Number of conversion rules in a voice gateway

Similar Questions

• Create different conversion rules in SRST mode

  Hello world

  Have you ever tried to create rules of different translations for 'normal behaviour' and 'Behaviour SRST'?

  This is my problem:

  In my company, we use waiting them console ARC for the reception, so on my H.323 gateway, I have the following translation-rule:

  translation of the voice-rule 1

  rule 1 / ^ $/ /850/ (because the provider sends just the extension is empty when you call the reception)

  The 850 leads to a CTI route point in CUCM, then the call goes to the CRA can.

  But when the CUCM is broken and phones of failover mode SRST, reception is not available in addition to the outside, because routing CTI 850 is no more accessible. Thus, when SRST mode, I stand at the door of entry use this translation:

  rule 1 / ^ $/ /500/

  so that calls to the home go to the actual DN of the receiving phone.

  It is a kind of a conditional routing... (if mode normal, like this, but in SRST translate, translate like that...)

  Have you guys tried to do something like that?

  I've heard of conversion rules of voice in SRST mode (in call-manager-backup mode), but it doesn't seem to work...

  Good day

  Rémi

  Use the command alias under SRST.

  Chris

  Sent by Cisco Support technique iPhone App

• Video support for voice gateway router can call?

  Hi all

  My gateway voice (Cisco 2921-V/K9) router to connect to the PSTN using SIP Trunk provider. CUCM 11 enter my ipphone and videoconferencing (SX20).

  If someone who using video-conferencing (SX20) to call my (SX20) videoconferencing via dial my number RTC, it only not Visual voice call of appeal.

  I have question: If the router voice gateway support video call?, if it can how to configure and there is an additional module?

  Thank you

  Cyriac

  Hi Cyriac,

  For a video call from point to point, you can see the "Configure the video telephony" section of the following link

  http://www.Cisco.com/c/en/us/TD/docs/voice_ip_comm/CUCM/Admin/9_1_1/CCMS...

  If video conferencing Setup check the following "feature Deprecation announces for video Conferencing and transcoding using PVDM3 on ISR G2 product Bulletin"

  http://www.Cisco.com/c/en/us/products/collateral/unified-communications/...

  Manish

• Cisco Voice Gateway with FXO of Telco, operation support for IP phone

  Hi all

  A very quick question.

  Incoming calls via telephone in a voice gateway company ending on a Cisco 7941 G.

  Is it possible for a Cisco 7941 IP handsets or others to instagate actuation when the call is active, then causes the RTC line provide the new key, and then a Telco on via DTMF the combined IP instagate transfer to another number.

  This feature works on a PABX Ericcson previously!

  Is hookflask taken analog device support ONLY?

  I think that the GUY supports, anyone got it working on CallManager?

  Concerning

  Andrew

  Fix Andrew, actuation on IP phone is supported only with the MAN not with CUCM.

  HTH

  GP.

  Pls rate useful messages!

• No info to display the conversion rules

  Hi all
  
  I have a simple FDM 11.1.1.2 application, import SQL (of an Oracle table) and for the most explicit mappings.
  Some scripts in the map, no logic accounts etc.
  
  When I click on conversion rules of show on the amount of export, it shows a blank screen.
  
  Any suggestions, what causes this?
  
  Thank you
  
  Marc

  Your FDM application is an Oracle DB right?

  When you insert records, you should take advantage of each Member of the source since Oracle is case sensitive. I recommend UCASE()

  Published by: TonyScalese on November 29, 2011 07:14

• Create a script pour changer a voice-conversion rules

  Hi all,

  I have a client who wants to when he is on leave and that its closed society the opportunity to redirect all calls arriving on its head-end Agency agreed to take messages.

  I told him he can call me before leaving me to but not pour of suitable ca him... he wants to be the Girardon him even.

  I wonder therefore if there is the possibility to create a script that will initiate a telnet on the CPU connection and back orders I would do if the customer called me...

  Thanks in advance,

  Roman

  Hello Romain, and I salvation also Eric who is interested in the solution.

  Pour to do what you want to do, we will use 2 tools: PLINK.exe and the AUTOIT.

  TUS plink.exe

  http://www.chiark.greenend.org.uk/~sgtatham/PuTTY/download.html

  PLINK is the version of PUTTY console.

  PLINK will help us to establish administration sessions encrypted with our equipment.

  This is the command-line so big advantage, UN script can therefore include it in. That's what we do.

  Copy the utility to a directory on your disk.

  Œuvres a session console in this directory and launch a SSH session on the UC500 (user id ADMIN and address IP UC 500 = uc500_IP_ADDR).

  Plink - ssh [email protected] / * / _IP_ADDR

  Œuvres administrator.

  OK on one our SSH client.

  TUS now more AUTOIT.

  http://www.autoitscript.com/AutoIt3/index.shtml

  Download the ZIP package version. How you install anything on your PC. Unzip the package and go.

  AutoIt is a suite of tools absolutely magical (the word is weak).

  With Autoit can completely automate any Windows application. And without exception.

  This script amendments tool of will automatically launch windows programs and in addition to simulate user action in the application. And this by moving the mouse, buttons of the clicking on and typing while reading the contents of the windows, and well tempered by typing keyboard to the. And all instead of user.

  On can USE autoit pour to allow any to answer E-Mails only PC!   (Make a PC completely human?   )

  Just imagine the power of such a tool.

  I use it for everything and anything.

  And in this case, I use pour automate connections admins on Cisco equipment.

  You'll find en example a script that allows to connect in SSH to your favorite UC 500. This script displays the result of a show version.

  Here's the script.  call the session_admin_1.au3

  EARLY SCRIPT ===>

  $ip_addr = '192.168.10.1 '.
  $user = "cisco".
  $password = "cisco".
  $enable_password = "cisco".
  $commande = "plink - ssh" & $ip_addr
  ;
  Run (@ComSpec & "/ c" & $commande, "", @SW_MAXIMIZE)
  ;
  Sleep (500)
  Send ($User)
  Send ("{Enter}")
  Sleep (500)
  Send ($Password)
  Send ("{Enter}")
  Sleep (500)
  Send ("Enable")
  Send ("{Enter}")
  Send ($enable_password)
  Send ("{Enter}")
  Send "(terme len de 0") "
  Send ("{Enter}")
  Send ("see the version")
  Send ("{Enter}")
  Sleep (500)

  ;===> END SCRIPT

  Copy in the same directory as PLINK.exe.

  And copies also in the same directory the package AUTOIT AUTOIT3.exe utility

  Pour the script of this Tester

  1 / copy your script into the directory where the utility autoit3.exe
  2 / works a session console in this directory
  3 / type the following command: c:\>repertoire_autoit\session_admin_1.au3 autoit3

  And normally you need to see an educational SSH session on your CPU and see the version displayed on the screen.

  The syntax of the commands in the script is quite simple to understand and very well before in the help file provided in the package.
  I think that you won't hurt to adaptar to your needs.

  Will notice me instructions pour to login at UC are 500 it contained in variables at the beginning of the script.

  If so wishes display windows question marks for the administrator, to ask him to type this information on can adapt the script as a suit.

  EARLY SCRIPT ===>

  $ip_addr = InputBox ("Login", "IP address?","", "", 200 150 ")
  $user = InputBox ("Login", "Administrator?", "","", 200-150)
  $password = InputBox ("Login", "Password?", "","", 200-150)
  $enable_password = InputBox ("Login", "Enable password?", "","", 200-150)
  $commande = "plink - ssh" & $ip_addr
  Run (@ComSpec & "/ c" & $commande, "", @SW_MAXIMIZE)
  Sleep (500)
  Send ($User)
  Send ("{Enter}")
  Sleep (500)
  Send ($Password)
  Send ("{Enter}")
  Sleep (500)
  Send ("Enable")
  Send ("{Enter}")
  Send ($enable_password)
  Send ("{Enter}")
  Send "(terme len de 0") "
  Send ("{Enter}")
  Send ("see the version")
  Send ("{Enter}")
  Sleep (500)

  ;===> END SCRIPT

  This allows not to let the admin passwords into a file readable in clear on the workstation.

  But the coup-ins me the first script and what I do, is to transform the script en executable with the AUT2EXE.exe utility.

  So passwords are hidden inside "l" executable. And they are no longer visible.

  Then the script launches directly clicking on it without needing the utility 'autoit.exe '. During the generation of the 'executable,
  on can choose its icon.

  So as you understand, I'll let you make your scripts yourself. A first script to make the configuration change, then another pour back
  the configuration in the initial state.

  It goes without saying that to write the scripts correctly! We can well imagine the power of the thing. Create executable files from the scripts
  is a good way to lock their use to a single use. On may, if necessary, make scripts more complex pay allow them to more controls. Let me know if it is necessary.

  The scope of Autoit is huge! Can be used to create complete applications. I will probably have the opportunity again to illustrate it in tea items.

  Want current month.

  Patrick

• SIP Dialer with voice gateway using the FXO Ports

  Hello

  We have a laboratory UCCE consisting of an outgoing SIP dialer and a h323 2811 voice modem router with 2 FXO ports. The Dialer reserve an agent and sending the call to the bridge, but the call fails immediately with the bridge return a "404 not found" error to the Dialer. Call directly from a CUCM extension attempts to connect successfully through the FXO. Any thoughts?

  IOS Ver - T4 15.1 (3)

  UCCE worm - 8.5.4

  The gateway configuration:

  version 15.1
  horodateurs service debug datetime msec
  Log service timestamps datetime msec
  no password encryption service
  sequence numbers service
  !
  hostname LAB-RS-ING_VXML-GW
  !
  boot-start-marker
  boot-end-marker
  !
  !
  logging buffered 50000000
  no logging monitor

  !
  No aaa new-model
  no location network-clock-participate 1
  !
  voice-card 0
  !
  voice-card 1
  dspfarm
  !
  dot11 syslog
  IP source-route
  !
  !
  IP cef
  !
  !
  !
  IP host mediaserver 10.15.80.83
  No ipv6 cef
  Authenticated MultiLink bundle-name Panel
  !
  !
  !
  !
  !
  !
  Circuits-port FXO group
  hunting-sequential diagram
  !
  !
  !
  voip phone service
  No IP trust to authenticate
  h323 connections allow h323
  allow connections h323 to SIP
  allow connections sip h323
  allow sip to sip connections
  no service additional sip refer
  service additional media - renegotiate
  signs before any
  H323
  Modem passthrough codec g711ulaw nse
  SIP
  interface FastEthernet0/0 source control binding
  bind media source interface FastEthernet0/0
  header-passage
  !
  voice class codec 1
  g711ulaw codec preference 1
  preferably 5 codec g729r8
  !
  vocal h323 class 1
  H225 timeout tcp establish 3
  !
  !
  !
  !
  translation-article 99 of the voice
  rule 1 / ^ 4 / /334/
  !
  !
  voice translation-profile PROFILE_TO_CVP
  translate called 99
  !
  !
  Crypto pki token removal timeout default 0
  !
  !
  !
  !
  Archives
  The config log
  hidekeys
  !
  !
  !
  !
  !
  !
  !
  interface Loopback0
  255.255.255.255 IP address 172.20.1.1
  !
  interface FastEthernet0/0
  IP 255.255.255.0
  automatic duplex
  automatic speed
  !
  interface FastEthernet0/1
  IP 255.255.255.0
  Shutdown
  automatic duplex
  automatic speed
  !
  !
  Router eigrp 15
  10.0.0.0 network
  Auto-resume
  !
  IP forward-Protocol ND
  !
  no ip address of the http server
  no ip http secure server
  !
  IP route 0.0.0.0 0.0.0.0 10.x.x.x
  !
  !
  !
  !
  control plan
  !
  !
  voice-port 0/0/0
  Ports FXO 1 trunk group
  surveillance cut dualtone Mid-communication
  output attenuation - 3
  no echo - cancel enable
  No non-linear
  No vad
  broadcast-maximum: 250
  broadcast / nominal 200
  minimum of playout / high-delay
  broadcast-delay mode fixed
  call waiting times - disconnect 5
  timeouts wait-version 5
  connection ÉRA 4930
  4930 description
  !
  voice-port 1/0/0
  2 ports FXO trunk-group
  surveillance cut dualtone Mid-communication
  output attenuation - 3
  no echo - cancel enable
  No non-linear
  No vad
  broadcast-maximum: 250
  broadcast / nominal 200
  minimum of playout / high-delay
  broadcast-delay mode fixed
  call waiting times - disconnect 5
  timeouts wait-version 5
  connection ÉRA 4198
  4198 description
  !
  voice-port 0/0/2
  !
  voice-port 0/0/3
  !
  !
  !
  profile MGCP default
  !
  !
  voice POTS dial-peer 1
  incoming called-number.
  direct line to inside
  !
  Dial-peer voice 2 voip
  session protocol sipv2
  incoming called-number.
  codec voice-class 1
  no class voice sip reset timer expires 183
  DTMF-relay h245 alphanumeric
  No vad
  !
  Dial-peer voice voip 4198
  destination-model 4198
  session target ipv4:10.15.80.81
  codec voice-class 1
  h323 voice-class 1
  voice-class sip rel1xx taken in charge "100rel.
  no class voice sip reset timer expires 183
  DTMF-relay h245 alphanumeric
  No vad
  !
  voice pots Dial-peer 301
  trunkgroup FXO Ports
  Description * MIA local calls *.
  destination-model 305 [2-9]...
  alert progress_ind activate 8
  progress_ind enable progress 8
  direct line to inside
  prefix 305
  !
  Dial-peer voice voip 4930
  destination-model 4930
  session target ipv4:10.15.80.81
  codec voice-class 1
  h323 voice-class 1
  voice-class sip rel1xx taken in charge "100rel.
  no class voice sip reset timer expires 183
  DTMF-relay h245 alphanumeric
  No vad
  !
  Dial-peer voice voip 7000
  Description of Agents
  destination-model 7...
  session protocol sipv2
  session target ipv4:10.15.80.81
  codec voice-class 1
  voice-class sip rel1xx taken in charge "100rel.
  no class voice sip reset timer expires 183
  !
  Dial-peer voice voip 9999
  Phone agent test description
  destination-model 9999
  session protocol sipv2
  session target ipv4:10.15.80.81
  codec voice-class 1
  voice-class sip rel1xx taken in charge "100rel.
  !
  Dial-peer voice voip 1444
  Phone agent test description
  destination-model 1444
  session protocol sipv2
  session target ipv4:10.15.80.81
  codec voice-class 1
  voice-class sip rel1xx taken in charge "100rel.
  !
  voice pots Dial-peer 303
  trunkgroup FXO Ports
  Description * LD calls *.
  destination-pattern 1 [2-9]...
  alert progress_ind activate 8
  progress_ind enable progress 8
  direct line to inside
  Forward-digit 11
  1 prefix
  !
  voice pots Dial-peer 302
  trunkgroup FXO Ports
  Description * MIA local calls *.
  destination-model 786 [2-9]...
  alert progress_ind activate 8
  progress_ind enable progress 8
  direct line to inside
  prefix 786
  !
  voice pots Dial-peer 9
  destination-model 91 [2-9]...
  alert progress_ind activate 8
  progress_ind enable progress 8
  direct line to inside
  port 0/0/0
  Forward-digit 11
  !
  !
  Dial-peer voice 1400 voip
  Description of Agents
  destination-model 14...
  session protocol sipv2
  session target ipv4:10.15.80.81
  codec voice-class 1
  voice-class sip rel1xx taken in charge "100rel.
  !
  !
  !
  !
  !
  Line con 0
  line to 0
  line vty 0 4
  opening of session
  transport of entry all
  !
  No Scheduler allocate
  end

  Do you have installation trunks in call to the dialer Manager?

• Video conferencing does not work with the voice gateway PVDM3?

  Hello world

  I'll create a meetme conferencing. However, what config it use VGA, audio conference is only allowed

  Here's the material:

  • PVDM3-32 x 2
  • VIC2-2FXO
  • VIC3-2FXS/DON'T
  • 9971 ip phone x 3

  Here is the config VG (work),

  VG (config) #.
  !
  voice-card 0
  dsp-booking voice-service 6
  !
  dspfarm profile 1 homogeneous conference video
  Codec g729br8
  Codec g729r8
  Codec g729abr8
  Codec g729ar8
  Codec h264 cif-cadence 15 bitrate 128 Kbps
  maximum of participants at the conference 4
  maximum 1 x 1 layout
  maximum sessions 1
  associate the PCRS application
  !

  Here is the config VG (failure),

  VG (config) #.
  !
  voice-card 0
  dsp-booking voice-service 6
  !
  dspfarm profile 1 homogeneous conference video
  Codec g729br8
  Codec g729r8
  Codec g729abr8
  Codec g729ar8
  Codec h264 vga-cadence 30 bitrate 1 MB/s
  maximum of participants at the conference 4
  maximum 1 x 1 layout
  maximum sessions 1
  associate the PCRS application
  !

  Here is the result of the test with different codec

  • OK - h264 qcif-cadence 15 bitrate 128 Kbps
  • OK - h264 cif-cadence 15 bitrate 128 Kbps
  • Fail - h264 cif-cadence 30 bitrate 704 Kbps
  • Fail - h264 vga-cadence 30 bitrate 1 MB/s

  When you use the Cisco.com DSP calculator ,
  1 profile of videoconferencing
  Session number: 1
  number of participants: 4
  Conference type: homogeneous - 1 class
  H.264 VGA: checked

  It returns:

  Allocation of Module DSP:
  Router Slot 0: 16 video (25%) 4 votes (7%) 44 available (68%)
  PVDM Slot 0/0: PVDM3-64

  So, why can't I use the VGA with PVDM3-64?

  Thanks in advance

  Sam

  Do not forget this:

  Announces Deprecation feature video conferencing and transcoding using PVDM3 on the product for the ISR G2 Bulletin

  http://www.Cisco.com/c/en/us/products/collateral/unified-communications/TDM-gateways/bulletin-C25-735945.html

• Calculation of rule DSP for voice routers

  Hi all, I know that Cisco has a DSP calculator, however, there are some values that he asks that I'm not sure of when you try to use it.

  Someone at - it a fundamental rule that they use when they decide how much DSP to add a router to voice FMC?

  I have a Cisco 2911 with 1 VIC2-4FXO, 1 port VIC3-2FXS/DID and currently I have a PVDM3 DSP DIMM 1 with 16 channels. I would add a VIC2-2FXO card for additional analog lines. I read that you need at least to have the same number or channels that you have FXS/FXO ports. If this is the case, then I am currently using only 6 of the 16 channels.

  Here is the command of DSP Group:

  DSP groups on slot 0:
  DSP 1:
  State: UP, firmware: 32.1.2
  Signal/voice of max channel: 16/16
  Max credits: 240, voice credits: 240, video credits: 0
  num_of_sig_chnls_allocated: 6
  Transcoding allocated channels: 1
  Group: FLEX_GROUP_VOICE, complexity: FLEX
  Credit splitting: 100, reserved credits: 0
  Signs of the allocated channels: 6
  Voice of the allocated channels: 0
  Appropriations used (rounded): 0
  Group: FLEX_GROUP_XCODE, complexity: MEDIUM
  Credit splitting: 0, reserved credits: 20
  Transcoding allocated channels: 0
  Appropriations used (rounded): 0
  Group: FLEX_GROUP_CONF, complexity: CONFERENCE
  Credit splitting: 0, reserved credits: 120
  Codec: CONF_G729, maximum of participants: 8
  Sessions by dsp: 4
  Slot: 0
  Idx device: 0
  PVDM slots: 0
  Type DSP: SP2600

  Thank you

  Dan

  Yes, the most basic math are need at least the same number of channels on your DSP (s) that you would have TDM0 channels, which means that you will only use G.711, because the number of channels, G.711. This just for the termination of the TDM.

  With what you have, if you decide to use G.729A/G.722, it would come down to 12 channels, with G.729/iLBC drops to 10, and you get only 3 channels, if you use the ICCS.

• Need help in RegEx - rule Translation simple voice

  I have a Cisco ISR router which performs SIP Cube operations.  When I call gets routed to my Microsoft Lync server set, it must be in E.164 format... which means that I have a + 1 prefixed to all numbers

  Someone can help me to create a voice translation rule that simply takes the input number, prefixes a + 1 on the front and sends it along?

  Here's what I use now, and it works.  But I want a translation rule that I can use for any number.

  voice translation-article 15

  rule 1 /3165551111/ + 13165551111.

  The simplest rule to do this would be:

  rule 1 / / + 1.

• Want to know primary and secondary configuration to the call, Manager with the voice gateways

  Hi all

  Hope you all are doing well, I wanted to know that we have two other PRI service provider and we want one of them are primary and secondary schools on the other. We have two supplier dedicated 4-4 finish lines. Please provide me with the part of configuration that are required in this case and how to re a SP for primary education and another SP for secondary lines.

  Thank you

  Arjun keita

  Hello Arjun,

  For the full bridge configuration you can check the guides below, but for the PRI redundancy, you create dial-peers and specify the feedback:

  https://www.Google.co.in/URL?SA=t&source=Web&RCT=j&URL=http: / / www.cisco...

  Voice POTS dial-peer 1

  Destination-pettern 0 t

  Port 0/0/0:15

  Preference 0

  Dial-peer voice 2 pots

  Port 0/0/1:15

  Preference 1

  Dest-model 0 t

  Dial-peer voice 3 pots

  Description incoming only

  Incoming called-number. T

  Direct inward dial

  Aseem

  (Please rate if useful)

• Do you have conversion rules

  Hello. I am very grateful to read you my post and thank you to everyone helping around. You do a great job.

  I'm currently setting up a CUCM cluster.

  Problem is the following.

  We have an autonomous group with a Publisher and Subscriber

  We have a PRI line with head 2046800 number configured on a gateway, MGCP (2821 router)
  The beach DID for this line is 204-6100 to 204-6899.

  Each IP phone extensions are allocated in the format 6yxx, where y is a number between 1 and 8, and x can be between 0 and 9.

  Thus, all receivers are 4 digits long and corresponding external number is the same extension preceded with 204; for example:
  6100 (internal) extension is 204-6100 (external)
  This configuration is functioning normally as it should and all users are able to dial in and out using their DIDs.

  For example: if I call 204-6123, extension 6123 will ring inside and;
  If extension 6123 calls me on my landline, 203-6123 caller id appears on my phone.

  The problem is the following:
  Management decided to keep only the last 3 digits in the extension number:
  So the first '6' will be chosen for all extensions.
     
  All the new extensions will be between 100 to 899
     
  I tried to create the extension number 101 and he attributed to a phone to test.
    
  (A) problem: the phone loses his DID, in other words, outgoing calls continue with the number of head which is 204-6800, incoming and 204-6101 doesn't get a 'no response '.
  on the side of the appellant.
    
  As a solution, I tried

  (1) to create a model of transformation of 101 at 6101. By that, it was expected that appeals to an external 204-6101
  fixed telephone would ring the 101 post.
         
  This was not the case. The problem remains the same as that described above in problem (a)

  I know that this could be a very small problem and am missing something somewhere. The reason being that I am really new in CUCM and weak in the translation stuff.

  Thanks again.
        
        
        
  Can you suggest a solution please?

  Of course, the translation will be the number that sends it to Telco, eg 7101, then to bottom of the template Page of translation the "called transform mask" 101

• is it possible to get to info on the cost (and period or number) programmatically conversation/SMS?

  Hi, I was wondering - is it possible to get information programmatically:

  -number of SMS Messages sent

  -number of sent mms

  -How long the user has been talkig

  -How much paid for talk/sms/mms-fondant on filling up the user. is it possible to get the user rate programmatically?

  Kind regards

  To count the number of outgoing sms and mms; implement SendListener.

  From the call duration use call log.

  and take the callcost/min, cost/sms, mms/cost as user input to first launch your application. I have thin in this way you can do your necessary things.

  Press the button Bravo thank the user who has helped you.

  If your problem has been resolved then please mark the thread as "accepted solution".

• Number of conversion date throws the error message.

  Hello

  I'm trying to convert a column number in a date column, but get the following error:

  SQL:

  Select to_date(INV_BALANCE_DT_WID,'dd/mm/yyyy') in the W_INVENTORY_DAILY_BAL_F;

  ERROR:

  Error at startup on line: 19 in the command.

  Select to_date(INV_BALANCE_DT_WID,'dd/mm/yyyy') from W_INVENTORY_DAILY_BAL_F

  Error report-

  SQL error: ORA-01861: literal does not match the format string

  01861 00000 - "literal does not format string.

  * Cause: Literals in the entry must be the same length as literals in

  the string format (with the exception of white space).  If the

  "FX" modifier has been switched on, the literal must match exactly.

  with no extra spaces.

  * Action: Fix format string to match the literal.

  SAMPLE DATA:

  INV_BALANCE_DT_WID - 20140101

  DESIRED OUTPUT:

  17 JUNE 15

  Could you please help me on this?

  Thanks in advance!

  The value is 20140101 then why do you use "dd/mm/yyyy '?

  SQL > SELECT TO_DATE(20140101,'YYYYMMDD')
  2 FROM TWO
  3.

  TO_DATE (2
  ---------
  1 JANUARY 14

  SQL >

  SY.

• Catalogue number after conversion of the LT5 to LTCC

  Hello

  I have a serious problem with my catalog after the installation of the LTCC.

  After the conversion of catalogue I see more photos in the right direction. When I see my pictures in the library, that everything seems ok, but when I use develop the function my pictures completely change the colors and appearance. Please take a look below.

  Can someone help me? Thank you for your support with kindness.

  I speak Italian, English and Spanish, so don't worry, I'll understand.

  See you soon

  Liam

  

  Hi LiamDotz,

  • Go to Lightroom > Preferences.
  • Also please go to preferences > Performance tab and uncheck the box "use graphics processor.
  • Restart Lightroom and see if it works.

  Kind regards

  Tanuj