Trunking on 2924XL

I'm trying to configure a port as trunk (dot1q), but for some reason I only have the choise between two options (access or multi)
Switch (config-if)# switchport mode ?
access
multi
So I'm missing the trunk option.
IOS Image running : c2900xl-hs-mz-112.8.6-sa6.bin
any help regarding this issue will be appreciated.
Thanx in advance

According to the release not below, the dot1q is supported from 11.2(8)SA5. Can you confirm you are running this or later release ?
http://www.cisco.com/en/US/tech/tk389/tk390/technologies_configuration_example09186a008009441a.shtml
Below is the capture from the link :
The Catalyst 2900XL/3500XL switches do not support Dynamic Trunking Protocol (DTP). Use the nonegotiate option for the switchport trunk command on the other side of the trunk link. Use of the nonegotiate option prevents the receipt of DTP frames from the peer that the XL switch cannot process.
And use "show port capabilities" to enable the port supports 802.1q.
Hope this helps.

Similar Messages

  • Trunking from a 2924xl to a 3550-24PWR

    I need to seet up an access port or a trunk port on the 2924xl that will allow multiple vlans.

    The following links should help get you started
    how trunks work on 2900xl
    http://www.cisco.com/en/US/products/hw/switches/ps637/products_configuration_guide_chapter09186a00800d9d3e.html#xtocid39
    configuring a trunk 2900xl
    http://www.cisco.com/en/US/tech/tk389/tk390/technologies_configuration_example09186a00800949fd.shtml
    http://www.cisco.com/univercd/cc/td/doc/product/lan/c2900xl/29_35wc6/scg/swvlan.htm
    configuring trunks on 3550
    http://www.cisco.com/en/US/products/hw/switches/ps646/products_configuration_guide_chapter09186a00800c9fd8.html#xtocid21

  • VTP Transparent Mode in 2924XL/3524XL

    I have a 2924XL ver12.0(5)WC11 connected to two 3550 ver12.1(22)EA5 via 802.1Q trunks. They are all in the same VTP domain. The 2924XL is in VTP Client mode & the two 3550s are in VTP Server mode. There are only 6 vlans - 1, 223, 1002, 1003, 1004, & 1005. I am attempting to change all switches to VTP Transparent mode.
    Starting w/the 2924XL, when I change mode from VPT Client mode to VTP Transparent mode, the interfaces w/the trunk links immediately bounce & when they come back up, communication is only established thru VLAN1 & am no longer to communicate to devices in VLAN223.
    Connection was restored when I reconfigured the 2924XL back to VTP Client mode.
    What is causing this problem & what is the proper way to convert to Transparent mode w/o interruption of service?
    Thanks!

    Humm ...
    well, please help me to understand the problem.
    The VTP Client saves the vlan infos in RAM, and not in NVRAM like VTP Server ... when you change the VTP mode Client to Transparent, why you lose you vlan infos?
    If you do a "sh vlan" on one of your switches that is in client mode, then change it to be in transparent mode, none of them would disappear.
    It will simply stop listening to vtp messages regarding the creation and deletion of vlans.
    Then, of course, you have to create the vlan database in NVRAM ...
    Thanks for your support
    Regards
    Andrea

  • [svn:bz-trunk] 11030: Tweak the deserialization of ASObjects to treat an empty string for the type of an object as null .

    Revision: 11030
    Author:   [email protected]
    Date:     2009-10-20 11:35:02 -0700 (Tue, 20 Oct 2009)
    Log Message:
    Tweak the deserialization of ASObjects to treat an empty string for the type of an object as null. It appears that there is some logic in the LC remoting code that relies on a non-null class name to always exist. This change reverts to the old behavior of not allowing empty string as a value for the ASObject.namedType.
    This should fix bug 2448442 and its duplicates caused by the recent serialization changes.
    I don't think this is the perfect fix. Pending further investigation, a better fix would be either:
    a. If it's OK to assume that empty string should always mean null for the type of the ASObject, the code that enforces it should be in the setter/getter inside ASObject and not in the deserializer.
    b. ASObject doesn't guarantee that a named type exists or is valid. In that sense an empty string is as bad as some random characters that cannot be a valid class name in java, so depending on how disruptive it may be, the fix should be in any logic that uses ASObject.getType().
    Modified Paths:
        blazeds/trunk/modules/core/src/flex/messaging/io/amf/AbstractAmfInput.java

    Hi Pavan,
    "In your payload there is no namespace prefix for the elements under PayloadHeader element."
    Yes, you are right - but this message is standard AQ Adapter Header message - it's not defined by me. I just used message which was automatically added to my project when I have defined AQ Adapter.
    "In your process is the default namespace is same as namespace value of tns ??"
    Do you mean targetNamespace? If yes it's different as it points to process "targetNamespace="http://xmlns.oracle.com/PF_SOA_jws/PF_APPS/APPS_PROCESS" (names of application and process have changed as I try different ways to do that)
    ns1 is: xmlns:ns1="http://xmlns.oracle.com/pcbpel/adapter/aq/PF_SOA/PF_APPS/PO_AQ"
    "another thing is tns and ns1 should have same values.."
    When I create a variable of header type, namespace ns1 is automatically created for it. I set it as property of receive activity. When process is instantiated on the serwer I get the error in which you can see that namespace is tns.
    Maybe I'm doing something wrong but I don't see how I could fix this in my process.
    You can see that the message I get on the server has nothing in common with the application/project/process names. Is it possible to define such variable?
    Regards
    Pawel
    PS:
    In Transformation xsl file, both variables (source and target) has tns namespace for Header and PayloadHeader, and no namespace for subfields.
    Edited by: pawel.fidelus on 2010-01-05 02:37

  • [svn:fx-trunk] 9407: Reordering the if-statement in isMeasureFixed() exposed that hostFormat can be null when measure() is called if styles changed and measure is done before the next commitProperties .

    Revision: 9407
    Author:   [email protected]
    Date:     2009-08-19 15:11:34 -0700 (Wed, 19 Aug 2009)
    Log Message:
    Reordering the if-statement in isMeasureFixed() exposed that hostFormat can be null when measure() is called if styles changed and measure is done before the next commitProperties.  This states test exposed this.
    We should rethink if we want to clear hostFormat rather than have a hostFormatChanged flag.  If there is no hostFormat at measure then it has to be fixed because there is no line break format to check for auto-size.
    QE notes:
    Doc notes:
    Bugs: SDK-22779
    Reviewer: Gordon
    Tests run: checkintests
    Is noteworthy for integration: no
    Ticket Links:
        http://bugs.adobe.com/jira/browse/SDK-22779
    Modified Paths:
        flex/sdk/trunk/frameworks/projects/spark/src/spark/primitives/RichEditableText.as

    Oh my god, it is too long! You definitely check out types, casting and especially ODP.Net (it does everything for you)... etc. They can help you to simplify your code. I do not have enough time to copy paste it to Studio and understand and solve your issue, so I got title of your message as your main question.
    In Oracle, you can create an autonumber field by using sequences object. This is really useful when you need to create a unique number to act as a primary key.
    Basically you can create a sequence simply typing;
    CREATE SEQUENCE MY_SEQUENCE;
    now you have a sequence called "MY_SEQUENCE"... Then, I advice you select a number from sequence;
    select MY_SEQUENCE.nextval from dual;
    I said I advice actually kinda must, although it called sequence, I cannot be sequential. Do not even try to predict the value. You can be sure that it is unique number so you can use it.
    Then insert you record and use that number part of your primary key. I think that's it. Have fun.

  • Issue with instant ringback when using sip trunk to SP

    Hi all,
    We use CUCM 8.0.2.
    We have a SIP trunk to a SP connected via one of our Cisco 2911 routers configured as a CUBE.
    Cisco IOS Software, C2900 Software (C2900-UNIVERSALK9-M), Version 15.0(1)M3, RELEASE SOFTWARE (fc2)
    c2900-universalk9-mz.SPA.150-1.M3.bin
    Cisco CISCO2911/K9 (revision 1.0)
    Technology Package License Information for Module:'c2900'
    Technology Technology-package
                      Current       Type
    ipbase        ipbasek9      Permanent
    security      securityk9    Permanent
    uc              uck9            Permanent
    data           None            None
    We also have several ISDN lines that run out via various Cisco routers configured as H323 gateways.
    We use 7945 and CIPC for our phones.
    We're having an issue with calls going via the SIP trunk where we hear ringing instantly after dialling - but before the actual device at the other end starts ringing (considerable difference).
    Using the SIP trunk: If I make a call to my mobile phone - I hear ringing instantly - about 3 rings before my mobile phone actually starts ringing - undesireable.
    Using H323 gateway: If I make a call to my mobile phone - I hear silence for a bit - then ringing when the mobile starts ringing - desired.
    Using SIP trunk: If I make a call to a landline that is ready - it rings instantly for at least 1 ring - before the actual phone I'm calling starts ringing - undesireable.
    Using H323 gateway: There is a momentary pause before hearing ringing on my phone and the phone I dialled - desired.
    Using SIP trunk: If I make a call to a landline that is off-hook (with no call-waiting/etc.) - it rings once and then returns the busy signal (the worst issue) - undesireable.
    Using H323 gateway: There is a momentary pause before hearing busy signal - desired.
    Phone to phone internally (same network): Operates as expected (instantly rings locally and on the phone I'm calling). Between phones that utilise the SIP trunk and phones that utilise the H323 gateways within the same network - communication is instant and as expected.
    Any ideas why this happens and how to stop it?
    I want it to not ring until the situation is known and that it can provide the appropriate feedback (ringing/busy/etc.).
    Some possibly relevant config (note that there is a known bug with this IOS that meant I had to declare the codec in each dial-peer as the voice class would not work):
    voice service voip
    address-hiding
    mode border-element
    allow-connections sip to sip
    sip
      bind control source-interface GigabitEthernet0/0
      bind media source-interface GigabitEthernet0/0
      header-passing error-passthru
      early-offer forced
      midcall-signaling passthru
    interface GigabitEthernet0/0
    ip address x.x.x.x 255.255.255.252
    ip access-group acl.SIP-IN in
    no ip redirects
    no ip unreachables
    ip verify unicast reverse-path
    ip virtual-reassembly
    duplex full
    speed 100
    no cdp enable
    gateway
    timer receive-rtp 1200
    sip-ua
    connection-reuse
    gatekeeper
    shutdown
    dial-peer voice 1 voip
    description *** INBOUND CALLS FROM CARRIER ***
    translation-profile incoming SIPTRUNK-INCOMING
    session protocol sipv2
    incoming called-number #blah blah#
    dtmf-relay rtp-nte
    codec g711alaw
    ip qos dscp cs5 media
    no vad
    dial-peer voice 61 voip
    description **** WA, SA AND NT NUMBERS ****
    destination-pattern 0[8]........
    session protocol sipv2
    session target ipv4:<MY SP's SIP SERVER>
    incoming called-number 0[8]........
    dtmf-relay rtp-nte
    codec g711alaw
    ip qos dscp cs5 media
    no vad
    dial-peer voice 81 voip
    description **** MOBILE NUMBERS ****
    destination-pattern 0[4]........
    session protocol sipv2
    session target ipv4:<MY SP's SIP SERVER>
    incoming called-number 0[4]........
    dtmf-relay rtp-nte
    codec g711alaw
    ip qos dscp cs5 media
    no vad
    dial-peer voice 500 voip
    description *** INBOUND SIP TRUNK TO CUCM PUB ***
    translation-profile outgoing SIPTRUNK-CALLING-ADD-0
    preference 1
    destination-pattern 5[12]..
    session protocol sipv2
    session target ipv4:<OUR CUCM PUBLISHER IP>
    dtmf-relay rtp-nte
    codec g711alaw
    ip qos dscp cs5 media
    no vad
    Any help or a point in the right direction would be greatly appreciated.
    Cheers,
    Brett

    I ended up resolving this issue as follows:
    In CUCM, under Device > Device Settings > SIP Profile.
    I modifed the profile relevant to my SIP trunk, under the "Trunk Specific Configuration", I set "SIP Rel1XX Options" from "Disabled" to "Send PRACK if 1xx Contains SDP".
    Now, I get the expected delay before hearing ringback.
    Solved!

  • Trunking issue with autonomous AP

    I was at a customers that wanted a new ssid configured on his 20 1252 AP's and to stand up a new one with the olds and new SSID on it. Added the vlans, trunked them down and configured the interfaces and SSID's on the existing AP's no biggie. The new AP(not really new been kicking around in a lab) is giving me some weird issues. I deleted the exisiting config from it and took one of the running configs fromm the others that are working, modified the name and ip and pasted it in. This should have worked just fine. However the trunk port on the 4507 will not come up. The radio's are up and so is the BVI, however I cannot ping to it since the trunk is down. I started looking at the config and the thing I notice is that the AP's are in vlan 2, and the native vlan is set for vlan 2 on all the working trunk ports on the switch. However the AP's do not have vlan 2 on them, and their native vlan is 1. This should not work! I always thought the native had to match on both sides?? In any case I am probably missing something stupid, I have been up since 4am it was a 4 hour drive to get here. Oh and I have to go back and make firewall changes at midnight. Here is the relavent configs please let me know if its something stupid I usually do a lot of controller based installs havnet done a autonomous one in years. Thanks!
    Here is a port on the switch
    interface GigabitEthernet3/9
    description AP 1 - SMT
    switchport access vlan 2
    switchport trunk native vlan 2
    switchport trunk allowed vlan 2,9,15
    switchport mode trunk
    end
    Here is the AP hanging off that works fine
    cstwap01#sh vlans
    Virtual LAN ID: 1 (IEEE 802.1Q Encapsulation)
    vLAN Trunk Interfaces: Dot11Radio0
    Dot11Radio1
    GigabitEthernet0
    This is configured as native Vlan for the following interface(s) :
    Dot11Radio0
    Dot11Radio1
    GigabitEthernet0
    show run on the AP
    dot11 mbssid
    dot11 vlan-name VoIPVLan vlan 9
    dot11 vlan-name Wireless-101-Legacy vlan 15
    dot11 ssid 101
    vlan 15
    authentication open
    mbssid guest-mode
    dot11 ssid VoIPVLan
    vlan 9
    authentication open
    mbssid guest-mod
    bridge irb
    interface Dot11Radio0
    no ip address
    no ip route-cache
    ssid 101
    ssid VoIPVLan
    interface Dot11Radio0.9
    encapsulation dot1Q 9
    no ip route-cache
    bridge-group 9
    bridge-group 9 subscriber-loop-control
    bridge-group 9 block-unknown-source
    no bridge-group 9 source-learning
    no bridge-group 9 unicast-flooding
    bridge-group 9 spanning-disabled
    interface Dot11Radio0.15
    encapsulation dot1Q 15
    no ip route-cache
    bridge-group 15
    bridge-group 15 subscriber-loop-control
    bridge-group 15 block-unknown-source
    no bridge-group 15 source-learning
    no bridge-group 15 unicast-flooding
    bridge-group 15 spanning-disabled
    interface GigabitEthernet0
    no ip address
    no ip route-cache
    duplex auto
    speed auto
    bridge-group 1
    no bridge-group 1 source-learning
    bridge-group 1 spanning-disabled
    interface GigabitEthernet0.9
    encapsulation dot1Q 9
    no ip route-cache
    bridge-group 9
    no bridge-group 9 source-learning
    bridge-group 9 spanning-disabled
    interface GigabitEthernet0.15
    encapsulation dot1Q 15
    no ip route-cache
    bridge-group 15
    no bridge-group 15 source-learning
    bridge-group 15 spanning-disabled
    interface BVI1
    ip address 10.190.2.51 255.255.255.0
    no ip route-cache
    ip default-gateway 10.190.2.1

    Hello Mike,
    actually, native VLAN doesn't have to match on two sides of a trunk, in fact it's quite common with autonomous APs. Your BVI1 is usually tied to a native VLAN 1, but switch port native VLAN is usually configured for whatever VLAN you want AP's management IP to belong to, in your case VLAN 2.
    When you connect two catalyst switches together with a trunk that has different native VLAN configured on two sides, you will keep getting logging messages warning you about it. It's not recommended, but some scenarios require such setup.
    Something else is wrong, maybe a physical issue. When you say the trunk is down, do you mean the interface is down ? Can you paste config of bad switch port and bad AP?
    Regards,
    Roman

  • 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.

  • ASA 5505 switch from Mode: access Port to Trunk on the fly via CSM ?

                       Hi
    Can I configure the Port at the ASA 5050 from Mode: access Port to trunk during the FW is running in a production area without console access ?
    As I know at the 5505 ist should work
    sincerley
    Alfred

    Are you using the same access port to access the ASA via CSM? If you are, then i would strongly recommend console access.
    If you are using a different port to make the changes, then yes you can configure it. However, I would still recommend configuring it during non production hour, just in case there is an issue.

  • ASA 5505 Trunking problem with cisco 3760

    Here is my asa 5505 configuration  and at bottom i am pasting cisco 3760 configuration, I am having trunking problem and seen other people having the same issue but no solution.
    Problem: when i connect any device on Switch 3760 port 1 which is trunked i cannot communicate with niether vlan.
    I have tried device with both static vlan configuration
    192.168.1.99 (vlan 2)
    172.168.1.99 (vlan 8)
    if i change port 1 to access mode or define native vlan it work with one vlan.
    interface 5 of Cisco ASA is connected with interface 25 (uplink) on  Cisco Switch
    interface Ethernet0/5
    switchport trunk allowed vlan 1-10
    switchport mode trunk
    interface GigabitEthernet1/0/25
      description **UPLINK**
    switchport trunk encapsulation dot1q
    switchport trunk allowed vlan 1-10
    switchport mode trunk
    On port 1 of cisco Switch i am creating trunk
    interface GigabitEthernet1/0/1
    description **LAN**
    switchport trunk encapsulation dot1q
    switchport trunk allowed vlan 1-10
    switchport mode trunk
    no logging event link-status
    spanning-tree portfast
    CISCO ASA Config i have configure the following vlans
    interface Vlan2
    nameif inside
    security-level 100
    ip address 192.168.1.1 255.255.255.0
    interface Vlan8
    description Server VLAN
    nameif Internal_LAN
    security-level 90
    ip address 172.168.1.1 255.255.255.0
    interface Ethernet0/0
    switchport access vlan 2
    on Cisco 3760 i have configure the following vlan.
    v
    vlan 2
    name inside
    vlan 8
    name Internal_LAN
    interface Null0
    no ip unreachables
    Message was edited by: macboy

    Hello,
    I can see many people have visited the post but there isn't any reply.
    I think this is because of the following:
    The configurations are long and most people dont want to read it. Try adding only relevant configuration to the issue.
    The problem description is not clear. You mention trunking problem but what exactly is the problem? Can you ping from switch to ASA? Is any Vlan information being passed or they are all failing. Try to be as much specific as possible.
    Which port of the switch connects to the ASA?
    Using portfast on a trunk (interface GigabitEthernet1/0/1) is not recommended.
    Regards,
    Felipe.
    Remember to rate useful posts.

  • Null Trunk in Termination Call Detail in HDS

    We are running CVP and ICM version 8.  On the HDS Termination Call Detail table, the Trunk field is showing Null.
    In the old version 7.2, the Trunk field showed values.  This field is used by our reporting application (Exony VIM 8).  It basically shows the agents extension on the report.
    I have TAC case open.  The engineer initially told me that the field is showing values in his test lab running versions 8 and 8.5.  Today he tells me that the Trunk field should show Null values in both versions.
    Is this by design or a bug?
    I've included the SQL queries from both 7.2 and 8.0 HDS in spreadsheet format.
    Any comment is appreciated.  Thanks.  

    Hi Raj,
    I am getting this anonymous even for Internal calls. Please find the below screen shot
    Thanks,
    Siva

  • Various questions on uplink profiles, CoS, native VLAN, downlink trunking

    I will be using vPC End Host Mode with MAC-pinning. I see I can further configure MAC-Pinning. Is this required or will it automatically forward packets by just turning it on? Is it also best not to enable failover for the vnics in this configuration? See this text from the Cisco 1000V deployment Guide:
    Fabric Fail-Over Mode
    Within the Cisco UCS M71KR-E, M71KR-Q and M81KR adapter types, the Cisco Unified Computing System can
    enable a fabric failover capability in which loss of connectivity on a path in use will cause remapping of traffic
    through a redundant path within the Cisco Unified Computing System. It is recommended to allow the Cisco Nexus
    1000V redundancy mechanism to provide the redundancy and not to enable fabric fail-over when creating the
    network interfaces within the UCS Service Profiles. Figure 3 shows the dialog box. Make sure the Enable Failover
    checkbox is not checked."
    What is the 1000V redundancy?? I didn't know it has redundancy. Is it the MAC-Pinning set up in the 1000V? Is it Network State Tracking?
    The 1000V has redundancy and we can even pin VLANs to whatever vNIC we want. See Cisco's Best Practices for Nexus 1000V and UCS.
    Nexus1000V management VLAN. Can I use the same VLAN for this and for ESX-management and for Switch management? E.g VLan 3 for everything.
    According to the below text (1000V Deployment Guide), I can have them all in the same vlan:
    There are no best practices that specify whether the VSM
    and the VMware ESX management interface should be on the same VLAN. If the management VLAN for
    network devices is a different VLAN than that used for server management, the VSM management
    interface should be on the management VLAN used for the network devices. Otherwise, the VSM and the
    VMware ESX management interfaces should share the same VLAN.
    I will also be using CoS and Qos to prioritize the traffic. The CoS can either be set in the 1000V (Host control Full) or per virtual adapter (Host control none) in UCS. Since I don't know how to configure CoS on the 1000V, I wonder if I can just set it in UCS (per adapter) as before when using the 1000V, ie. we have 2 choices.
    Yes, you can still manage CoS using QoS on the vnics when using 1000V:
    The recommended action in the Cisco Nexus 1000V Series is to assign a class of service (CoS) of 6 to the VMware service console and VMkernel flows and to honor these QoS markings on the data center switch to which the Cisco UCS 6100 Series Fabric Interconnect connects. Marking of QoS values can be performed on the Cisco Nexus 1000V Series Switch in all cases, or it can be performed on a per-VIF basis on the Cisco UCS M81KR or P81E within the Cisco Unified Computing System with or without the Cisco Nexus 1000V Series Switch.
    Something else: Native VLANs
    Is it important to have the same native VLAN on the UCS and the Cisco switch? And not to use the default native VLAN 1?   I read somewhere that the native VLAN is used for communication between the switches and CDP amongst others. I know the native VLAN is for all untagged traffic. I see many people set the ESXi management VLAN as native also, and in the above article the native VLAN (default 1) is setup. Why? I have been advised to leave out the native VLAN.
    Example:Will I be able to access a VM set with VLAN 0 (native) if the native VLAN is the same in UCS and the Cisco switch (Eg. VLAN 2)? Can I just configure a access port with the same VLAN ID as the native VLAN, i.e 2 and connect to it with a PC using the same IP network address?
    And is it important to trunk this native VLAN? I see in a Netapp Flexpod config they state this: "This configuration also leverages the native VLAN on the trunk ports to discard untagged packets, by setting the native VLAN on the port channel, but not including this VLAN in the allowed VLANs on the port channel". But I don't understand it...
    What about the downlinks from the FI to the chassis. Do you configure this as a port channel also in UCS? Or is this not possible with the setup described here with 1000V and MAC-pinning.
    No, port channel should not be configured when MAC-pinning is configured.
    [Robert] The VSM doesn't participate in STP so it will never send BPDU's.  However, since VMs can act like bridges & routers these days, we advise to add two commands to your upstream VEM uplinks - PortFast and BPDUFilter.  PortFast so the interface is FWD faster (since there's no STP on the VSM anyway) and BPDUFilter to ignore any received BPDU's from VMs.  I prefer to ignore them then using BPDU Gaurd - which will shutdown the interface if BPDU's are received.
    -Are you thinking of the upstream switch here (Nexus, Catalyst) or the N1kV uplink profile config?
    Edit: 26 July 14:23. Found answers to many of my many questions...

    Answers inline.
    Atle Dale wrote:
    Something else: Native VLANsIs it important to have the same native VLAN on the UCS and the Cisco switch? And not to use the default native VLAN 1?   I read somewhere that the native VLAN is used for communication between the switches and CDP amongst others. I know the native VLAN is for all untagged traffic. I see many people set the ESXi management VLAN as native also, and in the above article the native VLAN (default 1) is setup. Why? I have been advised to leave out the native VLAN.[Robert] The native VLAN is assigned per hop.  This means between the 1000v Uplinks port profile and your UCS vNIC definition, the native VLAN should be the same.  If you're not using a native VLAN, the "default" VLAN will be used for control traffic communication.  The native VLAN and default VLAN are not necessarily the same.  Native refers to VLAN traffic without an 802.1q header and can be assigned or not.  A default VLAN is mandatory.  This happens to start as VLAN 1 in UCS but can be changed. The default VLAN will be used for control traffic communication.  If you look at any switch (including the 1000v or Fabric Interconnects) and do a "show int trunk" from the NXOS CLI, you'll see there's always one VLAN allowed on every interface (by default VLAN 1) - This is your default VLAN.Example:Will I be able to access a VM set with VLAN 0 (native) if the native VLAN is the same in UCS and the Cisco switch (Eg. VLAN 2)? Can I just configure a access port with the same VLAN ID as the native VLAN, i.e 2 and connect to it with a PC using the same IP network address?[Robert] There's no VLAN 0.  An access port doesn't use a native VLAN - as its assigned to only to a single VLAN.  A trunk on the other hand carries multiple VLANs and can have a native vlan assigned.  Remember your native vlan usage must be matched between each hop.  Most network admins setup the native vlan to be the same throughout their network for simplicity.  In your example, you wouldn't set your VM's port profile to be in VLAN 0 (doens't exist), but rather VLAN 2 as an access port.  If VLAN 2 also happens to be your Native VLAN northbound of UCS, then you would configured VLAN 2 as the Native VLAN on your UCS ethernet uplinks.  On switch northbound of the UCS Interconnects you'll want to ensure on the receiving trunk interface VLAN 2 is set as the native vlan also.  Summary:1000v - VM vEthernet port profile set as access port VLAN 21000v - Ethernet Uplink Port profile set as trunk with Native VLAN 2UCS - vNIC in Service Profile allowing all required VLANs, and VLAN 2 set as NativeUCS - Uplink Interface(s) or Port Channel set as trunk with VLAN 2 as Native VLANUpstream Switch from UCS - Set as trunk interface with Native VLAN 2From this example, your VM will be reachable on VLAN 2 from any device - assuming you have L3/routing configured correctly also.And is it important to trunk this native VLAN? I see in a Netapp Flexpod config they state this: "This configuration also leverages the native VLAN on the trunk ports to discard untagged packets, by setting the native VLAN on the port channel, but not including this VLAN in the allowed VLANs on the port channel". But I don't understand it...[Robert] This statement recommends "not" to use a native VLAN.  This is a practice by some people.  Rather than using a native VLAN throughout their network, they tag everything.  This doesn't change the operation or reachability of any VLAN or device - it's simply a design descision.  The reason some people opt not to use a native VLAN is that almost all switches use VLAN 1 as the native by default.  So if you're using the native VLAN 1 for management access to all your devices, and someone connects in (without your knowing) another switch and simply plug into it - they'd land on the same VLAN as your management devices and potentially do harm.What about the downlinks from the FI to the chassis. Do you configure this as a port channel also in UCS? Or is this not possible with the setup descrived here with 1000V and MAC-pinning.[Robert] On the first generation hardware (6100 FI and 2104 IOM) port channeling is not possible.  With the latest HW (6200 and 2200) you can create port channels with all the IOM - FI server links.  This is not configurable.  You either tell the system to use Port Channel or Individual Links.  The major bonus of using a Port Channel is losing a link doesn't impact any pinned interfaces - as it would with individual server interfaces.  To fix a failed link when configured as "Individual" you must re-ack the Chassis to re-pinn the virtual interfaces to the remaining server uplinks.  In regards to 1000v uplinks - the only supported port channeling method is "Mac Pinning".  This is because you can't port channel physical interfaces going to separate Fabrics (one to A and one to B).  Mac Pinning gets around this by using pinning so all uplinks can be utilized at the same time.--[Robert] The VSM doesn't participate in STP so it will never send BPDU's.  However, since VMs can act like bridges & routers these days, we advise to add two commands to your upstream VEM uplinks - PortFast and BPDUFilter.  PortFast so the interface is FWD faster (since there's no STP on the VSM anyway) and BPDUFilter to ignore any received BPDU's from VMs.  I prefer to ignore them then using BPDU Gaurd - which will shutdown the interface if BPDU's are received.-Are you thinking of the upstream switch here (Nexus, Catalyst) or the N1kV uplink profile config?[Robert] The two STP commands would be used only when the VEM (ESX host) is directly connected to an upstream switch.  For UCS these two commands to NOT apply.

  • Unable to place call on calls on hold - SIP Trunk from CUCM to CUBE and from CUBE to ISTP

    Hi Cisco Community,
    I have a SIP Trunk setup between the CUCM and CUBE and another SIP Dial Peers from the CUBE to the ITSP. All incoming/outgoing calls, DTMF-Relay works well except one thing which is the ability to hold the call.
    On the SIP Trunk from the CUCM to the CUBE, I did not select “MTP” because when I do so, I am forced to select my preferred MTP codec which when selected G.729/G.729a, all my outgoing calls goes out using G.729r8. This codec works well for most of the calls until the ITSP replies back with G.729br8. When this condition occur, my call simply fails (this is very intermittent and only some random numbers).
    That said, I have some issues with DTMF Relay when I select MTP on the SIP Trunk. DTMF Relay only works if the call is G.729r8 all the way from the CUCM to the ITSP. If the ITSP replies back with G.729br8, the call might established but will simply be “voice-only”.
    The current setup is no MTP is selected and everything is working perfectly. I am happy with that until I place a call on hold, which when I do so, the call immediately terminate. Could you please help me understand why?
    I have all media resources configured such as G.729r8 MTP, G.729br8 MTP, G.711u MTP, Transcoding with all codecs, etc. All MRG and MRGL are configured on all devices and SIP Trunks.
    Below is an example of a call that is connected with the current setup:
    Note:
    IP: 10.18.81.2 (CUBE)
    IP: 10.18.81.11 (CUCM SUB)
    IP: 10.111.111.254 (ITSP SBC)
    PM-HO-VG-01#
    PM-HO-VG-01#
    Nov 30 11:44:29.938: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received: 
    INVITE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72063a5aba5d
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>
    Date: Sun, 30 Nov 2014 11:44:29 GMT
    Call-ID: [email protected]
    Supported: timer,resource-priority,replaces
    Min-SE:  1800
    User-Agent: Cisco-CUCM9.1
    Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
    CSeq: 101 INVITE
    Expires: 180
    Allow-Events: presence, kpml
    Supported: X-cisco-srtp-fallback,X-cisco-original-called
    Call-Info: <sip:10.18.81.11:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
    Cisco-Guid: 1020645888-0000065536-0000124117-0189862410
    Session-Expires:  1800
    Contact: <sip:[email protected]:5060>
    Max-Forwards: 70
    Content-Length: 0
    Nov 30 11:44:29.942: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72063a5aba5d
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>
    Date: Sun, 30 Nov 2014 11:44:29 GMT
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Allow-Events: telephone-event
    Server: Cisco-SIPGateway/IOS-15.2.4.M5
    Content-Length: 0
    Nov 30 11:44:29.946: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    INVITE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>
    Date: Sun, 30 Nov 2014 11:44:29 GMT
    Call-ID: [email protected]
    Supported: 100rel,timer,resource-priority,replaces,sdp-anat
    Min-SE:  1800
    Cisco-Guid: 1020645888-0000065536-0000124117-0189862410
    User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    CSeq: 101 INVITE
    Timestamp: 1417347869
    Contact: <sip:[email protected]:5060>
    Call-Info: <sip:10.18.81.2:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
    Expires: 180
    Allow-Events: telephone-event
    Max-Forwards: 69
    Session-Expires:  1800
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 301
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 7676 6958 IN IP4 10.18.81.2
    s=SIP Call
    c=I
    PM-HO-VG-01#N IP4 10.18.81.2
    t=0 0
    m=audio 22256 RTP/AVP 18 0 8 101
    c=IN IP4 10.18.81.2
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=yes
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    Nov 30 11:44:29.950: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Received: 
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Timestamp: 1417347869
    Nov 30 11:44:30.658: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Received: 
    SIP/2.0 180 Session Progress
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>;tag=71913148-1417348035284
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Timestamp: 1417347869
    Supported: 
    Contact: <sip:[email protected]:5060;transport=udp>
    Session: Media
    Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
    X-BroadWorks-Correlation-Info: bbf94839-a234-4237-95e6-a7037322f0f4
    Content-Type: application/sdp
    Content-Length: 355
    v=0
    o=BroadWorks 316169737 1 IN IP4 10.111.111.254
    s=-
    c=IN IP4 10.111.111.254
    t=0 0
    m=audio 20074 RTP/AVP 18 101 100
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=rtpmap:100 X-NSE/8000
    a=fmtp:100 200-202
    a=X-sqn:0
    a=X-cap: 1 audio RTP/AVP 100
    a=X-cpar: a=rtpmap:100 X-NSE/8000
    a=X-cpar: a=fmtp:100 200-202
    a=X-cap: 2 image udptl t38
    Nov 30 11:44:30.662: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    SIP/2.0 183 Session Progress
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72063a5aba5d
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>;tag=3C365010-1E42
    Date: Sun, 30 Nov 2014 11:44:29 GMT
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    Allow-Events: telephone-event
    Contact: <sip:[email protected]:5060>
    Supported: sdp-anat
    Server: Cisco-SIPGateway/IOS-15.2.4.M5
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 289
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 7965 2747 IN IP4 10.18.81.2
    s=SIP Call
    c=IN IP4 10.18.81.2
    t=0 0
    m=audio 22350 RTP/AVP 18 101 19
    c=IN IP4 10.18.81.2
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=yes
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=rtpmap:19 CN/8000
    a=ptime:20
    Nov 30 11:44:31.226: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Received: 
    SIP/2.0 180 Session Progress
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>;tag=71913148-1417348035284
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Timestamp: 1417347869
    Supported: 
    Contact: <sip:[email protected]:5060;transport=udp>
    Session: Media
    Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
    X-BroadWorks-Correlation-Info: bbf9
    PM-HO-VG-01#4839-a234-4237-95e6-a7037322f0f4
    Content-Type: application/sdp
    Content-Length: 355
    v=0
    o=BroadWorks 316169737 1 IN IP4 10.111.111.254
    s=-
    c=IN IP4 10.111.111.254
    t=0 0
    m=audio 20074 RTP/AVP 18 101 100
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=rtpmap:100 X-NSE/8000
    a=fmtp:100 200-202
    a=X-sqn:0
    a=X-cap: 1 audio RTP/AVP 100
    a=X-cpar: a=rtpmap:100 X-NSE/8000
    a=X-cpar: a=fmtp:100 200-202
    a=X-cap: 2 image udptl t38
    Nov 30 11:44:31.630: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Received: 
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>;tag=71913148-1417348035284
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Timestamp: 1417347869
    Supported: 
    Contact: <sip:[email protected]:5060;transport=udp>
    Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
    Accept: application/media_control+xml,application/sdp,application/xml
    X-BroadWorks-Correlation-Info: bbf94839-a234-4237-95e6-a7037322f0f4
    Content-Type: application/sdp
    Content-Length: 355
    v=0
    o=BroadWorks 316169737 1 IN IP4 10.111.111.254
    s=-
    c=IN IP4 10.111.111.254
    t=0 0
    m=audio 20074 RTP/AVP 18 101 100
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=rtpmap:100 X-NSE/8000
    a=fmtp:100 200-202
    a=X-sqn:0
    a=X-cap: 1 audio RTP/AVP 100
    a=X-cpar: a=rtpmap:100 X-NSE/8000
    a=X-cpar: a=fmtp:100 200-202
    a=X-cap: 2 image udptl t38
    Nov 30 11:44:31.630: //64511/3CD5D2000001/SIP/Call/sipSPICallInfo: 
    The Call Setup Information is:
    Call Control Block (CCB) : 0x0x3D7B1458
    State of The Call        : STATE_ACTIVE
    TCP Sockets Used         : NO
    Calling Number           : 27218091323
    Called Number            : 0862000000
    Source IP Address (Sig  ): 10.18.81.2
    Destn SIP Req Addr:Port  : 10.111.111.254:5060
    Destn SIP Resp Addr:Port : 10.111.111.254:5060
    Destination Name         : 10.111.111.254
    Nov 30 11:44:31.630: //64511/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo: 
    Number of Media Streams: 1
    Media Stream             : 1
    Negotiated Codec         : g729br8
    Negotiated Codec Bytes   : 20
    Nego. Codec payload      : 18 (tx), 18 (rx)
    Negotiated Dtmf-relay    : 6
    Dtmf-relay Payload       : 101 (tx), 101 (rx)
    Source IP Address (Media): 10.18.81.2
    Source IP Port    (Media): 22256
    Destn  IP Address (Media): 10.111.111.254
    Destn  IP Port    (Media): 20074
    Orig Destn IP Address:Port (Media): [ - ]:0
    Nov 30 11:44:31.630: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    ACK sip:[email protected]:5060;transport=udp SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC81D00
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>;tag=71913148-1417348035284
    Date: Sun, 30 Nov 2014 11:44:29 GMT
    Call-ID: [email protected]
    Max-Forwards: 70
    CSeq: 101 ACK
    Allow-Events: telephone-event
    Content-Length: 0
    Nov 30 11:44:31.634: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72063a5aba5d
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>;tag=3C365010-1E42
    Date: Sun, 30 Nov 2014 11:44:29 GMT
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    Allow-Events: telephone-event
    Contact: <sip:[email protected]:5060>
    Supported: replaces
    Supported: sdp-anat
    Server: Cisco-SIPGateway/IOS-15.2.4.M5
    Supported: timer
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 289
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 7965 2747 IN IP4 10.18.81.2
    s=SIP Call
    c=IN IP4 10.18.81.2
    t=0 0
    m=audio 22350 RTP/AVP 18 101 19
    c=IN IP4 10.18.81.2
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=yes
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=rtpmap:19 CN/8000
    a=ptime:20
    Nov 30 11:44:31.726: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received: 
    ACK sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72075e3a02c1
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>;tag=3C365010-1E42
    Date: Sun, 30 Nov 2014 11:44:29 GMT
    Call-ID: [email protected]
    Max-Forwards: 70
    CSeq: 101 ACK
    Allow-Events: presence, kpml
    Content-Type: application/sdp
    Content-Length: 236
    v=0
    o=CiscoSystemsCCM-SIP 9082578 1 IN IP4 10.18.81.11
    s=SIP Call
    c=IN IP4 10.18.80.40
    b=TIAS:8000
    b=AS:8
    t=0 0
    m=audio 21928 RTP/AVP 18 101
    a=rtpmap:18 G729/8000
    a=ptime:20
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    Nov 30 11:44:31.730: //64510/3CD5D2000001/SIP/Call/sipSPICallInfo: 
    The Call Setup Information is:
    Call Control Block (CCB) : 0x0x3D816D70
    State of The Call        : STATE_ACTIVE
    TCP Sockets Used         : NO
    Calling Number           : 0218091323
    Called Number            : 0862000000
    Source IP Address (Sig  ): 10.18.81.2
    Destn SIP Req Addr:Port  : 10.18.81.11:5060
    Destn SIP Resp Addr:Port : 10.18.81.11:5060
    Destination Name         : 10.18.81.11
    Nov 30 11:44:31.730: //64510/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo: 
    Number of Media Streams: 1
    Media Stream             : 1
    Negotiated Codec         : g729br8
    Negotiated Codec Bytes   : 20
    Nego. Codec payload      : 18 (tx), 18 (rx)
    Negotiated Dtmf-relay    : 6
    Dtmf-relay Payload       : 101 (tx), 101 (rx)
    Source IP Address (Media): 10.18.81.2
    Source IP Port    (Media): 22350
    Destn  IP Address (Media): 10.18.80.40
    Destn  IP Port    (Media): 21928
    Orig Destn IP Address:Port (Media): [ - ]:0
    Nov 30 11:44:31.730: //64510/3CD5D2000001/SIP/Call/sipSPICallInfo: 
    The Call Setup Information is:
    Call Control Block (CCB) : 0x0x3D816D70
    State of The Call        : STATE_ACTIVE
    TCP Sockets Used         : NO
    Calling Number           : 0218091323
    Called Number            : 0862000000
    Source IP Address (Sig  ): 10.18.81.2
    Destn SIP Req Addr:Port  : 10.18.81.11:5060
    Destn SIP Resp Addr:Port : 10.18.81.11:5060
    Destination Name         : 10.18.81.11
    PM-HO-VG-01#
    Nov 30 11:44:31.730: //64510/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo: 
    Number of Media Streams: 1
    Media Stream             : 1
    Negotiated Codec         : g729br8
    Negotiated Codec Bytes   : 20
    Nego. Codec payload      : 18 (tx), 18 (rx)
    Negotiated Dtmf-relay    : 6
    Dtmf-relay Payload       : 101 (tx), 101 (rx)
    Source IP Address (Media): 10.18.81.2
    Source IP Port    (Media): 22350
    Destn  IP Address (Media): 10.18.80.40
    Destn  IP Port    (Media): 21928
    Orig Destn IP Address:Port (Media): [ - ]:0
    PM-HO-VG-01#sh sip
    PM-HO-VG-01#sh sip-ua call
    PM-HO-VG-01#sh sip-ua calls 
    Total SIP call legs:2, User Agent Client:1, User Agent Server:1
    SIP UAC CALL INFO
    Call 1
    SIP Call ID                : [email protected]
       State of the call       : STATE_ACTIVE (7)
       Substate of the call    : SUBSTATE_NONE (0)
       Calling Number          : 27218091323
       Called Number           : 0862000000
       Bit Flags               : 0xC04018 0x10000100 0x0
       CC Call ID              : 64511
       Source IP Address (Sig ): 10.18.81.2
       Destn SIP Req Addr:Port : [10.111.111.254]:5060
       Destn SIP Resp Addr:Port: [10.111.111.254]:5060
       Destination Name        : 10.111.111.254
       Number of Media Streams : 1
       Number of Active Streams: 1
       RTP Fork Object         : 0x0
       Media Mode              : flow-through
       Media Stream 1
         State of the stream      : STREAM_ACTIVE
         Stream Call ID           : 64511
         Stream Type              : voice+dtmf (0)
         Stream Media Addr Type   : 1
         Negotiated Codec         : g729br8 (20 bytes)
         Codec Payload Type       : 18 
         Negotiated Dtmf-relay    : rtp-nte
         Dtmf-relay Payload Type  : 101
         QoS ID                   : -1
         Local QoS Strength       : BestEffort
         Negotiated QoS Strength  : BestEffort
         Negotiated QoS Direction : None
         Local QoS Status         : None
         Media Source IP Addr:Port: [10.18.81.2]:22256
         Media Dest IP Addr:Port  : [10.111.111.254]:20074
    Options-Ping    ENABLED:NO    ACTIVE:NO
       Number of SIP User Agent Client(UAC) calls: 1
    SIP UAS CALL INFO
    Call 1
    SIP Call ID                : [email protected]
       State of the call       : STATE_ACTIVE (7)
       Substate of the call    : SUBSTATE_NONE (0)
       Calling Number          : 0218091323
       Called Number           : 0862000000
       Bit Flags               : 0xC0401E 0x10000100 0x80004
       CC Call ID              : 64510
       Source IP Address (Sig ): 10.18.81.2
       Destn SIP Req Addr:Port : [10.18.81.11]:5060
       Destn SIP Resp Addr:Port: [10.18.81.11]:5060
       Destination Name        : 10.18.81.11
       Number of Media Streams : 1
       Number of Active Streams: 1
       RTP Fork Object         : 0x0
       Media Mode              : flow-through
       Media Stream 1
         State of the stream      : STREAM_ACTIVE
         Stream Call ID           : 64510
         Stream Type              : voice+dtmf (1)
         Stream Media Addr Type   : 1
         Negotiated Codec         : g729br8 (20 bytes)
         Codec Payload Type       : 18 
         Negotiated Dtmf-relay    : rtp-nte
         Dtmf-relay Payload Type  : 101
         QoS ID                   : -1
         Local QoS Strength       : BestEffort
         Negotiated QoS Strength  : BestEffort
         Negotiated QoS Direction : None
         Local QoS Status         : None
         Media Source IP Addr:Port: [10.18.81.2]:22350
         Media Dest IP Addr:Port  : [10.18.80.40]:21928
    Options-Ping    ENABLED:NO    ACTIVE:NO
       Number of SIP User Agent Server(UAS) calls: 1
    PM-HO-VG-01#
    PM-HO-VG-01#
    PM-HO-VG-01#
    As you can see, the call is connected and everything is working perfectly. When I press the hold button, here is what I get:
    NOTE: I have # debug ccsip messages and #debug ccsip calls (running)
    PM-HO-VG-01#
    PM-HO-VG-01#
    Nov 30 11:44:49.210: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received: 
    INVITE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720852ab8b92
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>;tag=3C365010-1E42
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    Supported: timer,resource-priority,replaces
    Min-SE:  1800
    User-Agent: Cisco-CUCM9.1
    Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
    CSeq: 102 INVITE
    Max-Forwards: 70
    Expires: 180
    Allow-Events: presence
    Supported: X-cisco-srtp-fallback
    Supported: Geolocation
    Contact: <sip:[email protected]:5060>
    Content-Type: application/sdp
    Content-Length: 244
    v=0
    o=CiscoSystemsCCM-SIP 9082578 2 IN IP4 10.18.81.11
    s=SIP Call
    c=IN IP4 0.0.0.0
    b=TIAS:8000
    b=AS:8
    t=0 0
    m=audio 21928 RTP/AVP 18 101
    a=rtpmap:18 G729/8000
    a=ptime:20
    a=inactive
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    Nov 30 11:44:49.218: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720852ab8b92
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>;tag=3C365010-1E42
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    CSeq: 102 INVITE
    Allow-Events: telephone-event
    Server: Cisco-SIPGateway/IOS-15.2.4.M5
    Content-Length: 0
    Nov 30 11:44:49.218: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    INVITE sip:[email protected]:5060;transport=udp SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC9241
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>;tag=71913148-1417348035284
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    Supported: 100rel,timer,resource-priority,replaces,sdp-anat
    Min-SE:  1800
    Cisco-Guid: 1020645888-0000065536-0000124117-0189862410
    User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    CSeq: 102 INVITE
    Max-Forwards: 70
    Timestamp: 1417347889
    Contact: <sip:[email protected]:5060>
    Call-Info: <sip:10.18.81.2:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
    Expires: 180
    Allow-Events: telephone-event
    Content-Type: application/sdp
    Content-Length: 271
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 7676 6959 IN IP4 10.18.81.2
    s=SIP Call
    c=IN IP4 0.0.0.0
    t=0 0
    m=audio 22256 RTP/AVP 18 101
    c=IN IP4 0.0.0.0
    a=inactive
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=yes
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=ptime:20
    Nov 30 11:44:49.278: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Received: 
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC9241
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>;tag=71913148-1417348035284
    Call-ID: [email protected]
    CSeq: 102 INVITE
    Timestamp: 1417347889
    Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
    Supported: 
    Accept: application/media_control+xml,application/sdp,application/xml
    Contact: <sip:[email protected]:5060;transport=udp>
    X-BroadWorks-Correlation-Info: bbf94839-a234-4237-95e6-a7037322f0f4
    Content-Type: application/sdp
    Content-Length: 360
    v=0
    o=BroadWorks 316169737 2 IN IP4 10.111.111.254
    s=-
    c=IN IP4 0.0.0.0
    t=0 0
    m=audio 20074 RTP/AVP 18 101 100
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=rtpmap:100 X-NSE/8000
    a=fmtp:100 200-202
    a=X-sqn:0
    a=X-cap: 1 audio RTP/AVP 100
    a=X-cpar: a=rtpmap:100 X-NSE/8000
    a=X-cpar: a=fmtp:100 200-202
    a=X-cap: 2 image udptl t38
    a=inactive
    Nov 30 11:44:49.278: //64511/3CD5D2000001/SIP/Call/sipSPICallInfo: 
    The Call Setup Information is:
    Call Control Block (CCB) : 0x0x3D7B1458
    State of The Call        : STATE_ACTIVE
    TCP Sockets Used         : NO
    Calling Number           : 27218091323
    Called Number            : 0862000000
    Source IP Address (Sig  ): 10.18.81.2
    Destn SIP Req Addr:Port  : 10.111.111.254:5060
    Destn SIP Resp Addr:Port : 10.111.111.254:5060
    Destination Name         : 10.111.111.254
    Nov 30 11:44:49.278: //64511/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo: 
    Number of Media Streams: 1
    Media Stream             : 1
    Negotiated Codec         : g729br8
    Negotiated Codec Bytes   : 20
    Nego. Codec payload      : 18 (tx), 18 (rx)
    Negotiated Dtmf-relay    : 6
    Dtmf-relay Payload       : 101 (tx), 101 (rx)
    Source IP Address (Media): 10.18.81.2
    Source IP Port    (Media): 22256
    Destn  IP Address (Media): 0.0.0.0
    Destn  IP Port    (Media): 20074
    Orig Destn IP Address:Port (Media): [ - ]:0
    Nov 30 11:44:49.282: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    ACK sip:[email protected]:5060;transport=udp SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECA2633
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>;tag=71913148-1417348035284
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    Max-Forwards: 70
    CSeq: 102 ACK
    Allow-Events: telephone-event
    Content-Length: 0
    Nov 30 11:44:49.282: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720852ab8b92
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>;tag=3C365010-1E42
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    CSeq: 102 INVITE
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    Allow-Events: telephone-event
    Contact: <sip:[email protected]:5060>
    Supported: replaces
    Supported: sdp-anat
    Server: Cisco-SIPGateway/IOS-15.2.4.M5
    Supported: timer
    Content-Type: application/sdp
    Content-Length: 271
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 7965 2748 IN IP4 10.18.81.2
    s=SIP Call
    c=IN IP4 0.0.0.0
    t=0 0
    m=audio 22350 RTP/AVP 18 101
    c=IN IP4 0.0.0.0
    a=inactive
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=yes
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=ptime:20
    Nov 30 11:44:49.282: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received: 
    ACK sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72094953dfea
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>;tag=3C365010-1E42
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    Max-Forwards: 70
    CSeq: 102 ACK
    Allow-Events: presence
    Content-Length: 0
    Nov 30 11:44:49.290: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received: 
    INVITE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720a6918040f
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>;tag=3C365010-1E42
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    Supported: timer,resource-priority,replaces
    Min-SE:  1800
    User-Agent: Cisco-CUCM9.1
    Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
    CSeq: 103 INVITE
    Max-Forwards: 70
    Expires: 180
    Allow-Events: presence
    Supported: X-cisco-srtp-fallback
    Supported: Geolocation
    Contact: <sip:[email protected]:5060>
    Content-Length: 0
    Nov 30 11:44:49.294: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720a6918040f
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>;tag=3C365010-1E42
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    CSeq: 103 INVITE
    Allow-Events: telephone-event
    Server: Cisco-SIPGateway/IOS-15.2.4.M5
    Content-Length: 0
    Nov 30 11:44:49.294: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    INVITE sip:[email protected]:5060;transport=udp SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECB16F3
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>;tag=71913148-1417348035284
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    Supported: timer,resource-priority,replaces,sdp-anat
    Min-SE:  1800
    Cisco-Guid: 1020645888-0000065536-0000124117-0189862410
    User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    CSeq: 103 INVITE
    Max-Forwards: 70
    Timestamp: 1417347889
    Contact: <sip:[email protected]:5060>
    Expires: 180
    Allow-Events: telephone-event
    Content-Length: 0
    Nov 30 11:44:49.338: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Received: 
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECB16F3
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>;tag=71913148-1417348035284
    Call-ID: [email protected]
    CSeq: 103 INVITE
    Timestamp: 1417347889
    Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
    Supported: 
    Accept: application/media_control+xml,application/sdp,application/xml
    Contact: <sip:[email protected]:5060;transport=udp>
    Content-Type: application/sdp
    Content-Length: 306
    v=0
    o=BroadWorks 316169737 3 IN IP4 10.111.111.254
    s=-
    c=IN IP4 10.111.111.254
    t=0 0
    m=audio 20074 RTP/AVP 18 101
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=X-sqn:0
    a=X-cap: 1 audio RTP/AVP 100
    a=X-cpar: a=rtpmap:100 X-NSE/8000
    a=X-cpar: a=fmtp:100 200-202
    a=X-cap: 2 image udptl t38
    Nov 30 11:44:49.342: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    SIP/2.0 2
    PM-HO-VG-01#00 OK
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720a6918040f
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>;tag=3C365010-1E42
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    CSeq: 103 INVITE
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    Allow-Events: telephone-event
    Contact: <sip:[email protected]:5060>
    Supported: replaces
    Supported: sdp-anat
    Server: Cisco-SIPGateway/IOS-15.2.4.M5
    Supported: timer
    Content-Type: application/sdp
    Content-Length: 289
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 7965 2749 IN IP4 10.18.81.2
    s=SIP Call
    c=IN IP4 10.18.81.2
    t=0 0
    m=audio 22350 RTP/AVP 18 101 19
    c=IN IP4 10.18.81.2
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=yes
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=rtpmap:19 CN/8000
    a=ptime:20
    Nov 30 11:44:49.350: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received: 
    ACK sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720b594cd517
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>;tag=3C365010-1E42
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    Max-Forwards: 70
    CSeq: 103 ACK
    Allow-Events: presence
    Content-Type: application/sdp
    Content-Length: 213
    v=0
    o=CiscoSystemsCCM-SIP 9082578 3 IN IP4 10.18.81.11
    s=SIP Call
    c=IN IP4 10.18.81.10
    t=0 0
    m=audio 4000 RTP/AVP 18
    a=X-cisco-media:umoh
    a=rtpmap:18 G729/8000
    a=ptime:20
    a=fmtp:18 annexb=no
    a=sendonly
    Nov 30 11:44:49.354: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    BYE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECC55
    From: <sip:[email protected]>;tag=3C365010-1E42
    To: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
    Max-Forwards: 70
    Timestamp: 1417347889
    CSeq: 101 BYE
    Reason: Q.850;cause=86
    P-RTP-Stat: PS=874,OS=17480,PR=872,OR=17440,PL=0,JI=0,LA=0,DU=17
    Content-Length: 0
    Nov 30 11:44:49.354: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    BYE sip:[email protected]:5060;transport=udp SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECD1ECD
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>;tag=71913148-1417348035284
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
    Max-Forwards: 70
    Timestamp: 1417347889
    CSeq: 104 BYE
    Reason: Q.850;cause=65
    P-RTP-Stat: PS=872,OS=17440,PR=952,OR=19040,PL=0,JI=0,LA=0,DU=17
    Content-Length: 0
    Nov 30 11:44:49.374: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Received: 
    SIP/2.0 200 Race Condition
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECD1ECD
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>;tag=71913148-1417348035284
    Call-ID: [email protected]
    Timestamp: 1417347889
    CSeq: 104 BYE
    Content-Length: 0
    Nov 30 11:44:49.374: //64511/3CD5D2000001/SIP/Call/sipSPICallInfo: 
    The Call Setup Information is:
    Call Control Block (CCB) : 0x0x3D7B1458
    State of The Call        : STATE_DEAD
    TCP Sockets Used         : NO
    Calling Number           : 27218091323
    Called Number            : 0862000000
    Source IP Address (Sig  ): 10.18.81.2
    Destn SIP Req Addr:Port  : 10.111.111.254:5060
    Destn SIP Resp Addr:Port : 10.111.111.254:5060
    Destination Name         : 10.111.111.254
    Nov 30 11:44:49.374: //64511/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo: 
    Number of Media Streams: 1
    Media Stream             : 1
    Negotiated Codec         : g729br8
    Negotiated Codec Bytes   : 20
    Nego. Codec payload      : 18 (tx), 18 (rx)
    Negotiated Dtmf-relay    : 6
    Dtmf-relay Payload       : 101 (tx), 101 (rx)
    Source IP Address (Media): 10.18.81.2
    Source IP Port    (Media): 22256
    Destn  IP Address (Media): 10.111.111.254
    Destn  IP Port    (Media): 20074
    Orig Destn IP Address:Port (Media): [ - ]:0
    Nov 30 11:44:49.374: //64511/3CD5D2000001/SIP/Call/sipSPICallInfo: 
    Disconnect Cause (CC)    : 65
    Disconnect Cause (SIP)   : 200
    Nov 30 11:44:49.406: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Received: 
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECC55
    From: <sip:[email protected]>;tag=3C365010-1E42
    To: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    CSeq: 101 BYE
    Content-Length: 0
    Nov 30 11:44:49.406: //64510/3CD5D2000001/SIP/Call/sipSPICallInfo: 
    The Call Setup Information is:
    Call Control Block (CCB) : 0x0x3D816D70
    State of The Call        : STATE_DEAD
    TCP Sockets Used         : NO
    Calling Number           : 0218091323
    Called Number            : 0862000000
    Source IP Address (Sig  ): 10.18.81.2
    Destn SIP Req Addr:Port  : 10.18.81.11:5060
    Destn SIP Resp Addr:Port : 10.18.81.11:5060
    Destination Name         : 10.18.81.11
    Nov 30 11:44:49.406: //64510/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo: 
    Number of Media Streams: 1
    Media Stream             : 1
    Negotiated Codec         : g729br8
    Negotiated Codec Bytes   : 20
    Nego. Codec payload      : 18 (tx), 18 (rx)
    Negotiated Dtmf-relay    : 6
    Dtmf-relay Payload       : 101 (tx), 101 (rx)
    Source IP Address (Media): 10.18.81.2
    Source IP Port    (Media): 22350
    Destn  IP Address (Media): 0.0.0.0
    Destn  IP Port    (Media): 21928
    Orig Destn IP Address:Port (Media): [ - ]:0
    Nov 30 11:44:49.406: //64510/3CD5D2000001/SIP/Call/sipSPICallInfo: 
    Disconnect Cause (CC)    : 86
    Disconnect Cause (SIP)   : 200
    PM-HO-VG-01#

    Hi Manish,
    Again, excellent feedback. Much appreciated.
    I will try the commands suggested above and see if I can get DTMF to work correctly while interworking H.323 and SIP.
    But my ultimate goal is to have SIP all way from the CUCM to the CUBE and from the CUBE to the ITSP.
    If I enable SIP Early Offer with MTP on the CUCM going to the CUBE, all SIP Invite sent from the CUCM to the CUBE uses G.729r8 as the codec and once the call is established using G.729r8 should the ITSP reply with that codec, the call succeed and I am able to see an active MTP session using G.729 when I issue the command # show sccp connections.
    One thing that I saw is that my ITSP love so much sending G.729br8 most of the times, so even if using SIP EO with MTP on the SIP Trunk to the CUBE, when I sent my INVITE out from the CUCM to the CUBE using G.729r8, specially on call center numbers such as 0800 numbers, you will see that the call established but the codec being negotiated is G.729br8 which is voice only (missing DTMF).
    I will be doing some intensive test again later on this week and will send the logs. 
    Here is my question to both of you:
    Which is the best way of having a proper SIP to SIP setup all the way that will not pose any problem?
    Do I have to enable Early Offer on the SIP Profile used by the CUBE SIP Trunk or should I use the normal Standard SIP Profile? Do I need to enable MTP on my CUBE SIP Trunk or not?
    From the CUBE point of view, I have a voice class codec that support G.729r8 or G.729br8 and the DTMF Relay method supported by the ITSP is RFC 2833.
    I will send more logs for each scenario. I think that we are getting close to the resolution of this problem.
    Thanks again for your support fellows.

  • How to configure a port channel with VLAN trunking (and make it work..)

    We're trying to configure a port channel group with trunked ports to connect a NetApp HA pair. We want to create two data LIFs and connect them to the switch stack.  We are trying to create 2 data lifs, one for cifs and one for nfs that are on different vlans.
    We want the same ports to be able to allow multiple vlans to communicate. (trunked)
    These data lifs should be able to fail over to different nodes in the HA pair and still be able to communicate on the network.
    What this means is that we have to connect 4 ports each for each node in the NetApp HA Pair to the switches and create a port channel of some type that allows for trunked vlans. When we configure the ports, the configuration is as follows (below):
    We are only able to configure an IP on one of the vlans.
    When we configure an IP from another vlan for the data lif, it does not respond to a ping.
    Does anyone have any idea what I'm doing wrong on the Cisco switch?
    interface GigabitEthernet4/0/12
    description Netapp2-e0a
    switchport trunk encapsulation dot1q
    switchport trunk allowed vlan 10,20,511,519
    channel-protocol lacp
    channel-group 20 mode active
    end
    interface GigabitEthernet4/0/13
    description Netapp2-e0c
    switchport trunk encapsulation dot1q
    switchport trunk allowed vlan 10,20,511,519
    channel-protocol lacp
    channel-group 20 mode active
    end
    interface GigabitEthernet6/0/12
    description Netapp2-e0b
    switchport trunk encapsulation dot1q
    switchport trunk allowed vlan 10,20,511,519
    channel-protocol lacp
    channel-group 20 mode active
    end
    interface GigabitEthernet6/0/13
    description Netapp2-e0d
    switchport trunk encapsulation dot1q
    switchport trunk allowed vlan 10,20,511,519
    channel-protocol lacp
    channel-group 20 mode active
    end
    interface Port-channel20
    description Netapp2-NFS
    switchport trunk encapsulation dot1q
    switchport trunk allowed vlan 10,20,511,519
    spanning-tree portfast
    spanning-tree bpduguard enable
    end

    Our problem was fixed by the storage people.  They changed the server end to trunk, and the encapsulation / etherchannel.
    I like all the suggestions, and they probably helped out with the configuration getting this to work.
    Thanks!
    interface Port-channel20
    description Netapp2-NFS
    switchport trunk encapsulation dot1q
    switchport trunk allowed vlan 10,20,511,519
    switchport mode trunk
    interface GigabitEthernet4/0/12
    switchport trunk encapsulation dot1q
    switchport trunk allowed vlan 10,20,511,519
    switchport mode trunk
    channel-protocol lacp
    channel-group 20 mode active
    interface GigabitEthernet4/0/13
    switchport trunk encapsulation dot1q
    switchport trunk allowed vlan 10,20,511,519
    switchport mode trunk
    channel-protocol lacp
    channel-group 20 mode active
    interface GigabitEthernet6/0/12
    switchport trunk encapsulation dot1q
    switchport trunk allowed vlan 10,20,511,519
    switchport mode trunk
    channel-protocol lacp
    channel-group 20 mode active
    interface GigabitEthernet6/0/13
    switchport trunk encapsulation dot1q
    switchport trunk allowed vlan 10,20,511,519
    switchport mode trunk
    channel-protocol lacp
    channel-group 20 mode active

  • [svn] 3120: When you point Flex Builder at a local sandbox trunk build, it couldn' t generate the html-templates folder correctly for new projects so we moved all the html templates up one level and removed the html-templates directory and adjusted build

    Revision: 3120
    Author: [email protected]
    Date: 2008-09-05 10:44:10 -0700 (Fri, 05 Sep 2008)
    Log Message:
    When you point Flex Builder at a local sandbox trunk build, it couldn't generate the html-templates folder correctly for new projects so we moved all the html templates up one level and removed the html-templates directory and adjusted build.xml's to accommodate the directory change
    Modified Paths:
    flex/sdk/trunk/build.xml
    flex/sdk/trunk/webapps/webtier/build.xml
    Added Paths:
    flex/sdk/trunk/templates/client-side-detection/
    flex/sdk/trunk/templates/client-side-detection/AC_OETags.js
    flex/sdk/trunk/templates/client-side-detection/index.template.html
    flex/sdk/trunk/templates/client-side-detection-with-history/
    flex/sdk/trunk/templates/client-side-detection-with-history/AC_OETags.js
    flex/sdk/trunk/templates/client-side-detection-with-history/history/
    flex/sdk/trunk/templates/client-side-detection-with-history/history/history.css
    flex/sdk/trunk/templates/client-side-detection-with-history/history/history.js
    flex/sdk/trunk/templates/client-side-detection-with-history/history/historyFrame.html
    flex/sdk/trunk/templates/client-side-detection-with-history/index.template.html
    flex/sdk/trunk/templates/express-installation/
    flex/sdk/trunk/templates/express-installation/AC_OETags.js
    flex/sdk/trunk/templates/express-installation/index.template.html
    flex/sdk/trunk/templates/express-installation/playerProductInstall.swf
    flex/sdk/trunk/templates/express-installation-with-history/
    flex/sdk/trunk/templates/express-installation-with-history/AC_OETags.js
    flex/sdk/trunk/templates/express-installation-with-history/history/
    flex/sdk/trunk/templates/express-installation-with-history/history/history.css
    flex/sdk/trunk/templates/express-installation-with-history/history/history.js
    flex/sdk/trunk/templates/express-installation-with-history/history/historyFrame.html
    flex/sdk/trunk/templates/express-installation-with-history/index.template.html
    flex/sdk/trunk/templates/express-installation-with-history/playerProductInstall.swf
    flex/sdk/trunk/templates/metadata/
    flex/sdk/trunk/templates/metadata/AC_OETags.js
    flex/sdk/trunk/templates/metadata/readme.txt
    flex/sdk/trunk/templates/no-player-detection/
    flex/sdk/trunk/templates/no-player-detection/AC_OETags.js
    flex/sdk/trunk/templates/no-player-detection/index.template.html
    flex/sdk/trunk/templates/no-player-detection-with-history/
    flex/sdk/trunk/templates/no-player-detection-with-history/AC_OETags.js
    flex/sdk/trunk/templates/no-player-detection-with-history/history/
    flex/sdk/trunk/templates/no-player-detection-with-history/history/history.css
    flex/sdk/trunk/templates/no-player-detection-with-history/history/history.js
    flex/sdk/trunk/templates/no-player-detection-with-history/history/historyFrame.html
    flex/sdk/trunk/templates/no-player-detection-with-history/index.template.html
    Removed Paths:
    flex/sdk/trunk/templates/html-templates/

    Remember that Arch Arm is a different distribution, but we try to bend the rules and provide limited support for them.  This may or may not be unique to Arch Arm, so you might try asking on their forums as well.

Maybe you are looking for