CUCM 8X to Dialogic CG6000

We recently converted a client that was running a Nortel Meridian to CUCM 8.
Everything went fine except for one problem.
They had an IVR Server with a Diaglogic CG6000 to the Nortel, that would allow callers to check financial information when they called in.
We have set up a T1 CAS off the CUCM gateway and terminated into the Dialogic. The calls go to the Diaglogic but disconnect without one or two seconds. I'm trying to verify we have all the correct setting on the gateway before we force the client to make changes on their side.
I know the Diaglogic cards are specific to the application on the server to maybe there is no other option to but get them involved.
Thanks,

Hi
Thats correct UCCX 7.x is not compatible with CUCM 8.x version.
UCCX talks to CUCM with the help of JTAPI protocol, Every major release of CUCM has a different version of JTAPI which UCCX downloads from CUCM when JTAPI sync is performed.
Unfortunately the JTAPI version of CUCM 8.x is not tested with UCCX 7.x version thus its not present in compatability matrix. UCCX 8.x which is out supports CUCM 8.x Also UCCX 7.0(2) which will be coming out later this year or early next year might have the support for 8.x - not sure on that though
Hope this helps
Anuj

Similar Messages

CUCM 8.6 Call Forwarding to External Number Issue

Hello,
Call forwarding worked without problems, we could forward our phones to external numbers and everything was ok, when somebody called to my phone, I could got the call to my cell phone.
But now when I forward my phone to external number and try to call to my phone I get busy trigger.
We didn't change configuration or install any update.
I think its my ISP-s problem, to whom we have SIP Trunk.
I don't understand log file, so can you tell what is the problem?
Here is log:
057729XXXX is called party, cell phone number
original calling party number is 240XXXXX, but it is forwarded to 2484XXX
INVITE sip:2484XXX@ISP-IP:5060 SIP/2.0
Via: SIP/2.0/UDP MY-CUCM-IP:5060;branch=z9hG4bK1003a84126249
From: <sip:057729XXXX@MY-CUCM>;tag=4052091~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP-IP>
Date: Wed, 18 Dec 2013 13:34:18 GMT
Call-ID: 16d82e80-2b11a45a-c43e7-84450d0a@MY-CUCM-IP
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Cisco-Guid: 0383266432-0000065536-0000191815-2219117834
Session-Expires: 1800
P-Asserted-Identity: <sip:057729XXXX@MY-CUCM-IP>
Remote-Party-ID: <sip:057729XXXX@MY-CUCM-IP>;party=calling;screen=yes;privacy=off
Contact: <sip:057729XXXX@MY-CUCM-IP:5060>
Max-Forwards: 68
Content-Type: application/sdp
Content-Length: 215
v=0
o=CiscoSystemsCCM-SIP 4052091 1 IN IP4 MY-CUCM-IP
s=SIP Call
c=IN IP4 MY-CUCM-IP
t=0 0
m=audio 29790 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
|2,100,56,1.173711429^MY-CUCM-IP^MTP_3
17:34:18.526 |EnvProcessUdpPort - EnvProcessUdpHandler::fireSignal() varId = 2|2,100,56,1.173711429^MY-CUCM-IP^MTP_3
17:34:18.526 |EnvProcessUdpHandler::fireSignal - SEND: index = 2, handler = 0xb2d59c98|*^*^*
17:34:18.526 |EnvProcessUdpPort::fireSignal - SEND, destination = ISP-IP:5060|*^*^*
17:34:18.526 |EnvProcessUdpPort - EnvProcessUdpHandler::send(buff, 1172, ISP-IP:5060)|*^*^*
17:34:18.536 |EnvProcessUdpHandler::handle_input - handle = 334|*^*^*
17:34:18.536 |EnvProcessUdpHandler::handle_input   Status: 0, Id: 2|*^*^*
17:34:18.536 |//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 358 from ISP-IP:[5060]:
[12623361,NET]
SIP/2.0 100 Trying
Call-ID: 16d82e80-2b11a45a-c43e7-84450d0a@MY-CUCM-IP
CSeq: 101 INVITE
From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052091~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP-IP>;tag=sip+1+b3a00013+867def6a
Via: SIP/2.0/UDP MY-CUCM-IP:5060;branch=z9hG4bK1003a84126249
Server: CISCO-SBC/2.x
Content-Length: 0
|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/Stack/Info/0x0/ccsip_spi_get_msg_type returned: 2 for event 1|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/Stack/Transport/0x0/context=(nil)|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/Stack/Transport/0x0/gConnTab=0xf484290, addr=ISP-IP, port=5060, connid=2, transport=UDP|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/Stack/Info/0x0/Return existing connection for port 5060 connId 2|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/Stack/Info/0x0/Checking Invite Dialog|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/Stack/Info/0xb1b50c90/INVITE response with no RSEQ - disable IS_REL1XX|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/SIPHandler/ccbId=0/scbId=0/sip_stop_timer: type=SIP_TIMER_TRYING value=500 retries=3|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/Stack/States/0xb1b50c90/0xb1b50c90 : State change from (STATE_SENT_INVITE, SUBSTATE_NONE) to (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_PROCEEDING)|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/SIPHandler/ccbId=0/scbId=0/sip_stop_timer: type=SIP_TIMER_EXPIRES value=180000 retries=0|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/SIPHandler/ccbId=0/scbId=0/sip_start_timer: type=SIP_TIMER_EXPIRES value=180000 retries=0|2,100,230,1.4901096^ISP-IP^*
17:34:18.561 |EnvProcessUdpHandler::handle_input - handle = 334|*^*^*
17:34:18.561 |EnvProcessUdpHandler::handle_input   Status: 0, Id: 2|*^*^*
17:34:18.561 |//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 396 from ISP-IP:[5060]:
[12623362,NET]
SIP/2.0 403 Forbidden
Call-ID: 16d82e80-2b11a45a-c43e7-84450d0a@MY-CUCM-IP
CSeq: 101 INVITE
From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052091~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP-IP>;tag=sip+1+b3a00013+867def6a
Via: SIP/2.0/UDP MY-CUCM-IP:5060;branch=z9hG4bK1003a84126249
Server: CISCO-SBC/2.x
Content-Length: 0
Contact: <sip:ISP-IP:5060>
[12623363,NET]
ACK sip:2484XXX@ISP-IP:5060 SIP/2.0
Via: SIP/2.0/UDP MY-CUCM-IP:5060;branch=z9hG4bK1003a84126249
From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052091~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP-IP>;tag=sip+1+b3a00013+867def6a
Date: Wed, 18 Dec 2013 13:34:18 GMT
Call-ID: 16d82e80-2b11a45a-c43e7-84450d0a@MY-CUCM-IP
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence
Content-Length: 0
INVITE sip:2484XXX@ISP's-Other-IP:5062 SIP/2.0
Via: SIP/2.0/UDP MY-CUCM-IP:5062;branch=z9hG4bK1003a95b8f3900
From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052092~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP's-Other-IP>
Date: Wed, 18 Dec 2013 13:34:18 GMT
Call-ID: 16d82e80-2b11a45a-c43e8-84450d0a@MY-CUCM-IP
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Cisco-Guid: 0383266432-0000065536-0000191816-2219117834
Session-Expires: 1800
P-Asserted-Identity: <sip:057729XXXX@MY-CUCM-IP>
Remote-Party-ID: <sip:057729XXXX@MY-CUCM-IP>;party=calling;screen=yes;privacy=off
Contact: <sip:057729XXXX@MY-CUCM-IP:5062>
Max-Forwards: 68
Content-Type: application/sdp
Content-Length: 215
v=0
o=CiscoSystemsCCM-SIP 4052092 1 IN IP4 MY-CUCM-IP
s=SIP Call
c=IN IP4 MY-CUCM-IP
t=0 0
m=audio 29792 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
|2,100,56,1.173711431^MY-CUCM-IP^MTP_3
17:34:18.567 |EnvProcessUdpPort - EnvProcessUdpHandler::fireSignal() varId = 0|2,100,56,1.173711431^MY-CUCM-IP^MTP_3
17:34:18.567 |EnvProcessUdpHandler::fireSignal - SEND: index = 0, handler = 0xa6b4d7c0|*^*^*
17:34:18.567 |EnvProcessUdpPort::fireSignal - SEND, destination = ISP's-Other-IP:5062|*^*^*
17:34:18.567 |EnvProcessUdpPort - EnvProcessUdpHandler::send(buff, 1177, ISP's-Other-IP:5062)|*^*^*
17:34:18.569 |EnvProcessUdpHandler::handle_input - handle = 335|*^*^*
17:34:18.569 |EnvProcessUdpHandler::handle_input   Status: 0, Id: 0|*^*^*
17:34:18.569 |//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 394 from ISP's-Other-IP:[5062]:
[12623365,NET]
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP MY-CUCM-IP:5062;branch=z9hG4bK1003a95b8f3900;rport=5062
From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052092~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP's-Other-IP>
Call-ID: 16d82e80-2b11a45a-c43e8-84450d0a@MY-CUCM-IP
CSeq: 101 INVITE
Server: kamailio (3.3.1 (x86_64/linux))
Content-Length: 0
17:34:18.587 |//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 375 from ISP's-Other-IP:[5062]:
[12623366,NET]
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP MY-CUCM-IP:5062;branch=z9hG4bK1003a95b8f3900;rport=5062
Call-ID: 16d82e80-2b11a45a-c43e8-84450d0a@MY-CUCM-IP
From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052092~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP's-Other-IP>;tag=dc6a4ae7
CSeq: 101 INVITE
Reason: Q.850;cause=0;text="unknown"
Content-Length: 0
|2,100,230,1.4901099^ISP's-Other-IP^*
[12623367,NET]
ACK sip:2484XXX@ISP's-Other-IP:5062 SIP/2.0
Via: SIP/2.0/UDP MY-CUCM-IP:5062;branch=z9hG4bK1003a95b8f3900
From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052092~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP's-Other-IP>;tag=dc6a4ae7
Date: Wed, 18 Dec 2013 13:34:18 GMT
Call-ID: 16d82e80-2b11a45a-c43e8-84450d0a@MY-CUCM-IP
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence
Content-Length: 0

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

How to change the message in the Missing Plug-in dialog?

Hi, all.
I want to change the message in the Missing Plug-in dialog
when a doc is opened containg data from a missing plug-in.
I defined MissingPluginURLValue and MissingPluginAlertValue in ID.h
which are used in ExtraPluginInfo.
But there is nothing changed in the Missing Plug-in dialog.
Can someone help me?
BestRegard.
Miya.

As far as I know you can't change that on CUCM.
On CME you can.
Bye,
Jan
Sent from Cisco Technical Support iPad App

CUCM and Avaya CS1000 SIP connection

                   Hello - looking for some help on a SIP trunk configuration between the 2 devices. Currently we are running CUCM 9.1 with Avaya Session Manager 6.3. We are having issues with the call completing from the CS 1000 to the CUCM. Below are the session traces from both the Session Manager and CUCM. The CS1000 currently has 4 digit extensions and the CUCM has 7 digit extensions. We translate the number in the Session Manager to send the 7 digits. If you could lead in the right direction I would appreciate it. WOuld this have something to do with the context coming out of Session Manager? It looks like CDP.UDP and then only the 4 digits and the CUCM needs the 7. I also attached the configurations guide used for this.
Session Manager trace:
   mil-ss-01                 CUCM 9
                       SM100                10.101.2.75             10.174.2.75
13:04:53.730 |<--OPTIONS-|           |           |           |           | (1) sip:10.5.1.30
13:04:53.731 |--200 OK-->|           |           |           |           | (1) 200 OK (OPTIONS)
13:04:53,734 |                    Request Adaptation                     | Adapter: mil-ss-01
13:04:53,734 |                    Request Adaptation                     | Adapter: mil-ss-01
13:04:56.183 |--INVITE-->|           |           |           |           | (2) T:7317;phone-context=cdp.udp F:anonymous@anonymous U:7317;phone-context=cdp.udp
13:04:56.184 |<--Trying--|           |           |           |           | (2) 100 Trying
13:04:56,185 |                  Remote host is trusted                   | Trusted
13:04:56,185 |                    Request Adaptation                     | Adapter: mil-ss-01
13:04:56,186 |                Applied ingress Adaptation                 | P-Asserted-Identity=<sip:[email protected]>, Request-URI=sip:[email protected], History-Info=<sip:[email protected]>;index=1, <sip:[email protected]>;index=1.1
13:04:56,186 |                Originating Location found                 | Location: mil-cs1000m-01
13:04:56,186 |        Try routing to determine if emergency call         | Location: mil-cs1000m-01
13:04:56,186 |                Request Dial Pattern route                 | for: sip:[email protected] Location: mil-cs1000m-01
13:04:56,186 |               Dial Pattern route parameters               | URI Domain: company.com Location: mil-cs1000m-01
13:04:56,186 |                 Trying Dial Pattern route                 | Domain: company.com Location: mil-cs1000m-01
13:04:56,186 |                    Dial Pattern found                     | for: 7317 Pattern: 7317
13:04:56,186 |                    Route Policy found                     | Pattern: 7317 RoutePolicyList: to_CUCM9
13:04:56,187 |                        Route found                        | for: sip:[email protected] SIPEntity: CUCM 9
13:04:56,187 |                     Entity Link found                     | SIPEntity: CUCM 9 EntityLink: mil-sessionmgr-01->TCP, biDirId=null, deny=false:5060
13:04:56,187 |                    Request Adaptation                     | Adapter: CUCM 9
13:04:56,188 |                 Applied egress Adaptation                 | NoAdaptationModuleExists=true, Request-URI=sip:[email protected];routeinfo=0-0, Remote-Party-ID=<sip:[email protected]>;party=calling;screen=no;privacy=off,
13:04:56,188 |                    Routing SIP request                    | SipEntity: CUCM 9 EntityLink: mil-sessionmgr-01->TCP:5060
13:04:56,189 |              No hostname resolution required              | Routing to: sip:10.5.131.12;transport=tcp;lr;phase=terminating
13:04:56.191 |           |--INVITE-->|           |           |           | (2) T:7317;phone-context=cdp.udp F:anonymous@anonymous U:7657317 P:terminating
13:04:56.196 |           |<--Trying--|           |           |           | (2) 100 Trying
13:04:56.198 |           |<--Not Fou-|           |           |           | (2) 404 Not Found
13:04:56.199 |           |----ACK--->|           |           |           | (2) sip:[email protected]
13:04:56,200 |                    Request Adaptation                     | Adapter: CUCM 9
13:04:56,201 |                    Request Adaptation                     | Adapter: CUCM 9
13:04:56,201 |                    Request Adaptation                     | Adapter: mil-ss-01
13:04:56.202 |<--Not Fou-|           |           |           |           | (2) 404 Not Found
13:04:56.203 |----ACK--->|           |           |           |           | (2) sip:7317
13:05:07,597 |                Remote host is not trusted                 | Host not trusted
13:05:07,597 |                Originating Location found                 | Location: mil-cs1000m-01
13:05:12.657 |           |<--------OPTIONS-------|           |           | (3) sip:10.5.2.51
13:05:12,659 |                Remote host is not trusted                 | Host not trusted
13:05:12,659 |                Originating Location found                 | Location: mil-cs1000m-01
13:05:12.660 |           |--------200 OK-------->|           |           | (3) 200 OK (OPTIONS)
13:05:16.877 |           |<--------------OPTIONS-------------|           | (4) sip:10.5.2.51
13:05:16,879 |                  Remote host is trusted                   | Trusted
13:05:16,879 |                    Request Adaptation                     | Adapter: mil-ss-01
13:05:16,879 |                Applied ingress Adaptation                 | P-Asserted-Identity=<sip:[email protected]>
13:05:16,879 |                Originating Location found                 | Location: sas-cs1000e-01
13:05:16.880 |           |--------------200 OK-------------->|           | (4) 200 OK (OPTIONS)
13:05:24.463 |           |<--------------------OPTIONS-------------------| (5) sip:10.5.2.51
13:05:24,465 |                Remote host is not trusted                 | Host not trusted
13:05:24,465 |                Originating Location found                 | Location: mil-cs1000m-01
13:05:24.466 |           |--------------------200 OK-------------------->| (5) 200 OK (OPTIONS)
CUCM Trace Invite:
SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.5.2.51 on port 46314 index 6 with 3188 bytes:
[1179,NET]
INVITE sip:[email protected] SIP/2.0
P-AV-Message-Id: 1_1
Route: <sip:10.5.131.12;transport=tcp;lr;phase=terminating>
History-Info: <sip:[email protected]>;index=1, <sip:[email protected]>;index=1.1
Remote-Party-ID: <sip:[email protected]>;party=calling;screen=no;privacy=off
Allow: INVITE, ACK, BYE, REGISTER, REFER, NOTIFY, CANCEL, PRACK, OPTIONS, INFO, SUBSCRIBE, UPDATE
Contact: <sip:00000000;[email protected]:5060;maddr=10.5.1.30;transport=tcp;user=phone;gsid=68cac530-5d21-11e3-8b45-78e3b505dc88>
Alert-Info: <cid:[email protected]>
Supported: 100rel, x-nortel-sipvc, replaces
Via: SIP/2.0/TCP 10.5.2.51;branch=z9hG4bK138498682095974-AP;ft=3
Via: SIP/2.0/TCP 10.5.2.50:15060;rport=30522;ibmsid=local.1372169047609_2400498_2400522;branch=z9hG4bK138498682095974
Via: SIP/2.0/TCP 10.5.2.50:15060;rport;ibmsid=local.1372169047609_2400497_2400521;branch=z9hG4bK129750805377494
Via: SIP/2.0/TCP 10.5.2.51;branch=z9hG4bK-3fa2ea5-894636b6-7d11effb-AP;ft=19;received=10.5.2.51;rport=19140
Via: SIP/2.0/TCP 10.5.1.30:5060;branch=z9hG4bK-3fa2ea5-894636b6-7d11effb
Record-Route: <sip:[email protected];transport=tcp;lr>
Record-Route: <sip:10.5.2.50:15060;transport=tcp;ibmsid=local.1372169047609_2400497_2400521;lr>
Record-Route: <sip:[email protected];transport=tcp;lr>
P-Charging-Vector: icid-value="68cac530-5d21-11e3-8b45-78e3b505dc88"
User-Agent: Nortel CS1000 SIP GW release_7.0 version_linux-6.50.00 AVAYA-SM-6.3.1.0.631004
P-Asserted-Identity: <sip:[email protected]>
From: <sip:[email protected];user=phone>;tag=ac493fe8-1e01050a-13c4-55013-3fa2ea5-2b7d5ad-3fa2ea5
To: <sip:7317;[email protected];user=phone>
Call-ID: abef5b48-1e01050a-13c4-55013-3fa2ea5-5d08551f-3fa2ea5
Max-Forwards: 66
CSeq: 1 INVITE
Content-Type: multipart/mixed;boundary=unique-boundary-1
Content-Length: 1063
Av-Global-Session-ID: 68cac530-5d21-11e3-8b45-78e3b505dc88
P-Location: SM;origlocname="mil-cs1000m-01";origsiglocname="mil-cs1000m-01";origmedialocname="mil-cs1000m-01";termlocname="Cisco BE6K";termsiglocname="Cisco BE6K";smaccounting="true"
--unique-boundary-1
Content-Type: application/sdp
SDP Message
====================================================
v=0
o=- 746 1 IN IP4 10.5.1.30
s=-
c=IN IP4 10.5.1.36
t=0 0
m=audio 5234 RTP/AVP 18 0 8 101 111
c=IN IP4 10.5.1.36
a=tcap:1 RTP/SAVP
a=crypto:XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX
a=crypto:XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX
a=pcfg:1 t=1
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:111 X-nt-inforeq/8000
a=ptime:20
a=sendrecv
--unique-boundary-1
Content-Type: application/x-nt-mcdn-frag-hex;version=linux-6.50.00;base=x2611
Content-Disposition: signal;handling=optional
0500bc05
0107130081900000a200
09090f00e9a4830001004000
1315070011fa0f00a10d02010102020100cc040000c56000
1e0403008183
4a1c0100180001001a011404000067353505000004000000000048710000
--unique-boundary-1
Content-Type: application/x-nt-epid-frag-hex;version=linux-6.50.00;base=x2611
Content-Disposition: signal;handling=optional
011201
3c:4a:92:f4:84:f4
--unique-boundary-1--
CUCM Trying Message:
SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.5.2.51 on port 46314 index 6
[1180,NET]
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 10.5.2.51;branch=z9hG4bK138498682095974-AP;ft=3,SIP/2.0/TCP 10.5.2.50:15060;rport=30522;ibmsid=local.1372169047609_2400498_2400522;branch=z9hG4bK138498682095974,SIP/2.0/TCP 10.5.2.50:15060;rport;ibmsid=local.1372169047609_2400497_2400521;branch=z9hG4bK129750805377494,SIP/2.0/TCP 10.5.2.51;branch=z9hG4bK-3fa2ea5-894636b6-7d11effb-AP;ft=19;received=10.5.2.51;rport=19140,SIP/2.0/TCP 10.5.1.30:5060;branch=z9hG4bK-3fa2ea5-894636b6-7d11effb
From: <sip:[email protected];user=phone>;tag=ac493fe8-1e01050a-13c4-55013-3fa2ea5-2b7d5ad-3fa2ea5
To: <sip:7317;[email protected];user=phone>
Date: Wed, 04 Dec 2013 19:12:41 GMT
Call-ID: abef5b48-1e01050a-13c4-55013-3fa2ea5-5d08551f-3fa2ea5
CSeq: 1 INVITE
Allow-Events: presence
Content-Length: 0
SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.5.2.51 on port 46314 index 6
[1181,NET]
SIP/2.0 404 Not Found
Via: SIP/2.0/TCP 10.5.2.51;branch=z9hG4bK138498682095974-AP;ft=3,SIP/2.0/TCP 10.5.2.50:15060;rport=30522;ibmsid=local.1372169047609_2400498_2400522;branch=z9hG4bK138498682095974,SIP/2.0/TCP 10.5.2.50:15060;rport;ibmsid=local.1372169047609_2400497_2400521;branch=z9hG4bK129750805377494,SIP/2.0/TCP 10.5.2.51;branch=z9hG4bK-3fa2ea5-894636b6-7d11effb-AP;ft=19;received=10.5.2.51;rport=19140,SIP/2.0/TCP 10.5.1.30:5060;branch=z9hG4bK-3fa2ea5-894636b6-7d11effb
From: <sip:[email protected];user=phone>;tag=ac493fe8-1e01050a-13c4-55013-3fa2ea5-2b7d5ad-3fa2ea5
To: <sip:7317;[email protected];user=phone>;tag=580~35abe322-3212-479c-8a2c-ab75ead590d5-21575914
Date: Wed, 04 Dec 2013 19:12:41 GMT
Call-ID: abef5b48-1e01050a-13c4-55013-3fa2ea5-5d08551f-3fa2ea5
CSeq: 1 INVITE
Allow-Events: presence
Reason: Q.850;cause=1
Content-Length: 0
SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.5.2.51 on port 46314 index 6 with 623 bytes:
[1182,NET]
ACK sip:[email protected] SIP/2.0
Route: <sip:10.5.131.12;transport=tcp;lr;phase=terminating>
Call-ID: abef5b48-1e01050a-13c4-55013-3fa2ea5-5d08551f-3fa2ea5
From: <sip:[email protected];user=phone>;tag=ac493fe8-1e01050a-13c4-55013-3fa2ea5-2b7d5ad-3fa2ea5
To: <sip:7317;[email protected];user=phone>;tag=580~35abe322-3212-479c-8a2c-ab75ead590d5-21575914
Via: SIP/2.0/TCP 10.5.2.51;branch=z9hG4bK138498682095974-AP;ft=3
Via: SIP/2.0/TCP 10.5.2.50:15060;rport=30522;ibmsid=local.1372169047609_2400498_2400522;branch=z9hG4bK138498682095974
CSeq: 1 ACK
Max-Forwards: 66
Content-Length: 0
SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.5.2.51 on port 46314 index 6
[1180,NET]
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 10.5.2.51;branch=z9hG4bK138498682095974-AP;ft=3,SIP/2.0/TCP 10.5.2.50:15060;rport=30522;ibmsid=local.1372169047609_2400498_2400522;branch=z9hG4bK138498682095974,SIP/2.0/TCP 10.5.2.50:15060;rport;ibmsid=local.1372169047609_2400497_2400521;branch=z9hG4bK129750805377494,SIP/2.0/TCP 10.5.2.51;branch=z9hG4bK-3fa2ea5-894636b6-7d11effb-AP;ft=19;received=10.5.2.51;rport=19140,SIP/2.0/TCP 10.5.1.30:5060;branch=z9hG4bK-3fa2ea5-894636b6-7d11effb
From: <sip:[email protected];user=phone>;tag=ac493fe8-1e01050a-13c4-55013-3fa2ea5-2b7d5ad-3fa2ea5
To: <sip:7317;[email protected];user=phone>
Date: Wed, 04 Dec 2013 19:12:41 GMT
Call-ID: abef5b48-1e01050a-13c4-55013-3fa2ea5-5d08551f-3fa2ea5
CSeq: 1 INVITE
Allow-Events: presence
Content-Length: 0
CUCM not found message:
SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.5.2.51 on port 46314 index 6
[1181,NET]
SIP/2.0 404 Not Found
Via: SIP/2.0/TCP 10.5.2.51;branch=z9hG4bK138498682095974-AP;ft=3,SIP/2.0/TCP 10.5.2.50:15060;rport=30522;ibmsid=local.1372169047609_2400498_2400522;branch=z9hG4bK138498682095974,SIP/2.0/TCP 10.5.2.50:15060;rport;ibmsid=local.1372169047609_2400497_2400521;branch=z9hG4bK129750805377494,SIP/2.0/TCP 10.5.2.51;branch=z9hG4bK-3fa2ea5-894636b6-7d11effb-AP;ft=19;received=10.5.2.51;rport=19140,SIP/2.0/TCP 10.5.1.30:5060;branch=z9hG4bK-3fa2ea5-894636b6-7d11effb
From: <sip:[email protected];user=phone>;tag=ac493fe8-1e01050a-13c4-55013-3fa2ea5-2b7d5ad-3fa2ea5
To: <sip:7317;[email protected];user=phone>;tag=580~35abe322-3212-479c-8a2c-ab75ead590d5-21575914
Date: Wed, 04 Dec 2013 19:12:41 GMT
Call-ID: abef5b48-1e01050a-13c4-55013-3fa2ea5-5d08551f-3fa2ea5
CSeq: 1 INVITE
Allow-Events: presence
Reason: Q.850;cause=1
Content-Length: 0
CUCM ACK message:
SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.5.2.51 on port 46314 index 6 with 623 bytes:
[1182,NET]
ACK sip:[email protected] SIP/2.0
Route: <sip:10.5.131.12;transport=tcp;lr;phase=terminating>
Call-ID: abef5b48-1e01050a-13c4-55013-3fa2ea5-5d08551f-3fa2ea5
From: <sip:[email protected];user=phone>;tag=ac493fe8-1e01050a-13c4-55013-3fa2ea5-2b7d5ad-3fa2ea5
To: <sip:7317;[email protected];user=phone>;tag=580~35abe322-3212-479c-8a2c-ab75ead590d5-21575914
Via: SIP/2.0/TCP 10.5.2.51;branch=z9hG4bK138498682095974-AP;ft=3
Via: SIP/2.0/TCP 10.5.2.50:15060;rport=30522;ibmsid=local.1372169047609_2400498_2400522;branch=z9hG4bK138498682095974
CSeq: 1 ACK
Max-Forwards: 66
Content-Length: 0
Thanks.

This document worked for us between CUCM BE6000 ver 9.0 and the Avaya.
The main focus on the Cisco side is this: Page 37 - 41
5.4. Define SIP Trunk Security Profile
Expand System  Security Profile and select SIP Trunk Security Profile. Click to
configure a SIP Trunk Security Profile.
Enter the following values and use defaults for remaining fields:
 Name Enter name
 Description Enter a brief description
 Incoming Transport Type Verify “TCP+UDP” is selected
 Outgoing Transport Type Verify “TCP” is selected
 Accept Out-of-Dialog REFER Enter
 Accept Unsolicited Notification Enter
 Accept Replaces Header Enter
Click . The screen below shows SIP Trunk Security Profile for the sample configuration
5.5. Define SIP Profile
Expand Device  Device Settings and select SIP Profile. Click to configure a SIP
Profile.
Under SIP Profile Information section, enter the following values and use defaults for
remaining fields:
 Name Enter name
 Description Enter a brief description
 Default MTP Telephony Event Payload Type Enter “120”
 Disable Early Media on 180 Enter
Note: Disabling Early Media allows local ringback to be used.
Under Parameters used in Phone section, scroll to end of section and enter the following values
and use defaults for remaining fields:
 RFC 2543 Hold Enter
Click . The screen below shows SIP Profile for the sample configuration.

Differnece in assigning N/W VRU CUCM and CVP

Hi All - I just want to understand below logic and why we no need to assign N/W VRU for CUCM RC.
Below is the call flow.
N/W VRU is assigned in CUCM_RC
IP Phone --> CTI R.P --> ICM --> CVP
When i assign N/W VRU for CUCM RC, ICM is sending only VRU Label and we have Max DNIS length as 10 for CVP, Since its sending only 10 Digits lablel as DN CVP its treating as new call instead of CORR ID routed call and ICM expecting DN.
// CVP
1514: 172.17.16.21: Jul 06 2014 21:03:53.067 +0400: %CVP_9_0_ICM-7-CALL: {Thrd=pool-1-thread-174-ICM-476} CALLGUID = 8288AB800001000000000021071011AC, DLGID = 100 [SIP_LEG] - Publishing ,, [ICM_NEW_CALL],   dialogueId=100,   sendSeqNo=1,   trunkGroupId=100,   trunkNumber=0,   serviceId=1,   dialedNumber=5555555555,   uui=,   callguid=8288AB800001000000000021071011AC,   rckey=,   rcday=,   rcseq=,   location=,   locationpkid=,   pstntrunkgroupid=172.17.16.7 ,   pstntrunkgroupchannelnum=2147483647,   sipheader=,   CallContext:,     user.media.id: 8288AB800001000000000021071011AC,     user.cvp_server_info: 172.17.16.21,, LEGID = 8288ab80-3b91817c-c0-71011ac, DNIS = 5555555555, ANI = 1008
// RTR
21:15:26:364 ra-rtr Trace: (65582 x 0 : 0 0) NewCall: CID=(151031,173), DN=2000, ANI=1008, CED=, RCID=5000, MRDID=1, CallAtVRU=0, OpCode=0
21:15:26:364 ra-rtr Trace:    RCKSeqNum=1, NIC_DN=
21:15:26:364 ra-rtr Trace: CallType(5000, 491): Init CT_SL_Timer: Threshold(20 + 2).
21:15:26:364 ra-rtr Trace: (65582 x 0 : 0 0) Correlation id for dialog is (101).
21:15:26:364 ra-rtr Trace: (65582 x 101 : 0 0) TransferToVRU: Label=5555555555, CorID=101, VRUID=5000, RCID=5000
21:15:26:364 ra-rtr Trace: (65582 x 101 : 0 0) TransferConnect sent. Dialog pending.
21:15:26:402 ra-rtr Trace: (108 x 0 : 0 0) NewCall: CID=(151031,174), DN=5555555555, ANI=1008, CED=, RCID=5002, MRDID=1, CallAtVRU=1, OpCode=0
21:15:26:402 ra-rtr Trace:    RCKSeqNum=0, NIC_DN=5555555555
21:15:26:402 ra-rtr Trace: (108 x 0 : 0 0) Unknown dialed number (5555555555) - and NO default call types
21:15:26:402 ra-rtr Trace: (108 x 0 : 0 0) Call route request from routing client CVP1_RC (ID 5002) with unknown DN of 5555555555.
21:15:26:402 ra-rtr Trace: (108 x 0 : 0 0) Deleting Dialog.
21:15:45:916 ra-rtr Trace: (65582 x 101 : 0 0) Dialog timed out callstate is :(1).
21:15:45:916 ra-rtr Trace: (65582 x 101 : 0 0) Dialog resuming (Script Node timed out.) status (2)
21:15:45:916 ra-rtr Trace: (65582 x 101 : 0 0) RouteComplete:
21:15:45:916 ra-rtr Trace:    Route: CID=(151031,173), Labels=0
21:15:45:916 ra-rtr Trace: CallType(5000, 491): Deleting CT_SL_Timer.
21:15:45:916 ra-rtr Trace: (65582 x 101 : 0 0) Dialog sending release call message to Routing Client ID(5000).
21:15:45:916 ra-rtr Trace: (65582 x 101 : 0 0) Deleting Dialog
Once i assigned DN in ICM for 5555555555 its started working fine and this shouln't be the case.
Later on i removed N/W VRU in CUCM_RC and ICM started sending label for CUCM along with CORR ID.
NO N/W VRU assigned to CUCM_RC
1618: 172.17.16.21: Jul 06 2014 21:17:09.265 +0400: %CVP_9_0_ICM-7-CALL: {Thrd=pool-1-thread-276-ICM-509} CALLGUID = 5CFCA1800001000000000027071011AC - Correlation ID routed call
1619: 172.17.16.21: Jul 06 2014 21:17:09.265 +0400: %CVP_9_0_ICM-7-CALL: {Thrd=pool-1-thread-276-ICM-509} CALLGUID = 5CFCA1800001000000000027071011AC, DLGID = 109 [SIP_LEG_PRERTE_CORRID] - Publishing ,, [ICM_REQUEST_INSTRUCTION],   dialogueId=109,   sendSeqNo=1,   trunkGroupId=200,   trunkNumber=0,   serviceId=2,   uui=,   correlationId=102,   location=,   locationpkid=,   pstntrunkgroupid=172.17.16.7 ,   pstntrunkgroupchannelnum=2147483647,   sipheader=,, LEGID = 5cfca180-3b918498-c6-71011ac, DNIS = 5555555555102, ANI = 1008
// RTR
21:17:09:735 ra-rtr Trace: CallType(5000, 492): Init CT_SL_Timer: Threshold(20 + 2).
21:17:09:735 ra-rtr Trace: (65583 x 0 : 0 0) Correlation id for dialog is (102).
21:17:09:735 ra-rtr Trace: (65583 x 102 : 0 0) TransferToVRU: Label=5555555555, CorID=102, VRUID=5000, RCID=5000
21:17:09:735 ra-rtr Trace: (65583 x 102 : 0 0) TransferConnect sent. Dialog pending.
21:17:09:760 ra-rtr Trace: (65583 109 102 : 0 0) RequestInstr: CID=(151031,175), CallState=1
21:17:09:760 ra-rtr Trace: (109 109 102 : 0 0) Dialog initiating 2nd phase of transfer.
21:17:09:760 ra-rtr Trace: (109 109 102 : 0 0) Correlation id for dialog is (103).
21:17:09:760 ra-rtr Trace: (109 109 103 : 0 0) TransferToVRU: Label=7777777777, CorID=103, VRUID=5000, RCID=5002
Regards,
Siva

Because CUCM is not VRU, and that tab is useful when deciding Type of VRU(Type 10,2 etc) if peripheral is VRU.
Just have a look at screenshot taken from Config guide, it clears says assign if Peripheral is VRU.
Regards
Chintan
~rate if helpful

CUCM-VCS Integration VCS B2BUA Encryption Call Failures

All,
I have the following scenario:
CUCM 9.1.2SU1
VCS X8.1.1
MX300 endpoints (CUCM registered)
We are not running in mixed mode on CUCM
We want media streams with external call parties to be encrypted. We do have TLS end-to-end but I don't believe we can support SRTP to the MX300s registered to UCM w/o provisioning mixed mode (based on Cisco docs). So, we are attempting to use Media encryption policy on the VCS. Specifically, we set one of the traversal client zone to use "Best effort". This works for most calls but we have seen a couple of calls fail.
From end user perspective, failures manifest as a call that gets connected and is immediately torn down.
On the VCS, we will see the following when looking at the call history:
The B2BUA Encryption component is disconnected after ~3 seconds. The disconnect reason is: B2BUA disconnected call on the ingress saying "mismatched transport type in answer".
Based on context clues, this points to TLS negotiation. The thing is, if I set the media policy back to "auto" then the call connects fine and the transport is TLS. At least, it reports TLS on my VCS-C and VCS-E.
Any pointers that someone is willing to toss my way?
Thanks in advance,
Bill (@ucguerrilla)

Won't help but I have a very similar but slightly different scenario with:
CUCM 9.1
VCS 8.2.2
Jabber 10 or CUCM registered TC endpoint
As for settings:
CUCM-VCS SIP trunk is TCP not encrypted (never got it to work following the doc step by step....)
VCS-C to VCS-E is TLS as setup on the doc.
On the VCS-C, the DNSZone Media Encrytion mode is set to "Auto"
Some SIP calls work perfectly (i.e. the Cisco test endpoints) but some users have issues. Dialing partners' cloud service video-conference, the call connects and gets dropped immediately. I created myself a trial account on that service to test and can reproduce it all the time. I can see the call coming in my cloud service client and when I accept it it just drops.
On the VCS-C,
I see a SIP 200 OK
an then a call component status=disconnected type=B2BUA
State
Inactive
Start time
2014-11-11 16:51:22
Duration
5 seconds
Disconnect reason summary
disconnected
Disconnect reason details
B2BUA disconnected call on the Egress saying "Received 'Request Timeout' to mid-dialog request"
But on the VCS-E in the call history, I only see and "408 request timeout".
When I call my Jabber account from that service it works well. But in that case the second call component with type B2BUA shows:
State
Inactive
Start time
2014-11-11 17:14:02
Duration
40 seconds
Disconnect reason summary
BYE
Disconnect reason details
Egress disconnected call
Tag
3d14cee5-01ad-4468-9e3b-e0925dde15d4
Box call serial number
1bc2473f-2a09-4dea-8ffd-a7e88a3ef05b
Have also no clue of what is happening

Filename in Save As dialog when saving PDF file

I have a web site serving up PDF files. Without going into details, the URLs of the PDFs are not just your basic URL but look simething like this:
http://www.domain.com/WorkFlowApp/Clients/demo1/secure/Promo%20Cover%20Cool-MedH i.pdf?userId=55b64ad5-28a2-490e-b3d6-4944099a390f#collab=CollabService@http://ww w.domain.com/ICCollab/IC_Service.asmx?WSDL
In this case the actual filename is "Promo Cover Cool-MedHi.pdf"
While viewing in Safari (OS X), if the user saves the PDF to their local machine, the Save As dialog comes up and the filename field is automatically populated. The problem is, instead of using just the filename, the field contains this:
http---www.domain.com-WorkFlowApp-Clients-demo1-secure-Promo%20Cover%20Cool-MedH i.pdf?userId=55b64ad5-28a2-490e-b3d6-4944099a390f#collab=CollabService@http---ww w.domain.com-ICCollab-IC_Service.asmx?WSDL
I know when performing the same action on Windows using Internet Explorer, the Save As dialog box filename field is populated correctly with just the filename (although it shows up URL encoded as Promo%20Cover%20Cool-MedHi.pdf, which is OK).
I'd like to be able to force Safari to use just the filename in the Save As dialog box. Has anyone come accross this problem and figured out a solution?

Hello Ebnul.nao
I am having this same problemwith trying to work this out.
Did you succeed in the end?
I will really appreciate your help if you did
Kind regards

Jabber 9.1 Registration To CUCM 8.6.2

Good evening,
I have a problem I am hoping someone can answer for me. I have a CUCM/ Unity Connection/ CUP platform which are all 8.6x
I have configured all the components and I am currently trying to get Jabber to work, I have 1 user who has 2 clients he wants to use. One client is is home Mac and the other his work Windows based laptop.
Jabber 8.6.4 on the Mac works 100% the client can call, access voicemail and use IM with no issues. The same user when he tries to login with his Windows based client gets IM and authenticates to Unity Connection OK but the client does not register with CUCM and therefore he cannot dial out and so all telephony functionality is unavailable. The options in the windows client only show Phone accounts for the voicemail server which all look OK.
It looks to me that the windows client is not getting the correct paramters from the jabber-xml file (infact I cant find one in the tftp server which is CUCM) or maybe I need to have 2 softphones detailed in CUCM, i.e one for each client (MAC & PC). I have all the CCMCIP and trunking etc working fine, along with the relevant AXL, LDAP and CTI users set up ok.
Anyone have an idea of where I may be going wrong ?
Thanks
S

Hi Jonathan
I have tried your suggestion but while the symptoms are slightly different I am still not able to get the Windows client to register with CUCM.
If I disable the softphone on the mac client then exit Jabber on the mac, then I open the Windows client initially the results are exactly the same. However if I then sign out of the windows client and back in I get the option on the bottom right of the client to choose the softphone source (i.e computer or deskphone) but then no matter which one I choose the client sits at the setting phone prompt and still does not register.
I guess one option for me may be to run a WIndows client on the home Mac using VMware fusion and use the same client, hopefully then the graceful softphone registration may work between devices running the same client?
Many thanks for your reply, very much appreciated.
Best
Steve

Any way to see ICONS in Open and Close Dialog boxes?

Is there any way to see ICONS in Open and Close Dialog boxes, instead of just lists of items names?
I know you can preview one at a time, but I'd love to see all my artwork as picture icons, just as you can in a finder window.
Windows does this, but not OSX that I know of. Is there a third party app that enables you to see icon view in the Open and Close Dialogs?

Thank you Steve Jobs. I guess that feature has been added in Leopard. Now there really is nothing that I'll miss from Windoze.

Open and Save Dialog Boxes Appear Briefly then Close

Open and save dialogs on my MDD 867 flash briefly and then close themselves in my administrator account. All the other accounts are fine. This makes me suspect something in the Library. I just did a wipe and install plus Migration Assistant to no avail--which furthers my suspicion of Library problems.
Anyone have any suggestions which file(s) could be the culprits?

More than likely it's a conflict/corrupted amongst the various preference (plist) files. Log into a good admin account, backup the bad account's folder, delete the bad account, selecting the save data option (which is stored in /Users/Deleted Users/ as a disk image), recreate the bad account using the same username/password combo, log out and back into the recreated original account. If the problem's solved, open the saved data dmg file in /Users/Deleted Users/, open the /Library/Preferences/ folder from the saved data, open the current /Users/restored account/Library/Preferences/ folder, and slowly copy plist files from the saved data folder to the current one that don't exist in it. Log out and back in to ensure there's no conflict and things still work correctly. Resolving conflicts or identifying corrupt plist files is a laborious process. Good luck.
See http://discussions.apple.com/thread.jspa?messageID=4703015 for more details on this process.

Open dialog box....

Hi Everyone,
I have designed a form in which user has to input the path where he wants to save his report.
Like c:\test.PDF
what i want to do is that when user click a button againts this field a dialog box should appear in which he can select the location, like in Notepad File > Open or File Save.
and after selecting the path and closing the box the path should be display in the text field which i have provid to my user. Is it possible to attach the default file extension automatically in the text box?
like if user have written c:\test it should be automatically converted into c:\test.PDF? Any Idea?
How is this posible in Developer 6i. I will be very thankful if someone can help me solving this problem.
Regards,
Imran

use GET_FILE_NAME - it's documented in the help - client server only

Open and Save dialogs are painfully slow. Is there a solution?

Open and Save dialogs are painfully slow. Is there a solution?

I pulled this part out of one of the topics Barney-15E linked to. It absolutely works! The glacially slow Finder response has been driving me batty in Mavericks. Each time I restarted, it would take forever to list the startup drive when I opened it. Open and Save dialogues were also painfully slow to respond. Now they all snap open and list instantly. I used TextWrangler to comment out the line in the file auto_master, then ran the automount command in Terminal.
Working workaround:
Use the following statements in Terminal.
sudo vi /etc/auto_master
In this file comment out /net with # (#/net .....)
sudo automount -vc
Fixed.
Confirmed here too both for Preview.app Open file sluggishness and Finder window population. Thanks Snaggletooh_DE!
Couple of notes:
(1) Some folks may be more comfortable using a GUI editor like TextWrangler instead of vi
     a) in the Finder use the GO menu and select Go to Folder
     b) type '/etc' ( without single quotes ) in the Go to folder dialog box and press the Go button
     c) Right ( Control ) click the auto_master file in the resulting Finder window. Select "Open with...." and use TextWrangler ( your choice )
     d) Comment out the line by inserting a '#' ( pound sign ) as noted in Snaggletooth_DE's instructions
     e) Save the file ( probably need to authenticate with your admin password )
     f) Do the 'sudo automount -vc' per Snaggletooth_DE's instructions. Will need to authenticate again.
(2) Notice Snaggletooth_DE described this as a "workaround" because it bypasses an Apple bug. Presumably most people have not changed their auto_master file and it worked fine in Mountain Lion and prior. In other words: If you haven't done so already, please continue to submit feedback and bug reports to Apple for this issue.
Kudos to Snaggletooth_DE for figuring out code is trying to look at network ( NFS ) volumes that maybe don't exist.

I am trying to open CR2 files from a Cannon EOS 1DX camera in Photoshop CS6 and I have already updated the Camera Raw Plug in but Photoshop is still giving me the cannot open files dialog box? Help?

I am trying to open CR2 files from a Cannon EOS 1DX camera in Photoshop CS6 and I have already updated the Camera Raw Plug in but Photoshop is still giving me the cannot open files dialog box? Help?

Canon 1DX support was added to ACR in version 6.7, 7.1. An updated CS6 should be at level ACR 8.6. If your ACR is not at level 8.6 try using CS6 menu Help>Updates. If that does not install ACR 8.6 try downloading the ACR and DNG 8.6 converter and see if ACR 8.6 will install into your CS6.
Adobe - Adobe Camera Raw and DNG Converter : For Windows
Adobe - Adobe Camera Raw and DNG Converter : For Macintosh

When I try to open photoshop I get a "help adobe improve it's produts" dialog box every time and it freezes

When I try to open photoshop I get a "help adobe improve it's produts" dialog box. No matter what I do, I get this box when I try to open a file in photosop never opens although it appears to be

Hi Steadyalberta,
Something might be preventing Photoshop from being able to save your selection on the dialog window properly. You might try right clicking and choose Run as Administrator on the Photoshop icon when launching (Windows) or test if the behavior persists when using a newly created admin user account.
Hope that helps,
- Dave

Why doesn't firefox close out when I'm done with it and close it; when next I try to open firefox again I get a dialog box telling me firefox is open and to close it or restart my computer? I'm tired of doing that one of more times a day.

1. This happens at least once a day, and sometimes multiple times. Today it's happened twice, thus far. My home page is FoxNews.com and if I leave it up overnight or for longer periods during the day firefox doesn't allow the refresh of the home page it just goes to yahoo 404--error can't find the webpage. Then when I enter FoxNews.com it adds a /error to the url and doesn't allow me to go to MY homepage. So I close firefox and then if I try to reopen it and then I get the dialog box saying firefox is open and to close it or restart my computer. Very aggravating. 2. Also the Adobe flash drive update always fails. 3. Webpages get in loops and fail to load to the point I have to stop and refresh or close out and reopen.

One situation in which Command+w or Ctrl+w might not work is if the "focus" is in a plugin such as the Flash player used on Youtube after you interact with the player controls. Firefox will continue to send the keyboard input to the plugin until you move the focus back to the page. Does that account for any of the problem?

CUCM 8X to Dialogic CG6000

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