,63,1.93259^10.10.10.85^*
    10:23:08.672 |//SIP/Stack/Info/0x0/sipConstructContainerContext #### Created container=0xb0b42f58|1,100,71,1.1^*^*
    10:23:08.672 |//SIP/SIPCdpc(1,74,2342)/ci=24377698/ccbId=175645/scbId=0/appendReasonHdr: appendReasonHdr - Invalid Disconnect Cause(cause=47), No Reason Header Appended|1,100,63,1.93259^10.10.10.85^*
    10:23:08.672 |//SIP/SIPCdpc(1,74,2342)/ci=24377698/ccbId=175645/scbId=0/appendRPIDHdrForOriginalCalledParty: SIP device does not Support Orig Dialled Phone nego: 0|1,100,63,1.93259^10.10.10.85^*
    I have been wondering whether this could be a codec issue, however the SIP phones we are using are configured with the following codecs:
    G711U
    G711A
    G722
    ILBC
    GSM
    and our SIP software is  also set to accept the first codec offered by the remote side.  It seems from the SIP client logs that G722 is being used as the codec to communicate with the Cisco phones, but perhaps I'm misinterpreting.
    I have attached a CUCM trace of a call from a SIP phone (ext. 491) to a Cisco handset (ext. 170) where the Cisco handset attempts to transfer the call to another SIP phone (ext. 492).  The trace snippet shown above is from this log.
    I would really appreciate it if someone more experienced with VoIP/SIP/CUCM could take a look and offer any ideas on what the issue might be, and also how we might be able to address it.  I can try to provide more info about our CUCM configuration if needed.
    Thanks in advance!

    Leslie, so here is what I found from the traces....
    To understand the difference we need to understand how cucm performs call transfers from a sccp signalling point and a sip signalling point
    SCCP
    When the transfer key is pressed
    1. CUCM sends a CloseReceiveChannel and StopMediaTransmission to the IP phone involved in active media (referenced by the callids)
    NB, here CUCM updates the call state on the phone to a call state of 8 which is "Hold"
    2.CUCM tells the held party to listen MOH from MOH server
    3.CUCM establishes newcall leg with the intended transfered destination..Once this call is connected
    4.CUCM receives a new Transfer instruction from the transferring phone to connect the held party
    5..CUCM sends a CloseReceiveChannel between the held phone and MOH server (to tear down the media)
    6. Next CUCM sends a CloseReceiveChannel and StopMediaTransmission to the transfering party & transfered party to remove Xferring party from call
    7. finally CUCM sends OpenReceiveChannel between the original called party and the transfered party..and call is done
    For SIP signalling. when the first transfer key is pressed
    1. CUCM sends invite (re-invite) with an inactive SDP (a=inactive) to indicate a break in media path
    2. CUCM sends a Delayed offer to insert MOH or to resume Media stream
    NB: CUCM expects a sendrecv offer with SDP to the DO. (NB:if cucm gets an inactive offer SDP in the 200 OK instead of providing a send-recv offer SDP, the media path remains in an inactive state and causes calls to dropcall will drop),CUCM sends an ACK with sendonly to the 200 OK
    3.CUCM establishes newcall leg with the intended transfered destination..Once this call is connected
    4.CUCM receives a new Transfer instruction from the transferring phone to connect the held party
    5. Next CUCM sends a re-invite with an inactive SDP to indicate a break in media path to MOH (in attempt to complete transfer)
    6.Next CUCM sends an inactive SDP to indicate a break in media path between transfering party & transfered party to remove Xferring party from call
    7. Next CUCM sends a DO re-invite to connect the transfered party. The far end then sends 200 OK with the required SDP to connect the call
    Now having explained all of these, we need to look at where the call is failing for SIP-----SCCP----SIP calls without MTP
    lets look at succesful SCCP-----SCCP-----SIP without MTP
    Point 4 above
    ++++++++Extension 170 presses the transfer button to connect the two calls (Callid=24378483)+++++++++++++
    (0003395) SoftKeyEvent softKeyEvent=4(Trnsfer) lineInstance=1 callReference=24378483
    Point 5 above
    ++++Next CUCM closed the media between extension 160 and MOH server callid=24378480(this is the only active call on this callid)+++
    (0003396) CloseReceiveChannel conferenceID=24378480 passThruPartyID=16777845.  myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a89(10.10.10.137)
    Point 6 Above
    +++++Next cucm closes the call between extension 170 and 490 callid=(24378483)++++++++
    (0003395) CloseReceiveChannel conferenceID=24378483 passThruPartyID=16777847.  myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a8b(10.10.10.139)
    (0003395) StopMediaTransmission conferenceID=24378483 passThruPartyID=16777847.  myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a8b(10.10.10.139)
    Point 6 above for the sip side (since the destination is SIP, to tear down media to SCCP phone, so as to connect the caller to the xfered party)
    +++++++Next CUCM sends a re-invite with a=inactive SDP to the sip phone ++++++++++++
    //SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 62220 index 1890
    [885626,NET]
    INVITE sip:[email protected]:62220;transport=tcp SIP/2.0
    Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK23332dbee978
    From: ;tag=192115~d8e94532-127d-4dca-bba0-64b1675da032-24378484
    o=CiscoSystemsCCM-SIP 192115 2 IN IP4 10.10.10.195
    s=SIP Call
    c=IN IP4 0.0.0.0
    m=audio 24560 RTP/AVP 9 101
    a=rtpmap:9 G722/8000
    a=ptime:20
    a=inactive-----------------------------------------------------Inactive
    Still part of Point 6 for SIP signalling
    ++++++++++++Next sip phone responds with a 200 OK recevonly SDP +++++++++++++++++++
    //SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 10.10.10.104 on port 62220 index 1890 with 683 bytes:
    [885628,NET]
    SIP/2.0 200 OK
    v=0
    o=- 18077 11099 IN IP4 10.10.10.104
    s=yasdjip
    c=IN IP4 10.10.10.104
    t=0 0
    a=ptime:20
    a=recvonly-------------------------------------a=recvonly
    Finally Point 7 above..
    +++++++++++++=Next cucm sends a DO re-invite to extension 492-sip phone++++++++++
    //SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 62220 index 1890
    [885630,NET]
    INVITE sip:[email protected]:62220;transport=tcp SIP/2.0
    Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK233534ffec4a
    From: ;tag=192115~d8e94532-127d-4dca-bba0-64b1675da032-24378484
    To: ;tag=5B0E9816C2CA6D70F3166FB972EDE4C2
    +++++++Next we get a 200 OK from sip phone with sdp=sendrecv+++++++++=
    /SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 10.10.10.104 on port 62220 index 1890 with 683 bytes:
    [885634,NET]
    SIP/2.0 200 OK
    Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK233534ffec4a
    Contact:
    From: ;tag=192115~d8e94532-127d-4dca-bba0-64b1675da032-24378484
    Call-ID: [email protected]
    v=0
    o=- 18077 11099 IN IP4 10.10.10.104
    s=yasdjip
    c=IN IP4 10.10.10.104
    t=0 0
    m=audio 16574 RTP/AVP 9 101
    a=rtpmap:101 TELEPHONE-EVENT/8000
    a=fmtp:101 0-15
    a=ptime:20
    a=sendrecv
    +Now CUCM sends an OpenReceiveChannel and start media xmission to sccp phone (callid=24378480) with media parameters of sip phone++++++
    (0003396) OpenReceiveChannel conferenceID=24378480 passThruPartyID=16777848 millisecondPacketSize=20 compressionType=6(Media_Payload_G722_64k) RFC2833PayloadType=101 qualifierIn=? sourceIpAddr=IpAddr.type:0 ipAddr:0x0a0a0a68000000000000000000000000(10.10.10.104). myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a89(10.10.10.137)
    (0003396) startMediaTransmission conferenceID=24378480 passThruPartyID=16777848 remoteIpAddress=IpAddr.type:0 ipAddr:0x0a0a0a68000000000000000000000000(10.10.10.104)
    remotePortNumber=16574 milliSecondPacketSize=20 compressType=6(Media_Payload_G722_64k) RFC2833PayloadType=101 qualifierOut=?. myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a89(10.10.10.137)
    +++++++++++=Next Phone sends its ACK+++++++++++++++
    (0003396) OpenReceiveChannelAck Status=0, IpAddr=IpAddr.type:0 ipAddr:0x0a0a0a89000000000000000000000000(10.10.10.137), Port=20352, PartyID=16777848
    +++++++++++=Next CUCM sends ACK to 200 OK from SIP Phone+++++++++++
    //SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 62220 index 1890
    [885635,NET]
    ACK sip:[email protected]:62220;transport=tcp SIP/2.0
    Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK23366067b8c0
    From: ;tag=192115~d8e94532-127d-4dca-bba0-64b1675da032-24378484
    To: ;tag=5B0E9816C2CA6D70F3166FB972EDE4C2
    Date: Tue, 19 Feb 2013 21:44:45 GMT
    Call-ID: [email protected]
    Max-Forwards: 70
    CSeq: 103 ACK
    Allow-Events: presence
    Content-Type: application/sdp
    Content-Length: 237
    v=0
    o=CiscoSystemsCCM-SIP 192115 3 IN IP4 10.10.10.195
    s=SIP Call
    c=IN IP4 10.10.10.137
    b=TIAS:64000
    b=AS:64
    t=0 0
    m=audio 20352 RTP/AVP 9 101
    a=rtpmap:9 G722/8000
    a=ptime:20
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    Now at this point all is well...and the call is connected....
    Now here is where the call is failing on the SIP-SCCP-SIP call without MTP
    From Point 2 above, CUCM sends a DO to insert MOH, and then gets response, then sends an ACK to 200 Ok back to SIP Phone..
    //SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 53361 index 1810
    [881160,NET]
    ACK sip:[email protected]:53361;transport=tcp SIP/2.0
    Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK22035ecc1fcb
    From: ;tag=190666~d8e94532-127d-4dca-bba0-64b1675da032-24378214
    To: "492" ;tag=97C34E1FB9A11F83DD8D8F5BA4C87C57
    Date: Tue, 19 Feb 2013 17:38:50 GMT
    Call-ID: 1CCA5149B966DC89AE0F752B8EF86480BC7102DB
    Max-Forwards: 70
    CSeq: 102 ACK
    o=CiscoSystemsCCM-SIP 190666 3 IN IP4 10.10.10.195
    s=SIP Call
    c=IN IP4 10.10.10.195---------------------------------------IP address of MOH server
    t=0 0
    m=audio 4000 RTP/AVP 0--------------------------------MOH port 4000
    a=rtpmap:0 PCMU/8000
    a=ptime:20
    a=sendonly---------------------------------------------------------sendonly
    +++++NOW Point 6 above (SIP) CUCM sends a=inactive to break media path to MOH server to connect caller and xfered party++++++
    //SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 53361 index 1810
    [881161,NET]
    INVITE sip:[email protected]:53361;transport=tcp SIP/2.0
    Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK22045bb7f918
    From: ;tag=190666~d8e94532-127d-4dca-bba0-64b1675da032-24378214
    To: "492" ;tag=97C34E1FB9A11F83DD8D8F5BA4C87C57
    Date: Tue, 19 Feb 2013 17:39:04 GMT
    Call-ID: 1CCA5149B966DC89AE0F752B8EF86480BC7102DB
    Supported: timer,resource-priority,replaces
    Min-SE:  1800
    User-Agent: Cisco-CUCM8.6
    Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
    CSeq: 103 INVITE
    Max-Forwards: 70
    Expires: 180
    Allow-Events: presence
    Remote-Party-ID: ;party=calling;screen=yes;privacy=off
    Contact:
    Content-Type: application/sdp
    Content-Length: 164
    v=0
    o=CiscoSystemsCCM-SIP 190666 4 IN IP4 10.10.10.195
    s=SIP Call
    c=IN IP4 0.0.0.0--------------------------------------------------------------------Media IP is 0.0.0.0
    t=0 0
    m=audio 4000 RTP/AVP 0
    a=rtpmap:0 PCMU/8000
    a=ptime:20
    a=inactive---------------------------------------------------------------------media inactive
    At this point, we should get a response back from the sip phone...
    and here is what we got..
    ++Trying which is expected++++
    //SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 10.10.10.104 on port 53361 index 1810 with 331 bytes:
    [881162,NET]
    SIP/2.0 100 Trying
    Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK22045bb7f918
    From: ;tag=190666~d8e94532-127d-4dca-bba0-64b1675da032-24378214
    Call-ID: 1CCA5149B966DC89AE0F752B8EF86480BC7102DB
    CSeq: 103 INVITE
    To: "492" ;tag=97C34E1FB9A11F83DD8D8F5BA4C87C57
    Content-Length: 0
    ++++++++Then we get a BYE+++++++++++++++
    /SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 10.10.10.104 on port 53361 index 1810 with 576 bytes:
    [881163,NET]
    BYE sip:[email protected]:5060;transport=tcp SIP/2.0
    Via: SIP/2.0/TCP 10.10.10.104:53361;branch=z9hG4bKa2vdQvR7J9OiMyjU;rport
    Contact:
    Max-Forwards: 70
    From: "492" ;tag=97C34E1FB9A11F83DD8D8F5BA4C87C57
    Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
    Supported: replaces, path
    User-Agent: Acrobits Softphone Business/2.4.8
    To: ;tag=190666~d8e94532-127d-4dca-bba0-64b1675da032-24378214
    Call-ID: 1CCA5149B966DC89AE0F752B8EF86480BC7102DB
    CSeq: 3 BYE
    Content-Length: 0
    So this is the root cause of the problem. Your SIP phone does not know how to respond to multiple media break between it and the MOH server.
    The difference between this and the succesful SCCP-SIP--SCCP, is that the held party was a sccp phone, hence the sip phone only has to process one a=inactive SDP message, where as in the SIP-SCCP-SIP, the help party was sip, so the sip phone has to process two a=inactive SDP messages
    Now what is the difference when MTP is involved! A Big difference. MTP stays in the media path. So there is never a break in media and no inactive SDP attribute is sent. The flow looks like below:
    for the initial call...The SIP phone sends its media to MTP and likewise the SCCP phone
    SIP------Media------MTP------------Media-------SCCP Phone
    When the new destination is dialled and transfer is commited,
    SIP-------------media----MTP--------media---------MTP
    The final invoite sent to connect 492 and 491 has MTP as the IP address to connect Media to.
    ++++++++Ivite to 492 ++++++++++++++
    INVITE sip:[email protected]:61303;transport=tcp SIP/2.0
    Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK231a3b24b862
    From: ;tag=192048~d8e94532-127d-4dca-bba0-64b1675da032-24378472
    To: ;tag=78FF5BF6C019A55EA020B69BB6A767E2
    Date: Tue, 19 Feb 2013 21:24:59 GMT
    Call-ID: [email protected]
    Supported: timer,resource-priority,replaces
    Min-SE:  1800
    User-Agent: Cisco-CUCM8.6
    Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
    CSeq: 102 INVITE
    Max-Forwards: 70
    Expires: 180
    Allow-Events: presence
    Remote-Party-ID: ;party=calling;screen=yes;privacy=off
    Contact:
    Content-Type: application/sdp
    Content-Length: 214
    v=0
    o=CiscoSystemsCCM-SIP 192048 1 IN IP4 10.10.10.195
    s=SIP Call
    c=IN IP4 10.10.10.195---------------------------------------------------------------the MTP ip address
    t=0 0
    m=audio 25038 RTP/AVP 0 101
    a=rtpmap:0 PCMU/8000
    a=ptime:20
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    +++++++Invite to 491 +++++++++++++++++
    //SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.94 on port 50376 index 1887
    [885429,NET]
    INVITE sip:[email protected]:50376;transport=tcp SIP/2.0
    Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK231b78d1b56
    From: ;tag=192046~d8e94532-127d-4dca-bba0-64b1675da032-24378467
    To: "491" ;tag=F13CE94DE942C47680356A647DC7F916
    Date: Tue, 19 Feb 2013 21:24:59 GMT
    Call-ID: AE7045FFB2D8D9C28D54651473A14A5D41B5B93C
    Supported: timer,resource-priority,replaces
    Min-SE:  1800
    User-Agent: Cisco-CUCM8.6
    Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
    CSeq: 101 INVITE
    Max-Forwards: 70
    Expires: 180
    Allow-Events: presence
    Remote-Party-ID: "Leslie2" ;party=calling;screen=no;privacy=off
    Contact:
    Content-Type: application/sdp
    Content-Length: 237
    v=0
    o=CiscoSystemsCCM-SIP 192046 1 IN IP4 10.10.10.195
    s=SIP Call
    c=IN IP4 10.10.10.195----------------------------------------MTP
    b=TIAS:64000
    b=AS:64
    t=0 0
    m=audio 25030 RTP/AVP 0 101
    a=rtpmap:0 PCMU/8000
    a=ptime:20
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    Wao! That was a long one isnt it...It was fun too.
    So now you can look at your sip phones and see if they can accept two inactive sdp messages within the same call. That way you can remove MTP. otherwise you will have MTP involved in every single call involving a sip phone, even if they do not involve transfers
    Please rate all useful posts
    "opportunity is a haughty goddess who waste no time with those who are unprepared"

  • 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                                    

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