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..

Similar Messages

  • 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.

  • X-lite Sip Phone Registration Rejected

    I have an x-lite sip phone that i added as a third party(basic) sip device on cisco call manager, settings are:
    UserID: <Extn No on CUCM>
    Domain:<ip address of the CUCM>
    Password:<CUCM Digest Credentials>
    Authorization Name: <UserID on CUCM>
    No Domain Proxy
    What could be missing?

    Take a look at Mr. Bell's writeup on this:
    http://www.netcraftsmen.net/component/content/article/70-unified-communications/766-sip-endpoints-in-cisco-cucm-x-lite-as-an-example.html
    Please remember to rate helpful responses and identify helpful or correct answers.

  • How do I get facetime on a MacBook Pro to work?  I keep getting a " server encountered problem during registration.." message.  I have FaceTime ver. 1.1.1

    How do I get facetime on a MacBook Pro to work?  I keep getting a " server encountered problem during registration.." message.  I have FaceTime ver. 1.1.1
    Thanks

    Icapper wrote:
    I will end up getting something other than Logitech speakers, since I'm just weird like that.
    Your not weird, your a audiophile.
    https://en.wikipedia.org/wiki/Audiophile
    Good sound costs money. And with a 5.1 system your usually doing surround sound decoding for BlueRay movies etc. for home theater purposes.
    The PC 5.1 surround sound systems require a audio card in a PC tower and mainly used for playing 3D games so that won't work for any Mac at all. So don't buy a PC 5.1 surround sound system for your Mac.
    Harmon Kardon has the GoPlay, it's a portable stereo with awesome sound (not as good as their theater systems) and you can hook up a analog male/male stereo mini cable to it from the Mac.
    $200 and it has a iPod dock and also takes like 8 batteries so it's portable.
    http://www.amazon.com/dp/B002GHBTNC
    There is also the Bose Wave clock/radio, you will need a stereo mini to RCA break out cable for that.
    http://www.bose.com/controller?url=/shop_online/wave_systems/index.jsp
    The GoPlay has much better sound than the Bose, I think the Bose are overpriced.

  • 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"

  • Incoming calls issue in Third Party SIP Phone

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

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

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

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

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

  • Paging Third Party SIP Phones connected to CUCM

    Current SetUp: CUCM - Cisco 3925, Two MCS 7816 (Call Control Server) and One MCS 7825 (Voice mail server)
    We have third party SIP phones configured in auto answer mode. These phones are used to make live announcements.
    To Do:
    There are approximately 80 phones in the system and the requirement is to select any combination of these phones to make Public announcement (or Paging).
    Is there an application that enables us to select any combination of phones on the fly to do paging? How can we select a mp3 file to play on a phone in an auto answer mode?
    Any help will be appreciated.
    Thanks
    Sid

    There's nothing built into call manager to do this.  You could investigate using the Cisco Unified Application Environment (CUAE) and write a script to do this, or there are some 3rd party applications that might work for you such as Berbee's Informacast.

  • 3rd Party SIP phone to CUCM via SIP Proxy

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

    This is most likely because of the following...
    Because third-party SIP phones do not send a MAC address, they must identify themselves by using digest authentication.
    The REGISTER message includes the following header:
    Authorization: Digest username="xxxxxxxxxx",realm="ccmsipline",nonce="GBauADss2qoWr6k9y3hGGVDAqnLfoLk5",uri="sip:172.18.197.224",algorithm=MD5,response="126c0643a4923359ab59d4f53494552e"
    The username, xxxxxxxxxxx, must match an end user that is configured in the End User Configuration window of Cisco Unified CallManager Administration. The administrator configures the SIP third-party phone with the user; for example, swhite, in the Digest User field of Phone Configuration window.
    See the following document.
    http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/5_1_3/ccmcfg/b09sip3p.html
    Also Try this bug CSCef88775

  • Add third party SIP Phone to CCM 5

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

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

  • Configure SPA2102 as SIP Phone to SIP Proxy Server

    I have a SIP2102 which needs to be configured as SIP phone to a SIP Proxy Server.  All calls will stay within the local network.  I need to point SPA2102 to my SIP Proxy Server and assign an extension such that it is recognized by the SIP Proxy Server as part of its pool of valid extensions.  The documentation is not clear when just trying to set these parameters.

    if what you are trying to accomplish is simply register the SPA2102 with your Proxy server then the only thing you need to do is configure a USER ID (this is the extension number you want the SPA to have) and a Proxy (IP address of your SIP server) -- outbound proxy is only needed if your server requires the device to have this..
    Both of these parameters can be found under Line 1 tab and these settings should come from your SIP server-- Internet port of the device must be connected to your network
    SIP Port is another to consider in case your SIP server is using another port other than 6060..
    | isolate! isolate! isolate! |

  • A question about call manager traces for Sip phones.

    So today I create a sip based ip communicator and pressed the new call button and heard a dial tone.  I started typing my telephone number. Half way through, I heard  another secondary dial tone (which indicates mis-configured route pattern somewhere) . 
    However, When I look at the call manager logs, I do not actually see the digits that I was typing. With SCCP, I can see the keypad button press messages in the traces, but here, I cannot see the pressed buttons in my CUCM traces. Can anyone help with telling me how I can see button presses going to call manager .   All I can see are the logs  below which came up as soon as I got the dial tone and the final sip invite messages. I see nothing in-between. 
    |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.xx.4.xx on port 56714 index 31809 with 973 bytes:
    [6387070,NET]
    NOTIFY sip:[email protected] SIP/2.0
    Via: SIP/2.0/TCP 10.x.x.66:56714;branch=z9hG4bK00005b1e
    To: <sip:[email protected]>
    From: <sip:[email protected]>;tag=00ffb00bc50a00340000499f-00006ab4
    Call-ID: [email protected]
    Date: Sat, 14 Feb 2015 14:17:40 GMT
    CSeq: 19 NOTIFY
    Event: dialog
    Subscription-State: active
    Max-Forwards: 70
    Contact: <sip:[email protected]:56714;transport=TCP>
    Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
    Content-Length: 350
    Content-Type: application/dialog-info+xml
    Content-Disposition: session;handling=required
    <?xml version="1.0" encoding="UTF-8" ?>
    <dialog-info xmlns:call="urn:x-cisco:parmams:xml:ns:dialog-info:dialog:callinfo-dialog" version="18" state="partial" entity="sip:[email protected]">
    <dialog id="12" call-id="[email protected]" local-tag="00ffb00bc50a003300006390-00002d4f"><state>trying</state></dialog>
    </dialog-info>
    SIPStationD(12991) - processCommonDialogNotifyInd:   Did 12 Sending Notified SIPOffHook to new Cdfc

    Here is a more detailed explanation of how SIP calls notify cucm when they go off hook to make a call. The digit dialled here is 4080
    +++++ Analysis of SIP Phone making a call +++++++++
    The user picks up the phone and the IP Phone sends a NOTIFY to CUCM to indicate the start of a new dialog. This dialog begings by an offhook event
    00869539.002 |14:58:13.837 |AppInfo  |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.50.16.1 on port 52910 index 2748 with 976 bytes:
    [46240,NET]
    NOTIFY sip:[email protected] SIP/2.0
    Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK00002531
    To: <sip:[email protected]>
    From: <sip:[email protected]>;tag=544e42f26d0b001e000056e7-0000311c
    Call-ID: [email protected]
    Date: Mon, 16 Feb 2015 12:58:13 GMT
    CSeq: 11 NOTIFY
    Event: dialog
    Subscription-State: active
    Max-Forwards: 70
    Contact: <sip:[email protected]:52910;transport=TCP>
    Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
    Content-Length: 350
    Content-Type: application/dialog-info+xml
    Content-Disposition: session;handling=required
    <?xml version="1.0" encoding="UTF-8" ?>
    <dialog-info xmlns:call="urn:x-cisco:parmams:xml:ns:dialog-info:dialog:callinfo-dialog" version="10" state="partial" entity="sip:[email protected]">
    <dialog id="6" call-id="[email protected]" local-tag="544e42f26d0b001d00007cc9-000044a3"><state>trying</state></dialog>
    </dialog-info>
    ++++ CUCM SIP stack processes the new connection for the phone+++++++
    00869540.001 |14:58:13.837 |AppInfo  |//SIP/Stack/Info/0x0/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 2 (SIP_NETWORK_MSG), for event 1 (SIPSPI_EV_NEW_MESSAGE)
    00869540.002 |14:58:13.837 |AppInfo  |//SIP/Stack/Transport/0x0/sipTransportProcessNWNewConnMsg: context=(nil)
    00869540.003 |14:58:13.837 |AppInfo  |//SIP/Stack/Transport/0x0/sipConnectionManagerProcessNewConnMsg: gConnTab=0xe81c0d70, addr=10.50.16.1, port=52910, connid=2748, transport=TCP
    ++++ Next CUCM allocates a call id for this call +++++
    00869546.002 |14:58:13.838 |AppInfo  |LineControl(66) - Get call instance=1 for CI=24419584
    +++Next CUCM sends a 200 OK to the NOTIFY request for the new dialog ++++
    00869555.007 |14:58:13.839 |AppInfo  |//SIP/Stack/Transport/0x0xe7df4d48/sipTransportPostSendMessage: Posting send for msg=0xefbe9910, addr=10.50.16.1, port=52910, connId=2748 for
    00869555.008 |14:58:13.839 |AppInfo  |//SIP/Stack/Info/0x0/act_dialog_pending_resp_event: Changing from State: SUBSCRIBE_STATE_DIALOG_PENDING to state SUBSCRIBE_STATE_ACTIVE
    00869556.000 |14:58:13.839 |SdlSig   |SIPSPISignal                           |wait                           |SIPTcp(1,100,71,1)               |SIPHandler(1,100,79,1)           |1,100,14,31314.75^10.50.16.1^SEP00909E9D106C |*TraceFlagOverrode
    00869556.001 |14:58:13.839 |AppInfo  |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.50.16.1 on port 52910 index 2748
    [46241,NET]
    SIP/2.0 200 OK
    Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK00002531
    From: <sip:[email protected]>;tag=544e42f26d0b001e000056e7-0000311c
    To: <sip:[email protected]>;tag=1822746380
    Date: Mon, 16 Feb 2015 12:58:13 GMT
    Call-ID: [email protected]
    CSeq: 11 NOTIFY
    Server: Cisco-CUCM10.5
    Content-Length: 0
    ++++ The IP Phone sends its connection ID to CUCM, its ip address and its port number+++++++++
    00869541.001 |14:58:13.838 |AppInfo  |SIPStationInit: connID=2748, SEP00909E9D106C, 10.50.16.1:52910, Routed signal by connection index to (1,100,73,66)
    ++++ Next CUCM informs us that the NOTIFY message is for an offhook event ++++++
    00869542.003 |14:58:13.838 |AppInfo  |SIPStationD(66) - processCommonDialogNotifyInd: Notified Dialogs - Did 6 State trying
    00869542.004 |14:58:13.838 |AppInfo  |SIPStationD(66) - processCommonDialogNotifyInd:   Did 6 Sending Notified SIPOffHook to new Cdfc
    00869542.010 |14:58:13.838 |AppInfo  |SIPStationD(66) - processSIPOffHook Primary Call Not-Found
    00869543.000 |14:58:13.838 |SdlSig   |SIPOffHookInd 
    +++ The next thing is the USER dials a digit on the phone ++++++
    This is where it gets a little complicated. So lets examine this. The first digit that is dialled generates an INVITE to CUCM like this:
    In this example the user dialled "4" first so we see an "INVITE sip:4@host-IP"
    00869559.002 |14:58:14.064 |AppInfo  |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.50.16.1 on port 52910 index 2748 with 1445 bytes:
    [46242,NET]
    INVITE sip:[email protected];user=phone SIP/2.0
    Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK000015ec
    From: "Emre ESEN" <sip:[email protected]>;tag=544e42f26d0b001d00007cc9-000044a3
    To: <sip:[email protected];user=phone>
    Call-ID: [email protected]
    Max-Forwards: 70
    Date: Mon, 16 Feb 2015 12:58:14 GMT
    CSeq: 101 INVITE
    User-Agent: Cisco-SIPIPCommunicator/9.1.1
    Contact: <sip:[email protected]:52910;transport=tcp>
    Expires: 180
    Accept: application/sdp
    Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
    Remote-Party-ID: "Emre ESEN" <sip:[email protected]>;party=calling;id-type=subscriber;privacy=off;screen=yes
    Supported: replaces,join,sdp-anat,norefersub,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-5.1.0,X-cisco-xsi-8.5.1
    Allow-Events: kpml,dialog
    Content-Length: 373
    Content-Type: application/sdp
    Content-Disposition: session;handling=optional
    v=0
    o=Cisco-SIPUA 21020 0 IN IP4 10.50.16.1
    s=SIP Call
    t=0 0
    m=audio 20250 RTP/AVP 0 8 18 9 116 124 101
    c=IN IP4 10.50.16.1
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:9 G722/8000
    a=rtpmap:116 iLBC/8000
    a=fmtp:116 mode=20
    a=rtpmap:124 ISAC/16000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=sendrecv
    +++++ NEXT CUCM sends a trying for the INVITE it received +++++++++++
    00869562.001 |14:58:14.065 |AppInfo  |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.50.16.1 on port 52910 index 2748
    [46243,NET]
    SIP/2.0 100 Trying
    Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK000015ec
    From: "Emre ESEN" <sip:[email protected]>;tag=544e42f26d0b001d00007cc9-000044a3
    To: <sip:[email protected];user=phone>
    Date: Mon, 16 Feb 2015 12:58:14 GMT
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Allow-Events: presence
    Content-Length: 0
    ++++NOW CUCM evaluates the DTMF supported by the phone to determine how to inform the phones to send the remaining dtmf digits++++
    From the INVITE cucm concludes that KPML and rtp-nte is supported
    00869566.009 |14:58:14.066 |AppInfo  |setEndpointsDtmfCaps: KPML Supported.
    00869566.010 |14:58:14.066 |AppInfo  |setEndpointsDtmfCaps: Detected inband DTMF support
    Next CUCM generates kpml event pkg which is going to be used to receive the remaining digits from the phone
    00869590.001 |14:58:14.067 |AppInfo  |SIPEventPkg::SIPEventPkg 0xe4a1d1e0 scbId[16725], event name[kpml; [email protected]; from-tag=544e42f26d0b001d00007cc9-000044a3], id[]
    +++ Next CUCM sends a SUBSCRIBE to the IP phone for kpml event +++++
    00869594.001 |14:58:14.068 |AppInfo  |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.50.16.1 on port 52910 index 2748
    [46244,NET]
    SUBSCRIBE sip:[email protected]:52910 SIP/2.0
    Via: SIP/2.0/TCP 10.28.132.111:5060;branch=z9hG4bKce719b37856
    From: <sip:[email protected]>;tag=480227084
    To: <sip:[email protected]>
    Call-ID: [email protected]
    CSeq: 101 SUBSCRIBE
    Date: Mon, 16 Feb 2015 12:58:14 GMT
    User-Agent: Cisco-CUCM10.5
    Event: kpml; [email protected]; from-tag=544e42f26d0b001d00007cc9-000044a3
    Expires: 7200
    Contact: <sip:[email protected]:5060;transport=tcp>
    Accept: application/kpml-response+xml
    Max-Forwards: 70
    Content-Type: application/kpml-request+xml
    Content-Length: 424
    <?xml version="1.0" encoding="UTF-8" ?>
    <kpml-request xmlns="urn:ietf:params:xml:ns:kpml-request" xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance" xsi:schemaLocation="urn:ietf:params:xml:ns:kpml-request kpml-request.xsd" version="1.0">
      <pattern criticaldigittimer="1000" extradigittimer="500" interdigittimer="15000" persist="persist">
        <regex tag="Backspace OK">[x#*+]|bs</regex>
      </pattern>
      </kpml-request>
     +++ Next we get a 200 OK to the SUBSCRIBE from the ip phone ++++
     00869595.002 |14:58:14.118 |AppInfo  |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.50.16.1 on port 52910 index 2748 with 459 bytes:
    [46245,NET]
    SIP/2.0 200 OK
    Via: SIP/2.0/TCP 10.28.132.111:5060;branch=z9hG4bKce719b37856
    From: <sip:[email protected]>;tag=480227084
    To: <sip:[email protected]>;tag=544e42f26d0b001f0000092c-0000070a
    Call-ID: [email protected]
    Date: Mon, 16 Feb 2015 12:58:14 GMT
    CSeq: 101 SUBSCRIBE
    Server: Cisco-SIPIPCommunicator/9.1.1
    Contact: <sip:[email protected]:52910;transport=TCP>
    Expires: 7200
    Content-Length: 0
    +++ NEXT the IP phones sends the remaining digit dialled on the phone to CUCM +++
    00869603.002 |14:58:14.183 |AppInfo  |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.50.16.1 on port 52910 index 2748 with 573 bytes:
    [46247,NET]
    NOTIFY sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK000045c8
    To: <sip:[email protected]>;tag=480227084
    From: <sip:[email protected]>;tag=544e42f26d0b001f0000092c-0000070a
    Call-ID: [email protected]
    Date: Mon, 16 Feb 2015 12:58:14 GMT
    CSeq: 1000 NOTIFY
    Event: kpml
    Subscription-State: active; expires=7200
    Max-Forwards: 70
    Contact: <sip:[email protected]:52910;transport=TCP>
    Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
    Content-Length: 0
    00869608.001 |14:58:14.183 |AppInfo  |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.50.16.1 on port 52910 index 2748
    [46248,NET]
    SIP/2.0 200 OK
    Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK000045c8
    From: <sip:[email protected]>;tag=544e42f26d0b001f0000092c-0000070a
    To: <sip:[email protected]>;tag=480227084
    Date: Mon, 16 Feb 2015 12:58:14 GMT
    Call-ID: [email protected]
    CSeq: 1000 NOTIFY
    Server: Cisco-CUCM10.5
    Content-Length: 0
    +++Next the IP phone sends the next digit. Here its important to note that the NOTIFY doesnt contain the next digit,
    the NOTIFY is still the same as the first digit but the next digit is carried in the xml document attached to the NOTIFY.
    At this point I will insert a paragraph from the RFC 4730 for SIP KPML
    +++++++++++++
    The event package uses SUBSCRIBE
       messages and allows for XML documents that define and describe filter
       specifications for capturing key presses (DTMF Tones) entered at a
       presentation-free User Interface SIP User Agent (UA).  The event
       package uses NOTIFY messages and allows for XML documents to report
       the captured key presses (DTMF tones), consistent with the filter
       specifications, to an Application Server +++++++++++++++++++++++++++
    00869609.002 |14:58:14.209 |AppInfo  |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.50.16.1 on port 52910 index 2748 with 877 bytes:
    [46249,NET]
    NOTIFY sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK00003c9d
    To: <sip:[email protected]>;tag=480227084
    From: <sip:[email protected]>;tag=544e42f26d0b001f0000092c-0000070a
    Call-ID: [email protected]
    Date: Mon, 16 Feb 2015 12:58:14 GMT
    CSeq: 1001 NOTIFY
    Event: kpml
    Subscription-State: active; expires=7200
    Max-Forwards: 70
    Contact: <sip:[email protected]:52910;transport=TCP>
    Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
    Content-Length: 209
    Content-Type: application/kpml-response+xml
    Content-Disposition: session;handling=required
    <?xml version="1.0" encoding="UTF-8"?>
    <kpml-response xmlns="urn:ietf:params:xml:ns:kpml-response" version="1.0" code="200" text="OK" suppressed="false" forced_flush="false" digits="0" tag="Backspace OK"/>
    00869622.001 |14:58:14.210 |AppInfo  |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.50.16.1 on port 52910 index 2748
    [46250,NET]
    SIP/2.0 200 OK
    Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK00003c9d
    From: <sip:[email protected]>;tag=544e42f26d0b001f0000092c-0000070a
    To: <sip:[email protected]>;tag=480227084
    Date: Mon, 16 Feb 2015 12:58:14 GMT
    Call-ID: [email protected]
    CSeq: 1001 NOTIFY
    Server: Cisco-CUCM10.5
    Content-Length: 0
    +++ Again we get the next digit ++++
    00869624.002 |14:58:14.262 |AppInfo  |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.50.16.1 on port 52910 index 2748 with 877 bytes:
    [46251,NET]
    NOTIFY sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK0000310f
    To: <sip:[email protected]>;tag=480227084
    From: <sip:[email protected]>;tag=544e42f26d0b001f0000092c-0000070a
    Call-ID: [email protected]
    Date: Mon, 16 Feb 2015 12:58:14 GMT
    CSeq: 1002 NOTIFY
    Event: kpml
    Subscription-State: active; expires=7200
    Max-Forwards: 70
    Contact: <sip:[email protected]:52910;transport=TCP>
    Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
    Content-Length: 209
    Content-Type: application/kpml-response+xml
    Content-Disposition: session;handling=required
    <?xml version="1.0" encoding="UTF-8"?>
    <kpml-response xmlns="urn:ietf:params:xml:ns:kpml-response" version="1.0" code="200" text="OK" suppressed="false" forced_flush="false" digits="8" tag="Backspace OK"/>
    00869637.001 |14:58:14.263 |AppInfo  |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.50.16.1 on port 52910 index 2748
    [46252,NET]
    SIP/2.0 200 OK
    Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK0000310f
    From: <sip:[email protected]>;tag=544e42f26d0b001f0000092c-0000070a
    To: <sip:[email protected]>;tag=480227084
    Date: Mon, 16 Feb 2015 12:58:14 GMT
    Call-ID: [email protected]
    CSeq: 1002 NOTIFY
    Server: Cisco-CUCM10.5
    Content-Length: 0
    +++ Finally we get the last digit ++++
    00869638.002 |14:58:14.390 |AppInfo  |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.50.16.1 on port 52910 index 2748 with 877 bytes:
    [46253,NET]
    NOTIFY sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK00006c1c
    To: <sip:[email protected]>;tag=480227084
    From: <sip:[email protected]>;tag=544e42f26d0b001f0000092c-0000070a
    Call-ID: [email protected]
    Date: Mon, 16 Feb 2015 12:58:14 GMT
    CSeq: 1003 NOTIFY
    Event: kpml
    Subscription-State: active; expires=7200
    Max-Forwards: 70
    Contact: <sip:[email protected]:52910;transport=TCP>
    Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
    Content-Length: 209
    Content-Type: application/kpml-response+xml
    Content-Disposition: session;handling=required
    <?xml version="1.0" encoding="UTF-8"?>
    <kpml-response xmlns="urn:ietf:params:xml:ns:kpml-response" version="1.0" code="200" text="OK" suppressed="false" forced_flush="false" digits="0" tag="Backspace OK"/>
    Once digit collection is completed CUCM proceeds to finalise its digit analysis process.
    Note that digit analysis is carried out for each digit that is recieved. I have only included the final DA here
    00869648.003 |14:58:14.391 |AppInfo  |Digit Analysis: star_DaReq: Matching SIP URL, Numeric User, user=4080
    00869648.004 |14:58:14.391 |AppInfo  |Digit Analysis: getDaRes data: daRes.ssType=[0] Intercept DAMR.sstype=[0], TPcount=[0], DAMR.NotifyCount=[0], DaRes.NotifyCount=[0]
    00869648.005 |14:58:14.391 |AppInfo  |Digit Analysis: getDaRes - Remote Destination [4080] isURI[0]
    00869648.012 |14:58:14.391 |AppInfo  |Digit analysis: match(pi="2", fqcn="9106", cn="9106",plv="5", pss="", TodFilteredPss="", dd="4080",dac="0")
    00869648.013 |14:58:14.391 |AppInfo  |Digit analysis: analysis results
    00869648.014 |14:58:14.391 |AppInfo  ||PretransformCallingPartyNumber=9106
    |CallingPartyNumber=9106
    |DialingPartition=
    |DialingPattern=4XXX
    |FullyQualifiedCalledPartyNumber=4080
    |DialingPatternRegularExpression=(4[0-9][0-9][0-9])
    |DialingWhere=
    +++++Once this is done CUCM then proceeds to send the call out to to the intended destination as configured in the RL ++++
    00869701.001 |14:58:14.435 |AppInfo  |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.250.0.13 on port 5060 index 2754
    [46256,NET]
    INVITE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/TCP 10.28.132.111:5060;branch=z9hG4bKce931ee3d74
    From: "Emre ESEN" <sip:[email protected]>;tag=16726~813ee89e-33db-4d58-9f6a-61542cc840ee-24419585
    To: <sip:[email protected]>
    Date: Mon, 16 Feb 2015 12:58:14 GMT
    Call-ID: [email protected]
    Supported: timer,resource-priority,replaces

Maybe you are looking for

  • How to get newly added recipient on a reply email vs the old recipients already on the chain

    Hi, Really need some help as hitting a dead end, hopefully someone here can help as always :-) I'm trying to differentiate between old and new recipients on an email chain when a user replies / forwards an email.  e.g. if a user receives an email wit

  • GetRows(), GetXML() VS. Oracle Objects...

    Hi! I've been working with OO4O for 2-3 months now. Since the Dynaset is RIDICULOUSLY slow, I always use GetRows() to get my data. But when I call GetRows() in a Dynaset containing objects (i.e. OraObject), all the objects fields are set to Nothing (

  • Links Disappearing...

    Hi All - In both Google Chrome (for Mac), Firefox, and Safari, my hyperlinks are disappearing. When they will occasionally show up, they disappear on rollover. Can anyone help? Thanks!

  • Bought PE12 thru Adobe.   The exe is not installed  ??

    I have adpbe Photoshop Elements 12.   When installing my camera card the drop down 'menu'  i,e Open With does not list elements 12 is this done thry 'exe'   ?  can someone help

  • Books on iPad in Canada

    Where is the content in the iBook store in Canada? I only see a bunch of old public domain stuff.