Trust List 8.5 BE to 9.1

Dears
Currently we are running in 8.5 BE. i installed fresh 9.1 and exported all phones from 8.5 to 9.1 successfully. after rebooting the phones they are registering successfully to 9.1 but they are pulling the new load file of 9.1 they are still on old load file of 8.5 BE When i delete the trust list file from phone and when they reset they are successfully upgrading to new load of 9.1.
But it is not possible for me to go manually and delete trust list file for 400 phones, so how i can achieve the task in bulk.

The document Manish has referenced dicusses all the options.
If you are keeping the IP address same for new cluster and both clusters can not be online at same time, use Rollback option. As said before this will only work if its done before migration is attempted. Please do the following in same order (refer the document in the below link for details) .
1) From the CUCM Enterprise params> Prepare Cluster for Rollback to pre-8.0 enterprise parameter to True
2) Restart the TVS service and then TFTP service
3) Reset phones: upon boot they will get an emplty ITL file. Your cluster will be ready for migration.
Reference:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/security/8_5_1/secugd/secusbd.html#wp1092162
Another thing I would recommend is doing this just before migration. Because once you set this rollback parameter to true all your phone services using https will stop working. Call processing etc. will not be affected. If you have to do this weekend before or way before the actual migration, the workaround is to change the secure URLs from https to http in enterprise parameters.
-Terry

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    <redirectedNumber>true</redirectedNumber>
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    Hello,
    I'm facing exactly the same problem, that is:
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    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#

  • CUCM 8.6.2 Phones not registering 7841 and 8831

    Hi all
    I am trying to register some new phones - 7841 and 8831.  We installed the COP and Device Pack and rebooted the cluster.  Firmware defaults in the server are sip78xx.10-1-1SR2-1 and sip8831.9-3-3-5. 
    These are the first phones of these models that we are trying to use.  All other devices are up and working.  I added the phones manually, auto-reg is disabled.  These new phones are both displaying the same behavior - they booted, but neither device will register.  They are both running the the same firmware version configured as the default in the server.
    I have looked at some other discussions but haven't seen any solutions that match up with our issue.   Your input and ideas appreciated, thanks

    I'm having this same problem. Over the weekend, I installed the latest device packs and restarted both servers in the cluster. I'm running CUCM 8.6.2 and the phones are 8831.
    What's really strange is that the phone is getting incorrect info from the switch port. It's getting a DHCP address from the VLAN2 subnet, when it should be receiving an IP from the VLAN100 subnet.
    VLAN2 = Data
    VLAN100 = Voice
    The original port config:
    interface FastEthernet0/25
     description Workstations on VLAN2 - Phones on VLAN 100
     switchport access vlan 2
     switchport trunk encapsulation dot1q
     switchport trunk native vlan 2
     switchport mode trunk
     switchport voice vlan 100
     srr-queue bandwidth share 10 10 60 20
     srr-queue bandwidth shape 10 0 0 0
     mls qos trust device cisco-phone
     mls qos trust cos
     auto qos voip cisco-phone
     spanning-tree portfast
    end
    I even changed the switch port configuration to this:
    interface FastEthernet0/25
     description Workstations on VLAN2 - Phones on VLAN 100
     switchport access vlan 100
     switchport trunk encapsulation dot1q
     switchport trunk native vlan 100
     switchport mode trunk
     switchport voice vlan 100
     srr-queue bandwidth share 10 10 60 20
     srr-queue bandwidth shape 10 0 0 0
     mls qos trust device cisco-phone
     mls qos trust cos
     auto qos voip cisco-phone
     spanning-tree portfast
    end
    VLAN 2 isn't even in the configuration, but the phone (even after a RESET) is still getting an IP from the DHCP server on VLAN2.
    Very strange behavior.
    Additional info:
    The phone status menu is showing a continuous message of:
    TFTP Timeout : SEPXXXXXXXXXXXX.cnf.xm.
    No Trust List Installed.
    Any suggestions / ideas would be appreciated.

  • SIP Trunk - No voice with Single Number Reach

    Hi Community.
    I setup SIP Trunk with the CCA. Everything is working Call In and Call Out. Call Forward and so on.
    But with Single Number reach is something wrong. The mobile phone is ringing and I can get the call, but I hear not any voice.
    Can someone please help me out? Below the config.
    version 15.1
    parser config cache interface
    no service pad
    service timestamps debug datetime msec
    service timestamps log datetime msec
    no service password-encryption
    service internal
    service compress-config
    service sequence-numbers
    dot11 ssid cisco-data
     vlan 1
     authentication open
    dot11 ssid cisco-voice
     vlan 100
     authentication open
    ip source-route
    ip cef
    ip dhcp relay information trust-all
    ip dhcp excluded-address 10.1.1.1 10.1.1.9
    ip dhcp excluded-address 10.1.1.241 10.1.1.255
    ip dhcp pool phone
     network 10.1.1.0 255.255.255.0
     default-router 10.1.1.1
     option 150 ip 10.1.1.1
    ip domain name site1.365873.trk.ipvoip.ch
    ip name-server 8.8.8.8
    ip inspect WAAS flush-timeout 10
    ip inspect name SDM_LOW dns
    ip inspect name SDM_LOW ftp
    ip inspect name SDM_LOW h323
    ip inspect name SDM_LOW https
    ip inspect name SDM_LOW icmp
    ip inspect name SDM_LOW imap
    ip inspect name SDM_LOW pop3
    ip inspect name SDM_LOW netshow
    ip inspect name SDM_LOW rcmd
    ip inspect name SDM_LOW realaudio
    ip inspect name SDM_LOW rtsp
    ip inspect name SDM_LOW esmtp
    ip inspect name SDM_LOW sqlnet
    ip inspect name SDM_LOW streamworks
    ip inspect name SDM_LOW tftp
    ip inspect name SDM_LOW tcp router-traffic
    ip inspect name SDM_LOW udp router-traffic
    ip inspect name SDM_LOW vdolive
    no ipv6 cef
    multilink bundle-name authenticated
    stcapp ccm-group 1
    stcapp
    isdn switch-type basic-net3
    voice call send-alert
    voice rtp send-recv
    voice service voip
     ip address trusted list
      ipv4 0.0.0.0 0.0.0.0
     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 sip refer
     fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw
     sip
      registrar server expires max 3600 min 3600
      localhost dns:site1.365873.trk.ipvoip.ch
      no update-callerid
    voice class codec 1
     codec preference 1 g711alaw
    voice register global
     mode cme
     source-address 10.1.1.1 port 5060
     load 9971 sip9971.9-2-2
     load 9951 sip9951.9-2-2
     load 8961 sip8961.9-2-2
     timezone 23
    voice source-group CCA_SIP_SOURCE_GROUP_CUE_CME
     access-list 2
     translation-profile incoming SIP_Incoming
    voice source-group CCA_SIP_SOURCE_GROUP_EXTERNAL
     access-list 3
    voice translation-rule 9
     rule 1 /0041449475090/ /90/
     rule 2 /0041449475091/ /91/
     rule 3 /0041449475092/ /92/
     rule 4 /0041449475093/ /93/
     rule 5 /0041449475094/ /94/
     rule 6 /0041449475095/ /95/
     rule 7 /0041449475096/ /96/
     rule 8 /0041449475097/ /97/
     rule 9 /0041449475098/ /98/
     rule 10 /0041449475099/ /99/
    voice translation-rule 410
     rule 1 /^0\(.*\)/ /\1/
     rule 15 /^..$/ /0041449475090/
    voice translation-rule 411
     rule 1 /^0\(.*\)/ /ABCD0\1/
    voice translation-rule 412
     rule 1 /^ABCD\(.*\)/ /\1/
    voice translation-rule 422
     rule 15 /^ABCD\(.*\)/ /\1/
    voice translation-rule 1000
     rule 1 /.*/ //
    voice translation-rule 1111
     rule 1 /^9\([1-9]\)$/ /004144947509\1/
     rule 15 /^..$/ /0041449475090/
    voice translation-rule 1112
     rule 1 /^0/ //
    voice translation-rule 2000
     rule 1 /0041449475098/ /98/
    voice translation-rule 2001
     rule 1 /0041449475097/ /97/
    voice translation-rule 2002
     rule 1 /^6/ //
    voice translation-rule 2222
    voice translation-profile AA_Profile
     translate called 2001
    voice translation-profile CALLER_ID_TRANSLATION_PROFILE
     translate calling 1111
    voice translation-profile CallBlocking
     translate called 2222
    voice translation-profile OUTGOING_TRANSLATION_PROFILE
     translate called 1112
    voice translation-profile PSTN_CallForwarding
     translate redirect-target 410
     translate redirect-called 410
    voice translation-profile PSTN_Outgoing
     translate calling 1111
     translate called 1112
     translate redirect-target 410
     translate redirect-called 410
    voice translation-profile SIP_Called_9
     translate calling 3265
     translate called 9
    voice translation-profile SIP_Incoming
     translate called 411
    voice translation-profile SIP_Passthrough
     translate called 412
    voice translation-profile SIP_Passthrough_CallBlocking
     translate called 422
    voice translation-profile VM_Profile
     translate called 2000
    voice translation-profile XFER_TO_VM_PROFILE
     translate redirect-called 2002
    voice translation-profile nondialable
     translate called 1000
    voice-card 0
     dspfarm
     dsp services dspfarm
    fax interface-type fax-mail
    license udi pid UC540W-BRI-K9 sn FGL163220SL
    archive
     log config
      logging enable
      logging size 600
      hidekeys
    username admin privilege 15 secret xxx
    username xxx password 0 ""
    username xxx password 0 ""
    ip tftp source-interface Loopback0
    bridge irb
    interface Loopback0
     description $FW_INSIDE$
     ip address 10.1.10.2 255.255.255.252
     ip access-group 101 in
     ip nat inside
     ip virtual-reassembly in
    interface FastEthernet0/0
     description $FW_OUTSIDE$
     no ip address
     ip inspect SDM_LOW out
     ip virtual-reassembly in
     ip verify unicast reverse-path
     load-interval 30
     shutdown
     duplex auto
     speed auto
    interface Integrated-Service-Engine0/0
     description cue is initialized with default IMAP group
     ip unnumbered Loopback0
     ip nat inside
     ip virtual-reassembly in
     service-module ip address 10.1.10.1 255.255.255.252
     service-module ip default-gateway 10.1.10.2
    interface FastEthernet0/1/0
     no ip address
     macro description cisco-desktop
     spanning-tree portfast
    interface FastEthernet0/1/1
     switchport voice vlan 100
     no ip address
     macro description cisco-phone
     spanning-tree portfast
    interface FastEthernet0/1/2
     switchport voice vlan 100
     no ip address
     macro description cisco-phone
     spanning-tree portfast
    interface FastEthernet0/1/3
     switchport voice vlan 100
     no ip address
     macro description cisco-phone
     spanning-tree portfast
    interface FastEthernet0/1/4
     switchport voice vlan 100
     no ip address
     macro description cisco-phone
     spanning-tree portfast
    interface FastEthernet0/1/5
     switchport voice vlan 100
     no ip address
     macro description cisco-phone
     spanning-tree portfast
    interface FastEthernet0/1/6
     switchport voice vlan 100
     no ip address
     macro description cisco-phone
     spanning-tree portfast
    interface FastEthernet0/1/7
     switchport voice vlan 100
     no ip address
     macro description cisco-phone
     spanning-tree portfast
    interface FastEthernet0/1/8
     no ip address
     macro description cisco-desktop
     spanning-tree portfast
    interface BRI0/1/0
     no ip address
     isdn switch-type basic-net3
     isdn point-to-point-setup
     isdn incoming-voice voice
     isdn sending-complete
     isdn static-tei 0
    interface BRI0/1/1
     no ip address
     shutdown
     isdn switch-type basic-net3
     isdn point-to-point-setup
     isdn incoming-voice voice
     isdn sending-complete
     isdn static-tei 0
    interface Dot11Radio0/5/0
     no ip address
     ssid cisco-data
     ssid cisco-voice
     speed basic-1.0 basic-2.0 basic-5.5 6.0 9.0 basic-11.0 12.0 18.0 24.0 36.0 48.0 54.0
     station-role root
     antenna receive right
     antenna transmit right
    interface Dot11Radio0/5/0.1
     encapsulation dot1Q 1 native
     bridge-group 1
     bridge-group 1 subscriber-loop-control
     bridge-group 1 spanning-disabled
     bridge-group 1 block-unknown-source
     no bridge-group 1 source-learning
     no bridge-group 1 unicast-flooding
    interface Dot11Radio0/5/0.100
     encapsulation dot1Q 100
     bridge-group 100
     bridge-group 100 subscriber-loop-control
     bridge-group 100 spanning-disabled
     bridge-group 100 block-unknown-source
     no bridge-group 100 source-learning
     no bridge-group 100 unicast-flooding
    interface Vlan1
     no ip address
     bridge-group 1
     bridge-group 1 spanning-disabled
    interface Vlan100
     no ip address
     bridge-group 100
     bridge-group 100 spanning-disabled
    interface BVI1
     description $FW_INSIDE$
     ip address 192.168.10.2 255.255.255.0
     ip access-group 102 in
     ip nat inside
     ip virtual-reassembly in
    interface BVI100
     description $FW_INSIDE$
     ip address 10.1.1.1 255.255.255.0
     ip access-group 103 in
     ip nat inside
     ip virtual-reassembly in
    ip forward-protocol nd
    ip http server
    ip http authentication local
    ip http secure-server
    ip http path flash:/gui
    ip dns server
    ip nat inside source list 1 interface FastEthernet0/0 overload
    ip route 0.0.0.0 0.0.0.0 192.168.10.1
    ip route 10.1.10.1 255.255.255.255 Integrated-Service-Engine0/0
    access-list 1 remark SDM_ACL Category=2
    access-list 1 permit 10.1.1.0 0.0.0.255
    access-list 1 permit 192.168.10.0 0.0.0.255
    access-list 1 permit 10.1.10.0 0.0.0.3
    access-list 2 remark CCA_SIP_SOURCE_GROUP_ACL_INTERNAL
    access-list 2 remark SDM_ACL Category=1
    access-list 2 permit 192.168.10.2
    access-list 2 permit 10.1.10.0 0.0.0.3
    access-list 2 permit 192.168.10.0 0.0.0.255
    access-list 2 permit 10.1.1.0 0.0.0.255
    access-list 3 remark CCA_SIP_SOURCE_GROUP_ACL_EXTERNAL
    access-list 3 remark SDM_ACL Category=1
    access-list 3 permit 212.147.47.216
    access-list 3 deny   any
    access-list 100 remark auto generated by SDM firewall configuration
    access-list 100 remark SDM_ACL Category=1
    access-list 100 deny   ip 192.168.10.0 0.0.0.255 any
    access-list 100 deny   ip host 255.255.255.255 any
    access-list 100 deny   ip 127.0.0.0 0.255.255.255 any
    access-list 100 permit ip any any
    access-list 101 remark auto generated by SDM firewall configuration##NO_ACES_8##
    access-list 101 remark SDM_ACL Category=1
    access-list 101 permit tcp 10.1.1.0 0.0.0.255 eq 2000 any
    access-list 101 permit udp 10.1.1.0 0.0.0.255 eq 2000 any
    access-list 101 deny   ip 10.1.1.0 0.0.0.255 any
    access-list 101 deny   ip 192.168.10.0 0.0.0.255 any
    access-list 101 deny   ip 192.168.1.0 0.0.0.255 any
    access-list 101 deny   ip host 255.255.255.255 any
    access-list 101 deny   ip 127.0.0.0 0.255.255.255 any
    access-list 101 permit ip any any
    access-list 102 remark auto generated by SDM firewall configuration##NO_ACES_6##
    access-list 102 remark SDM_ACL Category=1
    access-list 102 deny   ip 10.1.10.0 0.0.0.3 any
    access-list 102 deny   ip 10.1.1.0 0.0.0.255 any
    access-list 102 deny   ip 192.168.1.0 0.0.0.255 any
    access-list 102 deny   ip host 255.255.255.255 any
    access-list 102 deny   ip 127.0.0.0 0.255.255.255 any
    access-list 102 permit ip any any
    access-list 103 remark auto generated by SDM firewall configuration##NO_ACES_8##
    access-list 103 remark SDM_ACL Category=1
    access-list 103 permit tcp 10.1.10.0 0.0.0.3 any eq 2000
    access-list 103 permit udp 10.1.10.0 0.0.0.3 any eq 2000
    access-list 103 deny   ip 10.1.10.0 0.0.0.3 any
    access-list 103 deny   ip 192.168.10.0 0.0.0.255 any
    access-list 103 deny   ip 192.168.1.0 0.0.0.255 any
    access-list 103 deny   ip host 255.255.255.255 any
    access-list 103 deny   ip 127.0.0.0 0.255.255.255 any
    access-list 103 permit ip any any
    access-list 104 remark auto generated by SDM firewall configuration##NO_ACES_14##
    access-list 104 remark SDM_ACL Category=1
    access-list 104 deny   ip 10.1.10.0 0.0.0.3 any
    access-list 104 deny   ip 10.1.1.0 0.0.0.255 any
    access-list 104 permit ip any any
    access-list 104 permit udp host 8.8.8.8 eq domain any
    access-list 104 permit icmp any any echo-reply
    access-list 104 permit icmp any any time-exceeded
    access-list 104 permit icmp any any unreachable
    access-list 104 deny   ip 10.0.0.0 0.255.255.255 any
    access-list 104 deny   ip 172.16.0.0 0.15.255.255 any
    access-list 104 deny   ip 192.168.0.0 0.0.255.255 any
    access-list 104 deny   ip 127.0.0.0 0.255.255.255 any
    access-list 104 deny   ip host 255.255.255.255 any
    access-list 104 deny   ip host 0.0.0.0 any
    access-list 104 deny   ip any any
    control-plane
    bridge 1 route ip
    bridge 100 route ip
    voice-port 0/0/0
     cptone CH
     station-id name FAX
     station-id number 99
     caller-id enable
    voice-port 0/0/1
     cptone CH
     shutdown
     caller-id enable
    voice-port 0/0/2
     cptone CH
     shutdown
     caller-id enable
    voice-port 0/0/3
     cptone CH
     shutdown
     caller-id enable
    voice-port 0/1/0
     compand-type a-law
     cptone CH
     bearer-cap Speech
    voice-port 0/1/1
     compand-type a-law
     cptone CH
     bearer-cap Speech
    voice-port 0/4/0
     auto-cut-through
     signal immediate
     input gain auto-control -15
     description Music On Hold Port
    sccp local Loopback0
    sccp ccm 10.1.1.1 identifier 1 version 4.0
    sccp
    sccp ccm group 1
     associate ccm 1 priority 1
     associate profile 2 register mtpa4934c6ee4e0
    dspfarm profile 2 transcode
     description CCA transcoding for SIP Trunk VTX
     codec g711ulaw
     codec g711alaw
     codec g729ar8
     codec g729abr8
     maximum sessions 10
     associate application SCCP
    dial-peer cor custom
     name internal
     name local
     name local-plus
     name international
     name national
     name national-plus
     name emergency
     name toll-free
    dial-peer cor list call-internal
     member internal
    dial-peer cor list call-local
     member local
    dial-peer cor list call-local-plus
     member local-plus
    dial-peer cor list call-national
     member national
    dial-peer cor list call-national-plus
     member national-plus
    dial-peer cor list call-international
     member international
    dial-peer cor list call-emergency
     member emergency
    dial-peer cor list call-toll-free
     member toll-free
    dial-peer cor list user-internal
     member internal
     member emergency
    dial-peer cor list user-local
     member internal
     member local
     member emergency
     member toll-free
    dial-peer cor list user-local-plus
     member internal
     member local
     member local-plus
     member emergency
     member toll-free
    dial-peer cor list user-national
     member internal
     member local
     member local-plus
     member national
     member emergency
     member toll-free
    dial-peer cor list user-national-plus
     member internal
     member local
     member local-plus
     member national
     member national-plus
     member emergency
     member toll-free
    dial-peer cor list user-international
     member internal
     member local
     member local-plus
     member international
     member national
     member national-plus
     member emergency
     member toll-free
    dial-peer voice 1 pots
     destination-pattern 99
     port 0/0/0
     no sip-register
    dial-peer voice 2 pots
     port 0/0/1
     no sip-register
    dial-peer voice 3 pots
     port 0/0/2
     no sip-register
    dial-peer voice 4 pots
     port 0/0/3
     no sip-register
    dial-peer voice 5 pots
     description ** MOH Port **
     destination-pattern ABC
     port 0/4/0
     no sip-register
    dial-peer voice 6 pots
     description tcatch all dial peer for BRI/PRIv
     translation-profile incoming nondialable
     incoming called-number .%
     direct-inward-dial
    dial-peer voice 50 pots
     description ** incoming dial peer **
     incoming called-number ^AAAA$
     direct-inward-dial
     port 0/1/0
    dial-peer voice 51 pots
     description ** incoming dial peer **
     incoming called-number ^AAAA$
     direct-inward-dial
     port 0/1/1
    dial-peer voice 2000 voip
     description ** cue voicemail pilot number **
     translation-profile outgoing XFER_TO_VM_PROFILE
     destination-pattern 98
     b2bua
     session protocol sipv2
     session target ipv4:10.1.10.1
     voice-class sip outbound-proxy ipv4:10.1.10.1
     dtmf-relay rtp-nte
     codec g711ulaw
     no vad
    dial-peer voice 2001 voip
     description ** cue auto attendant number **
     translation-profile outgoing PSTN_CallForwarding
     destination-pattern 97
     b2bua
     session protocol sipv2
     session target ipv4:10.1.10.1
     voice-class sip outbound-proxy ipv4:10.1.10.1
     dtmf-relay rtp-nte
     codec g711ulaw
     no vad
    dial-peer voice 2012 voip
     description ** cue prompt manager number **
     translation-profile outgoing PSTN_CallForwarding
     destination-pattern 96
     b2bua
     session protocol sipv2
     session target ipv4:10.1.10.1
     voice-class sip outbound-proxy ipv4:10.1.10.1
     dtmf-relay rtp-nte
     codec g711ulaw
     no vad
    dial-peer voice 1000 voip
     permission term
     description ** Incoming call from SIP trunk (VTX) **
     session protocol sipv2
     session target sip-server
     incoming called-number .%
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     fax rate 14400
     fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1001 voip
     corlist outgoing call-local
     description ** star code to SIP trunk (VTX) **
     destination-pattern *..
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     fax rate 14400
     fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1003 voip
     description ** Passthrough Inbound Calls for PSTN from CUE **
     translation-profile incoming SIP_Passthrough
     b2bua
     session protocol sipv2
     session target ipv4:10.1.10.1
     incoming called-number ABCDT
     dtmf-relay rtp-nte
     codec g711ulaw
     no vad
    dial-peer voice 1005 voip
     description ** Passthrough Inbound Calls for MWI from CUE **
     b2bua
     session protocol sipv2
     session target ipv4:10.1.10.1
     incoming called-number A80T
     dtmf-relay rtp-nte
     codec g711ulaw
     no vad
    dial-peer voice 1009 voip
     description ** Passthrough Inbound Calls for Internal Extensions from CUE **
     b2bua
     session protocol sipv2
     session target ipv4:10.1.10.1
     incoming called-number ^..$
     dtmf-relay rtp-nte
     codec g711ulaw
     no vad
    dial-peer voice 1033 voip
     corlist outgoing call-local
     description **CCA*Switzerland*Short Code Services**
     translation-profile outgoing PSTN_Outgoing
     preference 1
     destination-pattern 0187
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1042 voip
     corlist outgoing call-emergency
     description **CCA*Switzerland*Ambulance / Poisioning**
     translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
     preference 1
     destination-pattern 0014[45]
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1041 voip
     corlist outgoing call-emergency
     description **CCA*Switzerland*REGA Air Rescue**
     translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
     preference 1
     destination-pattern 00333333333
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1025 voip
     corlist outgoing call-national
     description **CCA*Switzerland*National Destination Numbers**
     translation-profile outgoing PSTN_Outgoing
     preference 1
     destination-pattern 00[789]1.......
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1020 voip
     corlist outgoing call-national
     description **CCA*Switzerland*Regional Announcement VM**
     translation-profile outgoing PSTN_Outgoing
     preference 1
     destination-pattern 01600
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1040 voip
     corlist outgoing call-emergency
     description **CCA*Switzerland*REGA Air Rescue**
     translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
     preference 1
     destination-pattern 000333333333
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1043 voip
     corlist outgoing call-emergency
     description **CCA*Switzerland*Ambulance / Poisioning**
     translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
     preference 1
     destination-pattern 014[45]
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1035 voip
     corlist outgoing call-national
     description **CCA*Switzerland*Mobile Numbers**
     translation-profile outgoing PSTN_Outgoing
     preference 1
     destination-pattern 007[46789].......
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1024 voip
     corlist outgoing call-national-plus
     description **CCA*Switzerland*Personal Numbering**
     translation-profile outgoing PSTN_Outgoing
     preference 1
     destination-pattern 00878......
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1029 voip
     corlist outgoing call-national
     description **CCA*Switzerland*Voicemail Access**
     translation-profile outgoing PSTN_Outgoing
     preference 1
     destination-pattern 00860.........
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1036 voip
     corlist outgoing call-national
     description **CCA*Switzerland*VPN Access**
     translation-profile outgoing PSTN_Outgoing
     preference 1
     destination-pattern 00869.............
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1027 voip
     corlist outgoing call-national-plus
     description **CCA*Switzerland*Premium Rate (Business)**
     translation-profile outgoing PSTN_Outgoing
     preference 1
     destination-pattern 00900......
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1026 voip
     corlist outgoing call-national
     description **CCA*Switzerland*Test Numbers**
     translation-profile outgoing PSTN_Outgoing
     preference 1
     destination-pattern 00868T
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1034 voip
     corlist outgoing call-national-plus
     description **CCA*Switzerland*Shared Cost numbers**
     translation-profile outgoing PSTN_Outgoing
     preference 1
     destination-pattern 0084[0248]......
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1038 voip
     corlist outgoing call-emergency
     description **CCA*Switzerland*Emergency**
     translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
     preference 1
     destination-pattern 0011[278]
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1037 voip
     corlist outgoing call-toll-free
     description **CCA*Switzerland*Toll Free Numbers**
     translation-profile outgoing PSTN_Outgoing
     preference 1
     destination-pattern 00800......
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1039 voip
     corlist outgoing call-emergency
     description **CCA*Switzerland*Emergency**
     translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
     preference 1
     destination-pattern 011[278]
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1032 voip
     corlist outgoing call-national
     description **CCA*Switzerland*National Destination Numbers**
     translation-profile outgoing PSTN_Outgoing
     preference 1
     destination-pattern 00[23456]........
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1023 voip
     corlist outgoing call-international
     description **CCA*Switzerland*International Calls**
     translation-profile outgoing PSTN_Outgoing
     preference 1
     destination-pattern 000T
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1031 voip
     description **CCA*Switzerland*Premium Rate (Social)**
     translation-profile outgoing PSTN_Outgoing
     preference 1
     destination-pattern 0090[16]......
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1030 voip
     corlist outgoing call-national
     description **CCA*Switzerland*Short Code**
     translation-profile outgoing PSTN_Outgoing
     preference 1
     destination-pattern 014[0357]
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1045 voip
     corlist outgoing call-emergency
     description **CCA*Switzerland*REGA/Glaciers Air Rescue**
     translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
     preference 1
     destination-pattern 0141[45]
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1028 voip
     corlist outgoing call-national-plus
     description **CCA*Switzerland*Directory Enquiries**
     translation-profile outgoing PSTN_Outgoing
     preference 1
     destination-pattern 018[15].
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1021 voip
     corlist outgoing call-national
     description **CCA*Switzerland*Short Code**
     translation-profile outgoing PSTN_Outgoing
     preference 1
     destination-pattern 011[45].
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1022 voip
     corlist outgoing call-national
     description **CCA*Switzerland*Short Code Services**
     translation-profile outgoing PSTN_Outgoing
     preference 1
     destination-pattern 01[67].
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1044 voip
     corlist outgoing call-emergency
     description **CCA*Switzerland*REGA/Glaciers Air Rescue**
     translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
     preference 1
     destination-pattern 00141[45]
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 2002 voip
     description ** cue voicemail PSTN number **
     translation-profile outgoing VM_Profile
     destination-pattern xxx$
     b2bua
     session protocol sipv2
     session target ipv4:10.1.10.1
     voice-class sip outbound-proxy ipv4:10.1.10.1
     dtmf-relay rtp-nte
     codec g711ulaw
     no vad
    dial-peer voice 2003 voip
     description ** cue auto attendant PSTN number **
     translation-profile outgoing AA_Profile
     destination-pattern xxx$
     b2bua
     session protocol sipv2
     session target ipv4:10.1.10.1
     voice-class sip outbound-proxy ipv4:10.1.10.1
     dtmf-relay rtp-nte
     codec g711ulaw
     no vad
    dial-peer voice 1110 pots
     preference 9
     destination-pattern xxx
     port 0/0/0
     no sip-register
    dial-peer voice 3006 voip
     description SIP
     translation-profile incoming SIP_Called_9
     session protocol sipv2
     session target sip-server
     incoming called-number xxx.
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    no dial-peer outbound status-check pots
    sip-ua
     keepalive target dns:site1.365873.trk.ipvoip.ch
     authentication username xxx password 7 xxx
     no remote-party-id
     retry invite 2
     retry register 10
     timers connect 100
     timers keepalive active 100
     registrar dns:site1.365873.trk.ipvoip.ch expires 3600
     sip-server dns:site1.365873.trk.ipvoip.ch
     host-registrar
    telephony-service
     sdspfarm units 5
     sdspfarm transcode sessions 10
     sdspfarm tag 2 mtpa4934c6ee4e0
     video
     fxo hook-flash
     max-ephones 40
     max-dn 300
     ip source-address 10.1.1.1 port 2000
     auto assign 1 to 1 type bri
     calling-number initiator
     service phone videoCapability 1
     service phone ehookenable 1
     service phone ehookEnable 1
     service dnis overlay
     service dnis dir-lookup
     service dss
     timeouts interdigit 5
     system message SwissT.Net
     url services http://10.1.10.1/voiceview/common/login.do
     url authentication http://10.1.10.1/voiceview/authentication/authenticate.do
     cnf-file location flash:
     cnf-file perphone
     user-locale U4 load CME-locale-de_DE-German-8.1.2.2.tar
     network-locale U4
     load 521G-524G cp524g-8-1-17
     load 525G spa525g-7-5-4
     load 501G spa50x-30x-7-5-2b
     load 502G spa50x-30x-7-5-2b
     load 504G spa50x-30x-7-5-2b
     load 508G spa50x-30x-7-5-2b
     load 509G spa50x-30x-7-5-2b
     load 525G2 spa525g-7-5-4
     load 301 spa50x-30x-7-5-2b
     load 303 spa50x-30x-7-5-2b
     time-zone 23
     time-format 24
     date-format dd-mm-yy
     keepalive 30 auxiliary 4
     voicemail 98
     max-conferences 8 gain -6
     call-forward pattern .T
     call-forward system redirecting-expanded
     hunt-group logout HLog
     moh flash:/media/music-on-hold.au
     multicast moh 239.10.16.16 port 2000
     web admin system name cisco secret 5 xxx
     dn-webedit
     time-webedit
     transfer-system full-consult dss
     transfer-pattern .T
     transfer-pattern 0.T
     transfer-pattern 6.. blind
     secondary-dialtone 0
     night-service day Sun 17:00 09:00
     night-service day Mon 17:00 09:00
     night-service day Tue 17:00 09:00
     night-service day Wed 17:00 09:00
     night-service day Thu 17:00 09:00
     night-service day Fri 17:00 09:00
     night-service day Sat 17:00 09:00
     fac standard
     create cnf-files version-stamp Jan 01 2002 00:00:00
    ephone-template  1
     url services 1 http://10.1.10.1/voiceview/common/login.do VoiceviewExpress
     service phone webAccess 0
     softkeys remote-in-use  Newcall
     softkeys idle  Redial Pickup Mobility Newcall Cfwdall Gpickup Dnd Login
     softkeys seized  Cfwdall Endcall Redial Pickup Gpickup Callback
     softkeys connected  Hold Endcall Trnsfer Mobility TrnsfVM Confrn Acct Park
     button-layout 7931 2
    ephone-template  15
     url services 1 http://10.1.10.1/voiceview/common/login.do VoiceviewExpress
     softkeys remote-in-use  Newcall
     softkeys idle  Redial Newcall Mobility Cfwdall Pickup Gpickup Dnd Login
     softkeys seized  Cfwdall Endcall Redial Pickup Gpickup Callback
     softkeys connected  Hold Endcall Mobility Trnsfer TrnsfVM Confrn Acct Park
     button-layout 7931 2
    ephone-template  16
     url services 1 http://10.1.10.1/voiceview/common/login.do VoiceviewExpress
     softkeys remote-in-use  Newcall
     softkeys idle  Redial Newcall Mobility Cfwdall Pickup Gpickup Dnd Login
     softkeys seized  Cfwdall Endcall Redial Pickup Gpickup Callback
     softkeys connected  Hold Endcall Mobility Trnsfer TrnsfVM Confrn Acct Park
    ephone-template  17
     url services 1 http://10.1.10.1/voiceview/common/login.do VoiceviewExpress
     softkeys remote-in-use  CBarge Newcall
     softkeys idle  Redial Newcall Mobility Cfwdall Pickup Gpickup Dnd Login
     softkeys seized  Cfwdall Endcall Redial Pickup Gpickup Callback
     softkeys connected  Hold Endcall Mobility Trnsfer TrnsfVM Confrn Acct Park
    ephone-template  18
     url services 1 http://10.1.10.1/voiceview/common/login.do VoiceviewExpress
     softkeys remote-in-use  CBarge Newcall
     softkeys idle  Redial Newcall Mobility Cfwdall Pickup Gpickup Dnd Login
     softkeys seized  Cfwdall Endcall Redial Pickup Gpickup Callback
     softkeys connected  Hold Endcall Mobility Trnsfer TrnsfVM Confrn Acct Park
     button-layout 7931 2
    ephone-dn  9
     number BCD no-reg primary
     description MoH
     moh out-call ABC
    ephone-dn  292
     number xxx
     description SIP Main Number registration
     preference 10
    ephone-dn  293  dual-line
     number 90 secondary xxx no-reg both
     label Zentrale
     description 90
     name Zentrale
     call-forward busy 98
     call-forward noan 98 timeout 20
    ephone-dn  294  dual-line
     number 94 secondary xxx no-reg both
     label LL
     description Lehrling Lehrnende
     name Lehrling Lehrnende
     mobility
     snr xxx delay 1 timeout 30 cfwd-noan 98
     snr ring-stop
     call-forward busy 98
     call-forward noan 98 timeout 20
    ephone-dn  295  dual-line
     number 93 secondary xxx no-reg both
     label CM
     description
     name
     snr xxx delay 1 timeout 30 cfwd-noan 98
     snr ring-stop
     call-forward busy 98
     call-forward noan 98 timeout 10
    ephone-dn  296  dual-line
     number 92 secondary xxx no-reg both
     label EE
     description
     name
     mobility
     call-forward busy 98
     call-forward noan 98 timeout 20
    ephone-dn  297  dual-line
     number 91 secondary xxx no-reg both
     label RS
     description
     name
     mobility
     snr xxx delay 1 timeout 30 cfwd-noan 98
     snr ring-stop
     call-forward busy 98
     call-forward noan 98 timeout 10
    ephone-dn  298
     number 6.. no-reg primary
     description ***CCA XFER TO VM EXTENSION***
     call-forward all 98
    ephone-dn  299
     number A801.. no-reg primary
     mwi off
    ephone-dn  300
     number A800.. no-reg primary
     mwi on
    ephone  1
     device-security-mode none
     mac-address A44C.11A0.B648
     ephone-template 1
     max-calls-per-button 2
     username "xxx" password xxx
     type 525G2
     button  1:296 2:293 3m297 4m295
     button  5m294
    ephone  2
     device-security-mode none
     mac-address A44C.11A0.B566
     ephone-template 1
     max-calls-per-button 2
     username "xxx" password xxx
     type 525G2
     button  1:297 2:293 3m296 4m295
     button  5m294
    ephone  3
     device-security-mode none
     mac-address A44C.11A0.B5C4
     ephone-template 1
     max-calls-per-button 2
     username "xxx" password xxx
     type 525G2
     button  1:295 2:293 3m297 4m296
     button  5m294
    ephone  4
     device-security-mode none
     mac-address A44C.11A0.B67A
     ephone-template 1
     max-calls-per-button 2
     username "xxx" password xxx
     type 525G2
     button  1:294 2:293 3m297 4m296
     button  5m295
    alias exec cca_voice_mode PBX
    alias exec cca_vm_notification schedule from_time=00 to_time=24
    alias exec clid-ALL_BRI ;1:0-4;1:0-9;1:0-9;1:1-9
    alias exec clid-SIP ;1:1-9;1:1-9;1:1-9
    banner login ^CCisco Configuration Assistant. Version: 3.2 (3). Fri Jul 04 13:18:33 CEST 2014^C
    line con 0
     no modem enable
    line aux 0
    line 2
     no activation-character
     no exec
     transport preferred none
     transport input all
    line vty 0 4
     transport preferred none
     transport input all
    line vty 5 100
     transport preferred none
     transport input all
    ntp master
    ntp server 91.240.0.5 prefer
    en

    Hi Patrick
    I am working on this one as well. I have a UC560 with SIP Trunk provider Les.NET.
    It was working fine until a few weeks ago when something changed on the provider end and broke it. My hunch it is something to do with the SIP REFER.
    http://www.cisco.com/c/en/us/support/docs/voice-unified-communications/unified-communications-manager-express/91535-cme-sip-trunking-config.html
    Here is an excerpt from the above page:
    Call Transfer
    When a call comes in on an SIP trunk to an SCCP Phone or CUE AutoAttendant (AA) and is transferred, the CME by default will send a SIP REFER message to the SP proxy. Most SP Proxy Servers do not support the REFER method. This needs to be configured in order to force the CME to hairpin the call:
    Router(config)#voice service voip
    Router(conf-voi-serv)#no supplementary-service sip refer
    Figure 3 shows the behavior of the CME system with the REFER method disabled.

  • Cisco 9971 phone over VPN

    Hi
    I am trying to set up a anyconnect VPN for Cisco 9971, so that I can use it at home. the tunnel has up, I've tested it but Phone got the following error message:
    11:56:11 Updating Trust List
    11:56:11 Trust List updated
    11:56:12 SEP0011111111.cnf.xml.sgn (HTTP)
    11:56:13 VPN Error: VPN is not Configured.
    12:14:40 Reset requested by CUCM
    12:15:14 DNS Timeout 
    12:15:14 Updating Trust List
    12:15:14 Trust List updated
    12:15:15 SEP0011111111.cnf.xml.sgn (HTTP)
    12:15:16 VPN Error: VPN is not Configured.
    Any help would be appreciated.
    By the way, this is a SIP phone.

    Hi,
    You can purchase the phone proxy license. This elimiates the need to build a VPN tunnel for voice traffic.
    It is not mandatory to purchase this license however.
    From the ASA configuration guide:
    http://www.cisco.com/en/US/docs/security/asa/asa83/configuration/guide/unified_comm_phoneproxy.html#wp1144845
    "The  Cisco Phone Proxy on the adaptive security  appliance bridges IP  telephony between the corporate IP telephony  network and the Internet  in a secure manner by forcing data from remote  phones on an untrusted  network to be encrypted. "
    Don't forget to rate all posts that are helpful.

  • Calling issue with Cisco 7937 conference station

    Hi Friends,
    I am facing issue wiht Cisco 7937 conference station, our customer have various branch offices accross the world. All branches are connected over MPLS through service provider( SIP service provider) . there is a centralized CUCM and remote office have SIP Voice gateways .
    When making calls from once remote site to another using Cisco 6921 phones calls working fine
    When making calls from once remote site to another using Cisco 7937 conference station to make call  any phone at remote office, calls are getting disconneted, remote phone rings when calls,  but its gets fast busy tone when other party picks up the phone and  not able to talk.
    I suspect the issue with Codec but we have configured transcoders  in VG and registered with CUCM
    Please help me if any one experience such issue earlier.
    Regards
    Siva

    hi Basant,
    1. Actually tow phones A and B are registerd with centralized CUCM, A and B are located in two different locations, RTP traffic between And B pass through service provider. 
    Call Flow --> Phone A ---->CUCMRouterpattern--> SIP trunk ----> Voice gateway--->Service provider cloud---> Respective Voice Gateway---> CUCM -- Phone B
    Show Run
    =~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2014.02.27 15:14:52 =~=~=~=~=~=~=~=~=~=~=~=
    sh run
    Building configuration...
    Current configuration : 12139 bytes
    ! Last configuration change at 06:35:59 UTC Tue Feb 25 2014
    ! NVRAM config last updated at 11:16:38 UTC Mon Feb 24 2014 by administrator
    ! NVRAM config last updated at 11:16:38 UTC Mon Feb 24 2014 by administrator
    version 15.1
    service timestamps debug datetime msec
    service timestamps log datetime msec
    no service password-encryption
    hostname eucamvgw01
    boot-start-marker
    boot system flash:c2900-universalk9-mz.SPA.151-4.M5.bin
    boot-end-marker
    card type e1 0 0
    logging buffered 51200 warnings
    no logging console
    no aaa new-model
    no network-clock-participate wic 0
    no ipv6 cef
    ip source-route
    ip traffic-export profile cuecapture mode capture
    bidirectional
    ip cef
    ip multicast-routing
    ip domain name drreddys.eu
    ip name-server 10.197.20.1
    ip name-server 10.197.20.2
    multilink bundle-name authenticated
    stcapp ccm-group 2
    stcapp
    stcapp feature access-code
    stcapp feature speed-dial
    stcapp supplementary-services
    port 0/1/0
    fallback-dn 5428025
    port 0/1/1
    fallback-dn 5428008
    port 0/1/2
    fallback-dn 5421462
    port 0/1/3
    fallback-dn 5421463
    isdn switch-type primary-net5
    crypto pki token default removal timeout 0
    voice-card 0
    dsp services dspfarm
    voice call send-alert
    voice call disc-pi-off
    voice call convert-discpi-to-prog
    voice rtp send-recv
    voice service voip
    ip address trusted list
    ipv4 10.198.0.0 255.255.255.0
    ipv4 152.63.1.0 255.255.255.0
    address-hiding
    allow-connections sip to sip
    no supplementary-service h225-notify cid-update
    no supplementary-service sip moved-temporarily
    no supplementary-service sip refer
    fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
    fax-relay ans-disable
    sip
    rel1xx supported "track"
    privacy pstn
    no update-callerid
    early-offer forced
    call-route p-called-party-id
    voice class uri 100 sip
    host 41.206.187.71
    voice class codec 10
    codec preference 1 g711alaw
    codec preference 2 g711ulaw
    codec preference 3 ilbc
    codec preference 4 g729r8
    codec preference 5 g729br8
    voice class codec 20
    codec preference 1 g729br8
    codec preference 2 g729r8
    voice moh-group 1
    moh flash:moh/Panjo.alaw.wav
    description MOH G711 alaw
    multicast moh 239.1.1.2 port 16384 route 10.198.2.9
    voice translation-rule 1
    rule 1 /^012237280\(..\)/ /54280\1/
    rule 2 /^012236514\(..\)/ /54214\1/
    rule 3 /^01223651081/ /5428010/
    rule 4 /^01223506701/ /5428010/
    voice translation-rule 2
    rule 1 /^00\(.+\)/ /+\1/
    rule 2 /^0\(.+\)/ /+44\1/
    rule 3 /^\([0-9].+\)/ /+\1/
    voice translation-rule 3
    rule 1 /^9\(.+\)/ /\1/
    rule 2 /^\+44\(.+\)/ /0\1/
    rule 3 /^\+\(.+\)/ /00\1/
    voice translation-rule 4
    rule 1 /^54280\(..\)/ /12237280\1/
    rule 2 /^54214\(..\)/ /12236514\1/
    rule 3 /^\+44\(.+\)/ /\1/
    rule 4 /^.54280\(..\)/ /12237280\1/
    rule 5 /^.54214\(..\)/ /12236514\1/
    voice translation-rule 9
    rule 1 /^\(....\)/ /542\1/
    voice translation-rule 10
    voice translation-rule 11
    rule 1 /^\+44122372\(....\)/ /542\1/
    rule 2 /^\+44122365\(....\)/ /542\1/
    voice translation-rule 12
    voice translation-rule 13
    rule 1 /^\([18]...\)/ /542\1/
    voice translation-rule 14
    voice translation-profile MPLS-incoming
    translate calling 10
    translate called 9
    voice translation-profile MPLS-outgoing
    translate calling 11
    translate called 12
    voice translation-profile PSTN-incoming
    translate calling 2
    translate called 1
    voice translation-profile PSTN-outgoing
    translate calling 4
    translate called 3
    voice translation-profile SRST-incoming
    translate calling 14
    translate called 13
    license udi pid CISCO2921/K9 sn FGL145110RE
    hw-module ism 0
    hw-module pvdm 0/0
    username administrator privilege 15 secret 5 $1$syu5$DsxdOgfS7Wltx78o4PV.60
    redundancy
    controller E1 0/0/0
    ip tcp path-mtu-discovery
    ip scp server enable
    interface Embedded-Service-Engine0/0
    no ip address
    shutdown
    interface GigabitEthernet0/0
    description internal LAN
    ip address 10.198.2.9 255.255.255.0
    duplex auto
    speed auto
    interface ISM0/0
    ip unnumbered GigabitEthernet0/0
    service-module ip address 10.198.2.8 255.255.255.0
    !Application: CUE Running on ISM
    service-module ip default-gateway 10.198.2.9
    interface GigabitEthernet0/1
    description to TATA NGN
    ip address 115.114.225.122 255.255.255.252
    duplex auto
    speed auto
    interface GigabitEthernet0/2
    description SIP Trunks external
    ip address 79.121.254.83 255.255.255.248
    ip access-group SIP-InBound in
    ip traffic-export apply cuecapture size 8000000
    duplex auto
    speed auto
    interface ISM0/1
    description Internal switch interface connected to Internal Service Module
    no ip address
    shutdown
    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 10.198.2.1
    ip route 10.198.2.8 255.255.255.255 ISM0/0
    ip route 41.206.187.0 255.255.255.0 115.114.225.121
    ip route 77.37.25.46 255.255.255.255 79.121.254.81
    ip route 83.245.6.81 255.255.255.255 79.121.254.81
    ip route 83.245.6.82 255.255.255.255 79.121.254.81
    ip route 95.223.1.107 255.255.255.255 79.121.254.81
    ip route 192.54.47.0 255.255.255.0 79.121.254.81
    ip access-list extended SIP-InBound
    permit ip host 77.37.25.46 any
    permit ip host 83.245.6.81 any
    permit ip host 83.245.6.82 any
    permit ip 192.54.47.0 0.0.0.255 any
    permit icmp any any
    permit ip host 95.223.1.107 any
    deny ip any any log
    control-plane
    voice-port 0/1/0
    compand-type a-law
    timeouts initial 60
    timeouts interdigit 60
    timeouts ringing infinity
    caller-id enable
    voice-port 0/1/1
    compand-type a-law
    timeouts initial 60
    timeouts interdigit 60
    timeouts ringing infinity
    caller-id enable
    voice-port 0/1/2
    compand-type a-law
    timeouts initial 60
    timeouts interdigit 60
    timeouts ringing infinity
    caller-id enable
    voice-port 0/1/3
    compand-type a-law
    timeouts initial 60
    timeouts interdigit 60
    timeouts ringing infinity
    caller-id enable
    no ccm-manager fax protocol cisco
    ccm-manager music-on-hold bind GigabitEthernet0/0
    ccm-manager config server 152.63.1.19 152.63.1.100 172.27.210.5
    ccm-manager sccp local GigabitEthernet0/0
    ccm-manager sccp
    mgcp profile default
    sccp local GigabitEthernet0/0
    sccp ccm 10.198.2.9 identifier 3 priority 3 version 7.0
    sccp ccm 152.63.1.19 identifier 4 version 7.0
    sccp ccm 152.63.1.100 identifier 5 version 7.0
    sccp ccm 172.27.210.5 identifier 6 version 7.0
    sccp
    sccp ccm group 2
    bind interface GigabitEthernet0/0
    associate ccm 4 priority 1
    associate ccm 5 priority 2
    associate ccm 6 priority 3
    associate ccm 3 priority 4
    associate profile 1002 register CFB_UK_CAM_02
    associate profile 1001 register XCODE_UK_CAM_02
    associate profile 1000 register MTP_UK_CAM_02
    dspfarm profile 1001 transcode
    codec ilbc
    codec g722-64
    codec g729br8
    codec g729r8
    codec gsmamr-nb
    codec pass-through
    codec g711ulaw
    codec g711alaw
    codec g729ar8
    codec g729abr8
    maximum sessions 18
    associate application SCCP
    dspfarm profile 1002 conference
    codec g711ulaw
    codec g711alaw
    codec g729ar8
    codec g729abr8
    codec g729r8
    codec g729br8
    maximum sessions 2
    associate application SCCP
    dspfarm profile 1000 mtp
    codec g711alaw
    maximum sessions software 200
    associate application SCCP
    dial-peer cor custom
    name SRSTMode
    dial-peer cor list SRST
    member SRSTMode
    dial-peer voice 100 voip
    description *** Inbound CUCM ***
    translation-profile incoming PSTN-incoming
    incoming called-number .
    voice-class codec 10
    voice-class sip call-route p-called-party-id
    dtmf-relay rtp-nte
    no vad
    dial-peer voice 500 voip
    description *** Inbound TATA MPLS ***
    translation-profile incoming MPLS-incoming
    session protocol sipv2
    session target sip-server
    incoming called-number ....
    incoming uri from 100
    voice-class codec 20
    dtmf-relay rtp-nte
    no vad
    dial-peer voice 510 voip
    description *** Outbound TATA MPLS ***
    translation-profile outgoing MPLS-outgoing
    destination-pattern 54[013-9]....
    session protocol sipv2
    session target ipv4:41.206.187.71
    session transport udp
    voice-class codec 20
    dtmf-relay rtp-nte
    no vad
    dial-peer voice 520 voip
    description *** Outbound TATA MPLS ***
    translation-profile outgoing MPLS-outgoing
    destination-pattern 5[0-35-9].....
    session protocol sipv2
    session target ipv4:41.206.187.71
    session transport udp
    voice-class codec 20
    dtmf-relay rtp-nte
    no vad
    dial-peer voice 200 voip
    description *** Inbound M12 *** 01223651081, 01223651440 - 01223651489
    translation-profile incoming PSTN-incoming
    session protocol sipv2
    session target sip-server
    session transport udp
    incoming called-number 0122365....
    dtmf-relay rtp-nte
    codec g711ulaw
    no vad
    dial-peer voice 201 voip
    description *** Inbound M12 *** 012237280XX
    translation-profile incoming PSTN-incoming
    session protocol sipv2
    session target sip-server
    session transport udp
    incoming called-number 012237280..
    dtmf-relay rtp-nte
    codec g711ulaw
    no vad
    dial-peer voice 202 voip
    description *** Inbound M12 *** 01223506701
    translation-profile incoming PSTN-incoming
    session protocol sipv2
    session target sip-server
    session transport udp
    incoming called-number 01223506701
    dtmf-relay rtp-nte
    codec g711ulaw
    no vad
    dial-peer voice 210 voip
    description *** Outbound M12 ***
    translation-profile outgoing PSTN-outgoing
    destination-pattern +...T
    session protocol sipv2
    session target ipv4:83.245.6.81
    session transport udp
    dtmf-relay rtp-nte
    codec g711alaw
    no vad
    dial-peer voice 211 voip
    description *** Outbound ISDN for SRST and emergency ***
    translation-profile outgoing PSTN-outgoing
    destination-pattern 9.T
    session protocol sipv2
    session target ipv4:83.245.6.81
    session transport udp
    dtmf-relay rtp-nte
    codec g711alaw
    no vad
    dial-peer voice 212 voip
    description *** Outbound ISDN for emergency ***
    translation-profile outgoing PSTN-outgoing
    destination-pattern 11[02]
    session protocol sipv2
    session target ipv4:83.245.6.81
    session transport udp
    dtmf-relay rtp-nte
    codec g711alaw
    no vad
    dial-peer voice 2000 voip
    description *** Outbound to CUCM Primary ***
    preference 1
    destination-pattern 542....
    session protocol sipv2
    session target ipv4:152.63.1.19
    voice-class codec 10
    voice-class sip call-route p-called-party-id
    dtmf-relay rtp-nte
    no vad
    dial-peer voice 2001 voip
    description *** Outbound to CUCM Secondary ***
    preference 2
    destination-pattern 542....
    session protocol sipv2
    session target ipv4:152.63.1.100
    voice-class codec 10
    voice-class sip call-route p-called-party-id
    dtmf-relay rtp-nte
    no vad
    dial-peer voice 2002 voip
    description *** Outbound to CUCM Teritiary ***
    preference 3
    destination-pattern 542....
    session protocol sipv2
    session target ipv4:172.27.210.5
    voice-class codec 10
    voice-class sip call-route p-called-party-id
    dtmf-relay rtp-nte
    no vad
    dial-peer voice 999010 pots
    service stcapp
    port 0/1/0
    dial-peer voice 999011 pots
    service stcapp
    port 0/1/1
    dial-peer voice 999012 pots
    service stcapp
    port 0/1/2
    dial-peer voice 999013 pots
    service stcapp
    port 0/1/3
    sip-ua
    no remote-party-id
    gatekeeper
    shutdown
    call-manager-fallback
    secondary-dialtone 9
    max-conferences 4 gain -6
    transfer-system full-consult
    ip source-address 10.198.2.9 port 2000
    max-ephones 110
    max-dn 400 dual-line no-reg
    translation-profile incoming SRST-incoming
    moh flash:/moh/Panjo.ulaw.wav
    multicast moh 239.1.1.1 port 16384 route 10.198.2.9
    time-zone 22
    time-format 24
    date-format dd-mm-yy
    line con 0
    login local
    line aux 0
    line 2
    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 131
    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
    session-timeout 60
    exec-timeout 60 0
    privilege level 15
    login local
    transport input all
    line vty 5 15
    session-timeout 60
    exec-timeout 60 0
    privilege level 15
    login local
    transport input all
    scheduler allocate 20000 1000
    ntp server 10.1.30.1
    end
    eucamvgw01#
    Sh SCCP
    =~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2014.03.03 17:57:44 =~=~=~=~=~=~=~=~=~=~=~=
    SCCP Admin State: UP
    Gateway Local Interface: GigabitEthernet0/0
    IPv4 Address: 10.198.2.9
    Port Number: 2000
    IP Precedence: 5
    User Masked Codec list: None
    Call Manager: 10.198.2.9, Port Number: 2000
    Priority: 3, Version: 7.0, Identifier: 3
    Call Manager: 152.63.1.19, Port Number: 2000
    Priority: N/A, Version: 7.0, Identifier: 4
    Trustpoint: N/A
    Call Manager: 152.63.1.100, Port Number: 2000
    Priority: N/A, Version: 7.0, Identifier: 5
    Trustpoint: N/A
    Call Manager: 172.27.210.5, Port Number: 2000
    Priority: N/A, Version: 7.0, Identifier: 6
    Trustpoint: N/A
    MTP Oper State: ACTIVE - Cause Code: NONE
    Active Call Manager: 152.63.1.19, Port Number: 2000
    TCP Link Status: CONNECTED, Profile Identifier: 1000
    Reported Max Streams: 400, Reported Max OOS Streams: 0
    Supported Codec: g711alaw, Maximum Packetization Period: 30
    Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
    Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30
    Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30
    TLS : ENABLED
    Transcoding Oper State: ACTIVE - Cause Code: NONE
    Active Call Manager: 152.63.1.19, Port Number: 2000
    TCP Link Status: CONNECTED, Profile Identifier: 1001
    Reported Max Streams: 36, Reported Max OOS Streams: 0
    Supported Codec: ilbc, Maximum Packetization Period: 120
    Supported Codec: g722r64, Maximum Packetization Period: 30
    Supported Codec: g729br8, Maximum Packetization Period: 60
    Supported Codec: g729r8, Maximum Packetization Period: 60
    Supported Codec: gsmamr-nb, Maximum Packetization Period: 60
    Supported Codec: pass-thru, Maximum Packetization Period: N/A
    Supported Codec: g711ulaw, Maximum Packetization Period: 30
    Supported Codec: g711alaw, Maximum Packetization Period: 30
    Supported Codec: g729ar8, Maximum Packetization Period: 60
    Supported Codec: g729abr8, Maximum Packetization Period: 60
    Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
    Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30
    Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30
    Conferencing Oper State: ACTIVE - Cause Code: NONE
    Active Call Manager: 152.63.1.19, Port Number: 2000
    TCP Link Status: CONNECTED, Profile Identifier: 1002
    Reported Max Streams: 16, Reported Max OOS Streams: 0
    Supported Codec: g711ulaw, Maximum Packetization Period: 30
    Supported Codec: g711alaw, Maximum Packetization Period: 30
    Supported Codec: g729ar8, Maximum Packetization Period: 60
    Supported Codec: g729abr8, Maximum Packetization Period: 60
    Supported Codec: g729r8, Maximum Packetization Period: 60
    Supported Codec: g729br8, Maximum Packetization Period: 60
    Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
    Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30
    Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30
    TLS : ENABLED
    Alg_Phone Oper State: ACTIVE - Cause Code: NONE
    Active Call Manager: 152.63.1.19, Port Number: 2000
    TCP Link Status: CONNECTED, Device Name: AN71FEF7F070080
    Reported Max Streams: 1, Reported Max OOS Streams: 0
    Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
    Supported Codec: g711ulaw, Maximum Packetization Period: 20
    Supported Codec: g711alaw, Maximum Packetization Period: 20
    Supported Codec: g729r8, Maximum Packetization Period: 220Supported Codec: g729ar8, Maximum Packetization Period: 220
    Supported Codec: g729br8, Maximum Packetization Period: 220
    Supported Codec: g729r8, Maximum Packetization Period: 220
    Supported Codec: ilbc, Maximum Packetization Period: 120
    Alg_Phone Oper State: ACTIVE - Cause Code: NONE
    Active Call Manager: 152.63.1.19, Port Number: 2000
    TCP Link Status: CONNECTED, Device Name: AN71FEF7F070081
    Reported Max Streams: 1, Reported Max OOS Streams: 0
    Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
    Supported Codec: g711ulaw, Maximum Packetization Period: 20
    Supported Codec: g711alaw, Maximum Packetization Period: 20
    Supported Codec: g729r8, Maximum Packetization Period: 220
    Supported Codec: g729ar8, Maximum Packetization Period: 220
    Supported Codec: g729br8, Maximum Packetization Period: 220
    Supported Codec: g729r8, Maximum Packetization Period: 220
    Supported Codec: ilbc, Maximum Packetization Period: 120
    Alg_Phone Oper State: ACTIVE - Cause Code: NONE
    Active Call Manager: 152.63.1.19, Port Number: 2000
    TCP Link Status: CONNECTED, Device Name: AN71FEF7F070082
    Reported Max Streams: 1, Reported Max OOS Streams: 0
    Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
    Supported Codec: g711ulaw, Maximum Packetization Period: 20Supported Codec: g711alaw, Maximum Packetization Period: 20
    Supported Codec: g729r8, Maximum Packetization Period: 220
    Supported Codec: g729ar8, Maximum Packetization Period: 220
    Supported Codec: g729br8, Maximum Packetization Period: 220
    Supported Codec: g729r8, Maximum Packetization Period: 220
    Supported Codec: ilbc, Maximum Packetization Period: 120
    Alg_Phone Oper State: ACTIVE - Cause Code: NONE
    Active Call Manager: 152.63.1.19, Port Number: 2000
    TCP Link Status: CONNECTED, Device Name: AN71FEF7F070083
    Reported Max Streams: 1, Reported Max OOS Streams: 0
    Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
    Supported Codec: g711ulaw, Maximum Packetization Period: 20
    Supported Codec: g711alaw, Maximum Packetization Period: 20
    Supported Codec: g729r8, Maximum Packetization Period: 220
    Supported Codec: g729ar8, Maximum Packetization Period: 220
    Supported Codec: g729br8, Maximum Packetization Period: 220
    Supported Codec: g729r8, Maximum Packetization Period: 220
    Supported Codec: ilbc, Maximum Packetization Period: 120
    eucamvgw01#

  • Call forward to PSTN on cme

    Hi,
    unable to set up call forward to PSTN.
    /* Style Definitions */
    table.MsoNormalTable
    {mso-style-name:"Table Normal";
    mso-tstyle-rowband-size:0;
    mso-tstyle-colband-size:0;
    mso-style-noshow:yes;
    mso-style-priority:99;
    mso-style-qformat:yes;
    mso-style-parent:"";
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    mso-fareast-theme-font:minor-fareast;
    mso-hansi-font-family:Calibri;
    mso-hansi-theme-font:minor-latin;
    mso-bidi-font-family:"Times New Roman";
    mso-bidi-theme-font:minor-bidi;}
    I have tried activating the Call forward via the phone or manually via the config, but when I attempt a call to IP Communicator from PSTN or via extn I am not seeing re-INVITE which should be generated for the forwarded call. Am i missing something?
        PSTN / IP phone ------> Calling extn on CME (which is call forwarded to another PSTN number)
    config below:
    voice service voip
    ip address trusted list
      ipv4 0.0.0.0 0.0.0.0
    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 sip moved-temporarily
    no supplementary-service sip refer
    fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
    sip
      registrar server expires max 250 min 200
      asserted-id pai
      localhost dns:XXXXX
      outbound-proxy dns:XXXXX
    dial-peer voice 100 voip
    description ** Incoming call from SIP trunk **
    session protocol sipv2
    session target sip-server
    incoming called-number .
    voice-class codec 1 
    voice-class sip dtmf-relay force rtp-nte
    dtmf-relay rtp-nte
    no vad
    dial-peer voice 101 voip
    description ** Outgoinging call to SIP trunk **
    translation-profile outgoing SIPOUT
    destination-pattern 1T
    session protocol sipv2
    session target sip-server
    voice-class codec 1 
    voice-class sip dtmf-relay force rtp-nte
    voice-class sip profiles 101
    dtmf-relay rtp-nte
    no vad
    dial-peer voice 102 voip
    description ** Outgoinging call to SIP trunk **
    destination-pattern 0[2-9].T
    session protocol sipv2
    session target sip-server
    voice-class codec 1 
    voice-class sip dtmf-relay force rtp-nte
    dtmf-relay rtp-nte
    no vad
    no supplementary-service sip moved-temporarily
    no supplementary-service sip refer
    telephony-service
    max-ephones 4
    max-dn 12
    ip source-address 192.168.100.2 port 2000
    calling-number initiator
    timeouts interdigit 5
    load 7960-7940 P00308010200
    date-format dd-mm-yy
    max-conferences 8 gain -6
    call-forward pattern .T
    call-forward system redirecting-expanded
    transfer-system full-consult dss
    transfer-pattern .T
    transfer-pattern 0.T
    create cnf-files version-stamp Jan 01 2002 00:00:00
    ephone-dn  1  dual-line
    number 4961 secondary 99474961 no-reg both
    label 4961
    name 4961
    call-forward all 021605547

    /* Style Definitions */
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    mso-tstyle-rowband-size:0;
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    mso-fareast-theme-font:minor-fareast;
    mso-hansi-font-family:Calibri;
    mso-hansi-theme-font:minor-latin;}
    Does a direct call (without forwarding) work through this dial-peer? YES
    The session target of dial-peer 101 is the "sip-server". In wich way is configured? Is it an IP address or a name? FQDN
    Can you ping it from the CME? YES
    The CME can resolve the name via DNS? Resolved on the CME Can you post the sip-ua config?
    sip-ua
    credentials number 99474960 username 99474960 password 7 XXXXXXXXX realm as-test.xys.net 
    authentication username 99474960 password 7 XXXXXXX 
    calling-info pstn-to-sip asserted-id number set 99474960 
    no remote-party-id 
    disable-early-media 180 
    retry invite 2
    retry register 3
    timers connect 100 
    registrar dns:as-test.xys.net expires 60  sip-server dns:as-test.xys.net 
    host-registrar

  • CUCM 8.6 Call Forwarding to External Number Issue

    Hello,
    Call forwarding worked without problems, we could forward our phones to external numbers and everything was ok, when somebody called to my phone, I could  got the call to my cell phone.
    But now when I forward my phone to external number and try to call to my phone I get busy trigger.
    We didn't change configuration or install any update.
    I think its my ISP-s problem, to whom we have SIP Trunk.
    I don't understand log file, so can you tell what is the problem?
    Here is log:
    057729XXXX is called party, cell phone number
    original calling party number is 240XXXXX, but it is forwarded to 2484XXX
    INVITE sip:2484XXX@ISP-IP:5060 SIP/2.0
    Via: SIP/2.0/UDP MY-CUCM-IP:5060;branch=z9hG4bK1003a84126249
    From: <sip:057729XXXX@MY-CUCM>;tag=4052091~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
    To: <sip:2484XXX@ISP-IP>
    Date: Wed, 18 Dec 2013 13:34:18 GMT
    Call-ID: 16d82e80-2b11a45a-c43e7-84450d0a@MY-CUCM-IP
    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
    Expires: 180
    Allow-Events: presence
    Supported: X-cisco-srtp-fallback
    Supported: Geolocation
    Cisco-Guid: 0383266432-0000065536-0000191815-2219117834
    Session-Expires:  1800
    P-Asserted-Identity: <sip:057729XXXX@MY-CUCM-IP>
    Remote-Party-ID: <sip:057729XXXX@MY-CUCM-IP>;party=calling;screen=yes;privacy=off
    Contact: <sip:057729XXXX@MY-CUCM-IP:5060>
    Max-Forwards: 68
    Content-Type: application/sdp
    Content-Length: 215
    v=0
    o=CiscoSystemsCCM-SIP 4052091 1 IN IP4 MY-CUCM-IP
    s=SIP Call
    c=IN IP4 MY-CUCM-IP
    t=0 0
    m=audio 29790 RTP/AVP 8 101
    a=rtpmap:8 PCMA/8000
    a=ptime:20
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    |2,100,56,1.173711429^MY-CUCM-IP^MTP_3
    17:34:18.526 |EnvProcessUdpPort - EnvProcessUdpHandler::fireSignal() varId = 2|2,100,56,1.173711429^MY-CUCM-IP^MTP_3
    17:34:18.526 |EnvProcessUdpHandler::fireSignal - SEND: index = 2, handler = 0xb2d59c98|*^*^*
    17:34:18.526 |EnvProcessUdpPort::fireSignal - SEND, destination = ISP-IP:5060|*^*^*
    17:34:18.526 |EnvProcessUdpPort - EnvProcessUdpHandler::send(buff, 1172, ISP-IP:5060)|*^*^*
    17:34:18.536 |EnvProcessUdpHandler::handle_input - handle = 334|*^*^*
    17:34:18.536 |EnvProcessUdpHandler::handle_input   Status: 0, Id: 2|*^*^*
    17:34:18.536 |//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 358 from ISP-IP:[5060]:
    [12623361,NET]
    SIP/2.0 100 Trying
    Call-ID: 16d82e80-2b11a45a-c43e7-84450d0a@MY-CUCM-IP
    CSeq: 101 INVITE
    From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052091~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
    To: <sip:2484XXX@ISP-IP>;tag=sip+1+b3a00013+867def6a
    Via: SIP/2.0/UDP MY-CUCM-IP:5060;branch=z9hG4bK1003a84126249
    Server: CISCO-SBC/2.x
    Content-Length: 0
    |2,100,230,1.4901096^ISP-IP^*
    17:34:18.536 |//SIP/Stack/Info/0x0/ccsip_spi_get_msg_type returned: 2 for event 1|2,100,230,1.4901096^ISP-IP^*
    17:34:18.536 |//SIP/Stack/Transport/0x0/context=(nil)|2,100,230,1.4901096^ISP-IP^*
    17:34:18.536 |//SIP/Stack/Transport/0x0/gConnTab=0xf484290, addr=ISP-IP, port=5060, connid=2, transport=UDP|2,100,230,1.4901096^ISP-IP^*
    17:34:18.536 |//SIP/Stack/Info/0x0/Return existing connection for port 5060 connId 2|2,100,230,1.4901096^ISP-IP^*
    17:34:18.536 |//SIP/Stack/Info/0x0/Checking Invite Dialog|2,100,230,1.4901096^ISP-IP^*
    17:34:18.536 |//SIP/Stack/Info/0xb1b50c90/INVITE response with no RSEQ - disable IS_REL1XX|2,100,230,1.4901096^ISP-IP^*
    17:34:18.536 |//SIP/SIPHandler/ccbId=0/scbId=0/sip_stop_timer: type=SIP_TIMER_TRYING value=500 retries=3|2,100,230,1.4901096^ISP-IP^*
    17:34:18.536 |//SIP/Stack/States/0xb1b50c90/0xb1b50c90 : State change from (STATE_SENT_INVITE, SUBSTATE_NONE)  to (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_PROCEEDING)|2,100,230,1.4901096^ISP-IP^*
    17:34:18.536 |//SIP/SIPHandler/ccbId=0/scbId=0/sip_stop_timer: type=SIP_TIMER_EXPIRES value=180000 retries=0|2,100,230,1.4901096^ISP-IP^*
    17:34:18.536 |//SIP/SIPHandler/ccbId=0/scbId=0/sip_start_timer: type=SIP_TIMER_EXPIRES value=180000 retries=0|2,100,230,1.4901096^ISP-IP^*
    17:34:18.561 |EnvProcessUdpHandler::handle_input - handle = 334|*^*^*
    17:34:18.561 |EnvProcessUdpHandler::handle_input   Status: 0, Id: 2|*^*^*
    17:34:18.561 |//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 396 from ISP-IP:[5060]:
    [12623362,NET]
    SIP/2.0 403 Forbidden
    Call-ID: 16d82e80-2b11a45a-c43e7-84450d0a@MY-CUCM-IP
    CSeq: 101 INVITE
    From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052091~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
    To: <sip:2484XXX@ISP-IP>;tag=sip+1+b3a00013+867def6a
    Via: SIP/2.0/UDP MY-CUCM-IP:5060;branch=z9hG4bK1003a84126249
    Server: CISCO-SBC/2.x
    Content-Length: 0
    Contact: <sip:ISP-IP:5060>
    [12623363,NET]
    ACK sip:2484XXX@ISP-IP:5060 SIP/2.0
    Via: SIP/2.0/UDP MY-CUCM-IP:5060;branch=z9hG4bK1003a84126249
    From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052091~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
    To: <sip:2484XXX@ISP-IP>;tag=sip+1+b3a00013+867def6a
    Date: Wed, 18 Dec 2013 13:34:18 GMT
    Call-ID: 16d82e80-2b11a45a-c43e7-84450d0a@MY-CUCM-IP
    Max-Forwards: 70
    CSeq: 101 ACK
    Allow-Events: presence
    Content-Length: 0
    INVITE sip:2484XXX@ISP's-Other-IP:5062 SIP/2.0
    Via: SIP/2.0/UDP MY-CUCM-IP:5062;branch=z9hG4bK1003a95b8f3900
    From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052092~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
    To: <sip:2484XXX@ISP's-Other-IP>
    Date: Wed, 18 Dec 2013 13:34:18 GMT
    Call-ID: 16d82e80-2b11a45a-c43e8-84450d0a@MY-CUCM-IP
    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
    Expires: 180
    Allow-Events: presence
    Supported: X-cisco-srtp-fallback
    Supported: Geolocation
    Cisco-Guid: 0383266432-0000065536-0000191816-2219117834
    Session-Expires:  1800
    P-Asserted-Identity: <sip:057729XXXX@MY-CUCM-IP>
    Remote-Party-ID: <sip:057729XXXX@MY-CUCM-IP>;party=calling;screen=yes;privacy=off
    Contact: <sip:057729XXXX@MY-CUCM-IP:5062>
    Max-Forwards: 68
    Content-Type: application/sdp
    Content-Length: 215
    v=0
    o=CiscoSystemsCCM-SIP 4052092 1 IN IP4 MY-CUCM-IP
    s=SIP Call
    c=IN IP4 MY-CUCM-IP
    t=0 0
    m=audio 29792 RTP/AVP 8 101
    a=rtpmap:8 PCMA/8000
    a=ptime:20
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    |2,100,56,1.173711431^MY-CUCM-IP^MTP_3
    17:34:18.567 |EnvProcessUdpPort - EnvProcessUdpHandler::fireSignal() varId = 0|2,100,56,1.173711431^MY-CUCM-IP^MTP_3
    17:34:18.567 |EnvProcessUdpHandler::fireSignal - SEND: index = 0, handler = 0xa6b4d7c0|*^*^*
    17:34:18.567 |EnvProcessUdpPort::fireSignal - SEND, destination = ISP's-Other-IP:5062|*^*^*
    17:34:18.567 |EnvProcessUdpPort - EnvProcessUdpHandler::send(buff, 1177, ISP's-Other-IP:5062)|*^*^*
    17:34:18.569 |EnvProcessUdpHandler::handle_input - handle = 335|*^*^*
    17:34:18.569 |EnvProcessUdpHandler::handle_input   Status: 0, Id: 0|*^*^*
    17:34:18.569 |//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 394 from ISP's-Other-IP:[5062]:
    [12623365,NET]
    SIP/2.0 100 trying -- your call is important to us
    Via: SIP/2.0/UDP MY-CUCM-IP:5062;branch=z9hG4bK1003a95b8f3900;rport=5062
    From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052092~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
    To: <sip:2484XXX@ISP's-Other-IP>
    Call-ID: 16d82e80-2b11a45a-c43e8-84450d0a@MY-CUCM-IP
    CSeq: 101 INVITE
    Server: kamailio (3.3.1 (x86_64/linux))
    Content-Length: 0
    17:34:18.587 |//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 375 from ISP's-Other-IP:[5062]:
    [12623366,NET]
    SIP/2.0 403 Forbidden
    Via: SIP/2.0/UDP MY-CUCM-IP:5062;branch=z9hG4bK1003a95b8f3900;rport=5062
    Call-ID: 16d82e80-2b11a45a-c43e8-84450d0a@MY-CUCM-IP
    From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052092~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
    To: <sip:2484XXX@ISP's-Other-IP>;tag=dc6a4ae7
    CSeq: 101 INVITE
    Reason: Q.850;cause=0;text="unknown"
    Content-Length: 0
    |2,100,230,1.4901099^ISP's-Other-IP^*
    [12623367,NET]
    ACK sip:2484XXX@ISP's-Other-IP:5062 SIP/2.0
    Via: SIP/2.0/UDP MY-CUCM-IP:5062;branch=z9hG4bK1003a95b8f3900
    From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052092~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
    To: <sip:2484XXX@ISP's-Other-IP>;tag=dc6a4ae7
    Date: Wed, 18 Dec 2013 13:34:18 GMT
    Call-ID: 16d82e80-2b11a45a-c43e8-84450d0a@MY-CUCM-IP
    Max-Forwards: 70
    CSeq: 101 ACK
    Allow-Events: presence
    Content-Length: 0

    SIP/2.0 403 Forbidden error
    If your router is sending a SIP/2.0 403 Forbidden error to the SIP server you are registered to, there is a good chance your  router is blocking the incoming call due to the toll-faud prevention  feature that was added to IOS version 15.1(2)T.
    How to Identify if TOLLFRAUD_APP is Blocking Your Call
    If the TOLLFRAUD_APP is rejecting the call, it generates a Q.850       disconnect cause value of 21, which represents ‘Call Rejected’. The       debug voip ccapi inout command can be run to       identify the cause value.
    Additionally, voice iec syslog can be       enabled to further verify if the call failure is a result of the toll-fraud       prevention. This configuration, which is often handy to troubleshoot the origin       of failure from a gateway perspective, will print out that the call is being       rejected due to toll call fraud. The CCAPI and Voice IEC output is demonstrated       in this debug output:
    %VOICE_IEC-3-GW: Application Framework Core: Internal Error (Toll fraud call rejected):
    IEC=1.1.228.3.31.0 on callID 3 GUID=F146D6B0539C11DF800CA596C4C2D7EF
    000183: *Apr 30 14:38:57.251: //3/F146D6B0800C/CCAPI/ccCallSetContext:
       Context=0x49EC9978
    000184: *Apr 30 14:38:57.251: //3/F146D6B0800C/CCAPI/cc_process_call_setup_ind:
       >>>>CCAPI handed cid 3 with tag 1002 to app "_ManagedAppProcess_TOLLFRAUD_APP"
    000185: *Apr 30 14:38:57.251: //3/F146D6B0800C/CCAPI/ccCallDisconnect:
       Cause Value=21, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
    The Q.850 disconnect value that is returned for blocked calls can also       be changed from the default of 21 with this command:
    voice service voip
    ip address trusted call-block cause
    How to Return to Pre-15.1(2)T Behavior
    Source IP Address Trust List
    There are three ways to return to the previous behavior of voice       gateways before this trusted address toll-fraud prevention feature was       implemented. All of these configurations require that you are already running       15.1(2)T in order for you to make the configuration change.
    Explicitly enable those source IP addresses from which you would like           to add to the trusted list for legitimate VoIP calls. Up to 100 entries can be           defined. This below configuration accepts calls from those host           203.0.113.100/32, as well as from the network 192.0.2.0/24. Call setups from           all other hosts are rejected. This is the recommended method from a voice           security perspective.
    voice service voip
    ip address trusted list
      ipv4 203.0.113.100 255.255.255.255
      ipv4 192.0.2.0 255.255.255.0
    Configure the router to accept incoming call setups from all source           IP addresses.
    voice service voip
    ip address trusted list
      ipv4 0.0.0.0 0.0.0.0
    Disable the toll-fraud prevention application completely.
    voice service voip
    no ip address trusted authenticate
    Two-Stage Dialing
    If two-stage dialing is required, the following can be configured to       return behavior to match previous releases.
    For inbound ISDN calls:
    voice service pots
    no direct-inward-dial isdn
    For inbound FXO calls:
    voice-port
    secondary dialtone

  • CUCME 8.6 Call not forwarding Voicemail

    Hi frieds,
         In our office we are using  CUCME 8.6 on Cisco 2951 and unity express 8.5 in ISM module. As per our configuration  whenever user is busy or not answering , the call will forward to voicemail. Totally we have 24 PSTN line. So we have an additional gateway 2901. The Issue I’m facing is that, when a PSTN incoming call coming through the second gateway(2901), if the extension is busy or not answering the call is disconnecting instead of forwarding to voicemail.
    My 2951 configurations
    voice service voip
    ip address trusted list
    ipv4 172.16.19.80
    ipv4 172.16.19.81
    allow-connections h323 to h323
    allow-connections h323 to sip
    allow-connections sip to h323
    allow-connections sip to sip
    fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
    sip
    bind control source-interface GigabitEthernet0/1
    bind media source-interface GigabitEthernet0/1
    registrar server.
    Dial peer we are using for voice mail:
    dial-peer voice 99 voip
    destination-pattern 1099
    session protocol sipv2
    session target ipv4:172.16.19.81
    dtmf-relay sip-notify
    codec g711ulaw
    no vad.
    2901 Configurations
    voice service voip
    ip address trusted list
      ipv4 172.16.19.80
      ipv4 172.16.19.81
      ipv4 172.16.19.82
    allow-connections h323 to h323
    allow-connections h323 to sip
    allow-connections sip to h323
    allow-connections sip to sip
    fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
    dial-peer voice 99 voip
    destination-pattern 1099
    session protocol sipv2
    session target ipv4:172.16.19.81
    dtmf-relay sip-notify
    codec g711ulaw
    no vad
    ============================
    Debug CCSIP Calls
    Dec 15 15:23:23.448: //37497/8FD56BBAA9EA/SIP/Call/sipSPICallInfo:
    The Call Setup Information is:
    Call Control Block (CCB) : 0xAF40FD8
    State of The Call        : STATE_DEAD
    TCP Sockets Used         : YES
    Calling Number           : 5000
    Called Number            : 1099
    Source IP Address (Sig  ): 172.16.19.80
    Destn SIP Req Addr:Port  :
    Destn SIP Resp Addr:Port :
    Destination Name         :
    Dec 15 15:23:23.448: //37497/8FD56BBAA9EA/SIP/Call/sipSPIMediaCallInfo:
    Number of Media Streams: 1
    Media Stream             : 1
    Negotiated Codec         : No Codec
    Negotiated Codec Bytes   : 0
    Nego. Codec payload      : 255 (tx), 255 (rx)
    Negotiated Dtmf-relay    : 0
    Dtmf-relay Payload       : 0 (tx), 0 (rx)
    Source IP Address (Media): 172.16.19.80
    Source IP Port    (Media): 25364
    Destn  IP Address (Media):  -
    Destn  IP Port    (Media): 0
    Orig Destn IP Address:Port (Media): [ - ]:0
    Dec 15 15:23:23.448: //37497/8FD56BBAA9EA/SIP/Call/sipSPICallInfo:
    Disconnect Cause (CC)    : 47
    Disconnect Cause (SIP)   : 200
    For your reference I here attach a network diagram
    What the command which I missed?

    Check License status on your CUE, I had same issue.. Finally figured out its about license.. sh license status
    Sent from Cisco Technical Support iPhone App

  • The security level is set to High

    Windows 2008R2 terminal Server
    Office 2013
    Adobe Acrobat XI update 9
    When trying to create a PDF from a word document (have not tried other files yet), Adobe hangs for about 2 mins and then gives the following message
    The Security Level is set to High
    Please run the application which created this document, in the "Security Warning" dialog select the check box "Always trust macros from this source" and enable macro's created by Adobe Systems inc
    No 1. There is absolutely no need for an apostrophe in the second instance of the word macros
    Have deployed the Adobe Acrobat Administrative template and enabled the following setting
    'Automatically Trust Sites for Win OS Security Zones' (Elevates the trusted sites list in Internet Explorer to privileged locations so that they may bypass enhanced security restrictions. When enabled, the trust list is a union of IE's trust list and Acrobat's privileged locations list. GUI mapping: Edit > Preferences > Security (Enhanced) > Automatically trust sites for my Win OS security zones)
    - not a fix
    Have exported every digital signature from the pdf office dlls and imported to the computer certificate store - not a fix
    Have disabled every office macro and security setting - not a fix
    Does not matter if the file being converted is on a UNC path, mapped drive, or local drive
    Have added all file locations containing office docs to trusted folders in Word and Adobe - not a fix
    R-Click context menus for combining and conversion work fine however I understand that this uses the Adobe PDF Printer and not the office addons
    Opening a file in Word and converting to a PDF using the Addon is fine as is printing to the PDF Printer
    This issue only occurs from within the Adobe Acrobat Application 'Create file from PDF' and currently only seems to affect Office documents
    I cannot see how to give Adobe any more trust

    Solved
    I was running Office in a 'RunVirtual' environment. This man explains it best
      http://ppe.blogs.technet.com/b/gladiatormsft/archive/2014/02/05/app-v-5-on-run-virtual-rds -run-virtual-virtualizable-ext…
    Essentially Office and Acrobat are installed Natively however all Office Apps are configured to run in a Virtual environment so that Office Addins which are true AppV applications can be linked into Office.
    My 'Empty' 'RunVirtual Office package did not have 'Com Integration' enabled
    Adobe Acrobat makes use of a Com Addin for Office, so Office was unable to expose that to Adobe Acrobat until the 'Empty' 'RunVirtual Office package was updated accordingly

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