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,

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     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: Third Party SIP Phone "Caller ID" is not displaying for outgoing calls

    Hi Team,
    we are running CUCM 9.1(2a),
    we have integrated Third Party SIP Phone(Avaya 1230 SIP Phone) with CUCM,
    Issue: Third Party SIP Phone "Caller ID" is not displaying for outgoing calls, we are able to see only the dailed Number,
    When "A" calls to "B", "A" can see only the dailed number of "B" but not the "Caller ID"
    Regards
    Ananthakumar

    Are A and B both Avaya phones?
    So it looks like you're not seeing the alerting name/connected name getting updated then?  Do you have alerting names configured on the directory numbers?  Might need to take a look at the SIP messaging to see if the alerting name/connected name is being sent to the Avaya phones and maybe they just aren't displaying it.  Might just be something that needs to be tweaked in the 46xxsettings.txt file.

  • SIP to SIP Call Failures on CME to CME - sip-ua conflict/issue?

    Hi,
    I have two existing CME systems which I wish to allow internal calls between. These calls will go over an IPSec VPN. However the calls are failing.
    Phones DN22xx - London CME 2801 - PIX505 --- Internet ---ASA5505 - India CME 2801 - Phones DN400x
    I have configured dial peers on both CME's and the IPSec VPN. I can ping between both systems. The VPN allows traffic between the interface IP's of the CME systems only.
    London CME (local SCCP phones 22xx):
    interface FastEthernet0/0.100
    encapsulation dot1Q 100 native
    ip address 10.0.10.250 255.255.255.0
    voice class codec 101
    codec preference 1 g729r8
    codec preference 2 g711ulaw
    codec preference 3 g711alaw
    dial-peer voice 25 voip
    description *** SIP Peer to India ***
    answer-address 400.
    destination-pattern 400.
    voice-class codec 101
    session protocol sipv2
    session target ipv4:192.168.15.10
    incoming called-number 400.
    no vad
    India CME (Local SSCP phones 400x):
    interface FastEthernet0/0
    ip address 192.168.15.10 255.255.255.0
    voice class codec 100
    codec preference 1 g729r8
    codec preference 2 g711ulaw
    codec preference 3 g711alaw
    dial-peer voice 10 voip
    description *** SIP Peer to London UK ***
    answer-address 22..
    destination-pattern 22..
    voice-class codec 100
    session protocol sipv2
    session target ipv4:10.0.10.250
    incoming called-number 22..
    no vad
    The CME system at India also has an existing SIP dial peer to a service provider and has sip-ua configured (username, password, realm and registrar).
    A call from India (4005) to London (DN2207) fails, the ccsip debug attached. I'm assuming its because the sip-ua configuration is being used for these calls to when I don't want it to be. The from field shows “From: <sip:[email protected]” when I need this to be the internal IP 192.168.15.10.
    Can anyone offer any assistance with this?
    Regards,
    Chris

    Hi,
    thanks for your input however thats not the problem. 201.196.128.56 isn't an address on the router, it only has one IP and its 192.168.15.10.
    The 201.196.128.56 address is the NAT'd address on the firewall. So that when a SIP call is made to the internet with sip-ua the from address is the public IP.
    Chris

  • 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)|*^*^*

  • 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 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 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!

  • 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
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    Please rate all useful posts
    "opportunity is a haughty goddess who waste no time with those who are unprepared"

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