TMSPE Dial Plan Configuration in TMS

Similar Questions

• C60 dial plan

  Hi all

  currently I am deployment with codec 60 vcs in our customer.

  they have endpoint existing videoconferencing (polycom) in the other site.

  Topologi is like this:

  Codec - point of video endpoints (register) - VCS-(WAN)-

  video endpoints in the other site may not register for VCS, usually connects per IP.

  My question is, if I register codec 60 for VCS (using H.323 or SIP), how can one codec can still dial the other end (by IP) is not to sign up for VCS?

  and also, the other endpoint can call the 60 codec using intellectual property?

  I know that the codec can call a point to use the intellectual property, but if the codec is registered in VCS, I always confuse in the configuration of the dial plan.

  Is my question is clear, or need some explanations?

  y at - it an idea?

  Thank you

  Kind regards

  Anju Josua

  It is important that there is no nat or video ports closed by a firewall between the involved Codec, VCS - C and the Polycom video end point.

  In general, the VCS - C is set to place calls ip address unknown through the VCS - E (see the deployment of vcs - c/e guide).

  VCS need to know the internal IP is directly accessible.

  Happening if endpoint registers with the SCV - C (is there a reason why you do not simply save

  the end of polycom in your VCS - C?)

  The other option is to add a void area local based ip where you add your internal networks (or just ip endpoints)

  so the VCS can directly call endpoint.

  Please remember useful frequency responses and identify useful or correct answers.

• Area code to Dial Plan via the 504G 7.5.4 file xml

  I can't get my phone to add a default when area code automatically scheduled via the .xml file.

  I use a Cisco SPA504G 7.5.4 current firmware version

  Commissioning is performed using tftp.

  Trying to get the phone to add '1805' when the 7 digits are composed.

  E.g.   Dial 555-1212 have phone send the call to 1-805-555-1212

  Via the web interface, I can add that in and it works perfectly, see below:

  (* xx. | * xx. |) [3469] 11. 0 | 00 | <:1805>xxxxxxx | 1xxx [2-9] xxxxxxS0: xxxxxxxxxxxx.)

  However, adds the same <:1805>in the file xml does not work. The phone will not download the xml file.  I'm guessing that the '<" and="" "="">' are breaking somehow however I tried several different configurations oh spaces with no luck.  I confirmed that my config file loads successfully, so the only line that change is specifying the numbering plan and after changing this line unique, does not load the file. Here is a list of the lines I've tried.

  (* xx. | * xx. |) [3469] 11. 0 | 00 | <:1805>xxxxxxx | 1xxx [2-9] xxxxxxS0: xxxxxxxxxxxx.)

  (* xx. | * xx. |) [3469] 11. 0 | 00 | <:1805>xxxxxxx | 1xxx [2-9] xxxxxxS0: xxxxxxxxxxxx.)

  (* xx. | * xx. |) [3469] 11. 0 | 00 | <:1805>xxxxxxx | 1xxx [2-9] xxxxxxS0: xxxxxxxxxxxx.)

  (* xx. | * xx. |) [3469] 11. 0 | 00 | <:1805>xxxxxxx | 1xxx [2-9] xxxxxxS0: xxxxxxxxxxxx.)

  (* xx. | * xx. |) [3469] 11. 0 | 00 | <: 1805="">xxxxxxx | 1xxx [2-9] xxxxxxS0: xxxxxxxxxxxx.)

  (* xx. | * xx. |) [3469] 11. 0 | 00 | < :1805=""> xxxxxxx | 1xxx [2-9] xxxxxxS0: xxxxxxxxxxxx.)

  (* xx. | * xx. |) [3469] 11. 0 | 00 | < :1805=""> xxxxxxx | 1xxx [2-9] xxxxxxS0: xxxxxxxxxxxx.)

  (* xx. | * xx. |) [3469] 11. 0 | 00 | < :1805=""> xxxxxxx | 1xxx [2-9] xxxxxxS0: xxxxxxxxxxxx.)

  I know I'm just playing with spaces, but I'm not having any luck.  I also tried these same variations with substitute the x first after the [2-9] code with no luck

  Help?

  It is standard XML quoting:

  -

  >   -   >

  So, something like:

  (* xx. | * xx. |) [3469] 11. 0 | 00 | <:1805>xxxxxxx | 1xxx [2-9] xxxxxxS0: xxxxxxxxxxxx.)

• Need help basic Dial Plan

  Hello

  I have a dial enough (basic) issue of regime.  I need all incoming calls to the PSTN to directly access our standard of Auto number express unit which is currently extension 502.  I thought I had set up properly in CME (8.6), but I get a message that the number is not reached.  Any help is appreciated as I may have missed something. Here is the abbreviated Config:

  Current configuration: 12212 bytes
  !
  ! Last configuration change to 14:11:35 PST Tuesday, August 26, 2014 by routeradmin
  ! NVRAM config last updated at 14:11:37 PST Tuesday, August 26, 2014 by routeradmin
  ! NVRAM config last updated at 14:11:37 PST Tuesday, August 26, 2014 by routeradmin
  version 15.1
  horodateurs service debug datetime msec
  Log service timestamps datetime msec
  encryption password service
  !
  hostname voicerouter
  !
  boot-start-marker
  boot-end-marker
  !
  !
  map of type t1 0 0
  Select the secret 4 t3Ic8qb5ewMDWvTa4L9l1WMKprop/NnpCj9WtdkqGGc
  !
  AAA new-model
  !
  !
  AAA authentication login default local
  AAA authorization exec local logon
  AAA authorization network local logon
  !
  !
  !
  !
  !
  AAA - the id of the joint session
  !
  clock timezone PST - 8 0
  clock to summer time recurring PST
  network-clock-participate wic 0
  network-clock-select 1 T1 0/0/0
  !
  No ipv6 cef
  IP source-route
  IP cef
  !
  !
  !
  DHCP excluded-address IP 192.168.8.1 192.168.8.19
  !
  IP dhcp pool owhvoip
  network 192.168.8.0 255.255.248.0
  default router 192.168.8.1
  option 150 ip 192.168.8.1
  Rental 30
  !
  !
  !
  Authenticated MultiLink bundle-name Panel
  !
  !
  !
  !
  primary ISDN switch type - or
  !
  Crypto pki token removal timeout default 0
  !
  !
  voice-card 0
  dspfarm
  DSP services dspfarm
  !
  !
  !
  voip phone service
  h323 connections allow h323
  allow connections h323 to SIP
  allow connections sip h323
  allow sip to sip connections
  Fax protocol t38 ls-redundancy version 0 0 hs-redundancy 0 help none
  !
  the voice of class custom cptone jointone
  Conference of two colors
  frequency 600 900
  300 150 300 100 300 50 Cadence
  !
  !
  !
  !
  translation of the voice-rule 1
  rule 1 /9400/ /502/
  !
  voice translation-rule 2
  rule 1. * / /5415489400/
  !
  !
  voice translation-profile Inbound_Calls_To_CUE
  translate 1 called
  !
  Local-ILD voice translation-profile
  definition of call 2
  !
  !
  license udi pid CISCO2911/K9 sn FTX1641AHX3
  HW-module pvdm 0/0
  !
  HW-module sm 1
  !
  !
  !
  username routeradmin password 7 091649040910450B 41
  username privilege 15 password 7 cmeadmin 03104803040E375F5E4D5D51
  !
  redundancy
  !
  !
  !
  !
  controller T1 0/0/0
  long CableLength 0dB
  time intervals PRI - Group 1-10, 24
  !
  security of the zone of confidence
  the internet security zone
  zone-pair security trusted internet source of internet destination
  zone-pair source security not reliable-is trust internet destination trust
  !
  !
  !
  !
  !
  !
  !
  interface Loopback0
  IP 192.168.17.1 255.255.248.0
  !
  the Embedded-Service-Engine0/0 interface
  no ip address
  Shutdown
  !
  interface GigabitEthernet0/0
  Description Internet
  DHCP IP address
  no ip redirection
  no ip proxy-arp
  Shutdown
  automatic duplex
  automatic speed
  !
  interface GigabitEthernet0/1
  IP 192.168.8.1 255.255.248.0
  automatic duplex
  automatic speed
  !
  interface GigabitEthernet0/2
  no ip address
  Shutdown
  automatic duplex
  automatic speed
  !
  interface Serial0/0/0:23
  no ip address
  encapsulation hdlc
  primary ISDN switch type - or
  ISDN incoming-voice
  No cdp enable
  !
  the integrated-Service-naturel1/0 interface
  IP unnumbered Loopback0
  the ip address of the service module 192.168.17.2 255.255.248.0
  ! Application: CUE running on NME
  Service-module ip default gateway - 192.168.17.1
  No keepalive
  !
  IP forward-Protocol ND
  !
  IP http server
  local IP http authentication
  no ip http secure server
  IP http path flash:/cme-gui-8.6.0
  !
  IP route 192.168.17.2 255.255.255.255 integrated-Service-naturel1/0
  !
  !
  !
  !
  !
  !
  flash TFTP server: apps31.9 - 3 - 1ES26.sbn
  Flash: term31.default.loads TFTP server
  flash TFTP server: SCCP31.9 - 3-1 SR 4 - 1s .loads
  Server TFTP flash: jar31sccp.9 - 3 - 1ES26.sbn
  flash TFTP server: dsp31.9 - 3 - 1ES26.sbn
  Server TFTP flash: cvm31sccp.9 - 3 - 1ES26.sbn
  flash TFTP server: cnu31.9 - 3 - 1ES26.sbn
  !
  !
  !
  control plan
  !
  !
  voice-port 0/0/0:23
  !
  !
  !
  profile MGCP default
  !
  SCCP local GigabitEthernet0/1
  SCCP ccm 192.168.8.1 identifier 1 version4.0
  SCCP
  !
  SCCP ccm group 15
  associate the ccm 1 priority 1
  register the associated profile 1 CFBOWH
  attempts of KeepAlive 5
  !
  Dial-peer voice voip 500
  destination-model 5...
  session protocol sipv2
  session target ipv4:192.168.17.2
  SIP DTMF-relay-notify
  Codec g711ulaw
  No vad
  !
  voice pots Dial-peer 10
  translation-profile entering Inbound_Calls_To_CUE
  !
  Dial-peer voice 20 pots
  translation-profile outgoing Local-LTD
  destination-style 9...
  port 0/0/0:23
  !
  !
  !
  !
  access controller
  Shutdown
  !
  !
  phone service
  Protocol ipv4 mode
  conference material
  Department of Health-folder-buffer 90
  No auto-reg ephone
  authentication credentials cmeadmin tshbavsp$ $4
  Max-joined 50
  Max - dn 200
  IP source address 192.168.8.1 port 2000
  dnis dir-search service
  delays of transfer-reminder 30
  Oregon's wild harvest message system
  URL of http://192.168.17.2/voiceview/common/login.do services
  URL authentication http://192.168.8.1/CCMCIP/authenticate.asp
  location of the cnf file flash:
  load 7931 SCCP31.9 - 3-1 SR 4 - 1s .loads
  zone schedule-5
  time format 24
  voice mail 500
  MAX conferences 8-6 win
  application of system call-Park
  ground of appeal forwards. T
  Department of health flash: / moh.wav
  Web admin system name cmeadmin secret $5 1$60ro$u.0r/cno/OD2JmtvPq4w9.
  DN-webedit
  transfer-figures-collect-orig-call
  transfer full-consult system
  transfer-model. T
  Standard AEC
  create the files-cnf version-stamp 7960 26 August 2014 12:49:29

  Hello

  Can you try to add your pots peers 10 dial

  !
  voice pots Dial-peer 10
  / / DESC * MATCH INCOMING CALLS *.
  translation-profile entering Inbound_Calls_To_CUE
  incoming called-number.
  port 0/0/0:23
  !

  Then repeat the test

  Concerning
  Alex

• Conductor BBFRR planned Conference in TMS - granularity alias missing

  Hi all

  I've been supported configuration for regular calls of telepresence with driver X2.1 and TMS 14.2.2.

  Everything is fine - the only thing miss me, and I wonder about is regex support to the boss of the Alias in aliases for telepresence driver tab in TMS.

  The only variable that actually "is" is the % sign. It corresponds to the 5 channels of digit numbers starting with 10000.

  Is it possible to use 'normal' regexes here as in CUCM bosses or VCS search rules?

  I wish I could write an E.164 range has for example, as [email protected] / * / or

  40955511\[email protected] / * / -while I have 100 alias in this form.

  So far, all my efforts proved fruitless, as everything that I entered as a model of Alias, ends with a completely ridicilous regular expression on the far right, which makes me think of nothing is supported except the % sign. A way to change or modify it?

  Support of all thoughts of eut or TAC?

  Anyone with a similar experience?

  If it doesn't "stick" I guess I'll have to find an exotic way to the use of models of translation.

  Another question:

  With BBFRR, this alias tab alias PTC Conference is one that is driven by XML endpoint?

  Thank you

  Mihail

  Hi Mihail,

  In TMS 14.1 and 14.2, it is hard coded as '%' grows to 10000. However, this was changed to 14.3, which will allow us to release later in July. The draft of 14.3 release notes:

  

  Kjetil-

• Change of dial in number of TMS for video bridge

  A user request that the wheel (SIP) number of their session video bridge be a specific number.  After you create the meeting, you can go to the TMS conference settings and change the number.  However, it is not actually change.  The number he has been created with is the single dial that will work.  You can go to the MCU and change there, but it shows that the meetings which are currently spend.  This specific meeting is recurring and not for a few days.

  I was curious to know if there is something missing with TMS, that it is not passing the new number to the MCU correctly.  Any help is appreciated.

  Thank you

  As much as I know there is no way to change a conference number in TMS yourself to MSDS to connect endpoints specifically to something you want.  I tried and got the same results as you, TMS will always use all that it automatically generated for the road.

• Planning of the TMS/head of Orchestra/CUCM

  Hi guys,.

  I'm usually confused as the scheduling how works between CUCM (CM) and Conductor (TC).

  Current configuration:

  CUCM: 10.5

  Conductor: XC3.0.2 and I have lot of vTPS and MCU

  TMS: 14.6.2

  Ad-hoc and Renedevous works between CUCM and conductor, but I can't find any reference (blind) how intergrate CUCM and conductor.

  I currently use the SIP Trunk for appointment to call Schdeuled numbers, it works but once I try to create and trunk identical sip rather by sending to the actual ip address of drivers instead of the virtual IP address, I get the below error:

  Thank you guys.

  Event = "a request arrives on an IP address that is not configured as an appointment or ad-hoc IP address".

  Hello

  You must add two IP address under location, ad hoc and other conference for appointment.

  You must create a SIP Trunk of CUCM to address IP Rendezvous for video calling Rendezvous.

  Create one SIP Trunk, the other pointing to the Ad-hoc IP address and then map it under the conference as resources of ad-hoc conference bridge.

  Steps to follow:

  Go to system > network interfaces > IP and then add two IP address in the same subnet as the IP address of a conductor.

  Refer to page 35 of the document below.

  http://www.Cisco.com/c/dam/en/us/TD/docs/Telepresence/infrastructure/con...

  Now go to go ├a Conference Setup > locations.

  Map IP ad hoc for the Ad Hoc conference settings.

  Same map IP address of appointment for the conference date settings.

  See page 38 and 48 of the same document.

  Note If you select the type of conference both under location then under the same place, you will get two ad-hoc and option address IP Rendezvous.

  Kind regards

  RACLOT

• The planning configuration fails on the new server of Db.

  Hi guys,.
  
  I want to reconfigure my EPM 113 system on new computer server Db, but with the old databases, (restored from the old machine to Db)
  All components have been configured well, except for planning.
  
  Here's the scenario,
  I have 3 of the system,
  A: EPM 11.1.1.3
  B: SQL server 2005
  C: SQL server 2008
  
  Now I want to upgrade DB to import all the basics when re configuring EPM system with system 'C', then 'B' 'C'
  All components have been configured correctly but at configuration time of the workspace in the configurator to connect to the base is oriented planning database.
  so I changed my workspace database and configured it get successfully but after I tried to configure the planning he get failed.
  
  any idea? Did I miss a step?
  
  Here is the content of the journal,
  
  'configtool.log' line: -.
  (June 11, 2012, 16:10:25), com.hyperion.planning.ConfigToolUtils, DEBUG, instance of CSS is
  (June 11, 2012, 16:10:25), com.hyperion.css.CSSSystem, DEBUG, 20:3502: the file HSSLogger.properties is missing.
  (June 11, 2012, 16:11:32), com.hyperion.planning.HspDBConfigurator, ERROR, error happened: 20:3008: unable to connect to the native repertoire.
  (June 11, 2012, 16:11:32), com.hyperion.cis.config.wizard.RunAllTasksWizardAction, ERROR, error:
  com.hyperion.cis.config.ProcessingException
  at com.hyperion.planning.HspDBConfigurator.configure(HspDBConfigurator.java:209)
  at com.hyperion.cis.config.wizard.RunAllTasksWizardAction.executeDbConfigTask(RunAllTasksWizardAction.java:658)
  at com.hyperion.cis.config.wizard.RunAllTasksWizardAction.execute(RunAllTasksWizardAction.java:202)
  at com.installshield.wizard.RunnableWizardBeanContext.run (unknown Source)
  
  
  Kind regards
  Narayan

  You know that 11.1.1.3 is not supported with SQL Server 2008, SQL Server 2008 supports the 11.1.1.4

  See you soon

  John
  http://John-Goodwin.blogspot.com/

• 11.1.2 planning Configuration options

  11.1.2 on my Win 2008 Dev box configuration and has got a menu, I'm not sure what to check
  
  Configurator for EPM for planning, I have a window talking on the activation of each product under license and watch
  
  Capital investment planning
  Public sector planning and budgeting
  Workforce planning
  
  I know that the Public and the workforce are licensing extra on top of planning, is also capital expenditures? We have just a license for the 'planning' and the documentation is not particular to what option to select as appropriate
  
  JTS

  Hello

  Yes, they are all separately licensed, have a read of the price list:-http://www.oracle.com/us/corporate/pricing/price-lists/business-intelligence-price-list-071347.pdf

  See you soon

  John
  http://John-Goodwin.blogspot.com/

• Research of exception with the &lt; reading-writing-support-map-plan &gt; configuration.

  Research of exception with configuring < reading-writing-support-map-plan >, which is set against a simple store database cache implementation. The SimpleCacheEventStoreImpl class implements interface dumps.
  
  Exception in thread "main" java.lang.UnsupportedOperationException: configureCache: reading-writing-support-map-plan
  at com.tangosol.net.DefaultConfigurableCacheFactory.configureCache(DefaultConfigurableCacheFactory.java:995)
  at com.tangosol.net.DefaultConfigurableCacheFactory.ensureCache(DefaultConfigurableCacheFactory.java:277)
  at com.tangosol.net.CacheFactory.getCache(CacheFactory.java:689)
  at com.tangosol.net.CacheFactory.getCache(CacheFactory.java:667)
  at Sample.SimpleEventStoreConsumer.main (SimpleEventStoreConsumer.java:10)
  
  
  The cache store is interfaced with the SimpleEventStoreConsumer program (where I have a sales and licensing option) through the cache configuration descriptor following. The performance of the SimpleEventStoreConsumer, the exception occurs to try to get the named cache to cache factory
  
  
  <>cache-config
  < cache-system-mapping >
  
  <>cache-mapping
  < name-cache > Evt * < / cache-name >
  < scheme name > SampleDatabaseScheme < / system-name >
  < / cache-mapping >
  
  < / cache-system-mapping >
  
  <>- cached patterns
  < reading-writing-support-map-plan >
  
  < scheme name > SampleDatabaseScheme < / system-name >
  
  < internal-cache-system >
  < local plan >
  < system-Ref > SampleMemoryScheme < / plan-ref >
  < / local plan >
  < / internal-cache-system >
  
  < dumps-plan >
  schema < class >
  > class name < com.emc.srm.cachestore.SimpleCacheEventStoreImpl < / class name >
  < init-params >
  < init-param >
  java.lang.String < param-type > < / param-type >
  {cache name} < param-value > < / param-value >
  < / init-param >
  < / init-params >
  < / class-system >
  < / dumps-plan >
  
  < / reading-writing-support-map-plan >
  
  < local plan >
  < scheme name > SampleMemoryScheme < / system-name >
  < / local plan >
  
  
  < / cache-plans >
  < / cache-config >

  Try to wrap your reading-writing-support-map-pattern within a distributed system

  
  
  SampleDatabaseScheme

  
  .........
  .........
  

  ..........
  

• 10-digit dial plan design

  Hi all

  My company is migrating from a 4-digit 10-digit dianplan. The idea is to keep the calls intra-site with 4 digits and called intersite 10-digit.

  The question is should I change the extensions on CUCM 10-digit numbers and add a rule of translation on the bridge to prefix the numbers when a call comes, who also send 10-digit CUCM and add a translation in CUCM when a 4 digits is called prefix the remaining 6.

  or

  Continue to use the 4 digits on number callmanager directory configuration and add a mask of external number, calls intra-site of way still use the 4 digits and there is no need of translations for which...

  Just add a translation on the CUCM for outgoing calls to another site, but in the same cluster (remove the first 6 digits) and another translation for incoming calls to another cluster.

  What do you think is the best?

  Rafael,

  In the circumstances of the dishes is good.  Excellent writing upwards in the direction SRND 8.X...

  HTH,

  Art

• Unity and length Variable Extentions/Dial Plan

  I have an Avaya switch that runs the numbers 4 and 5 extensions and I want to connect to a new unit 4.x for voicemail. Any problem with the extensions of variable length running in unity?

  Also I finally transitioning all users Avaya to a 5 digit numbering plan. All the problems in the future when chaning a 5 4-digit mailbox?

  Also long extensions are unique, you can have lengths without problem - 3, 4, 5 and 10 - go wild.

  No problem change them after the fact. The only restrictions are minimum 3-digit (I forgot the maximum off hand - it's like 30 digits), and they must be unique in the field of composition - that's all.

• Dial plans

  I would like to know how to change my plan of the student plan Photoshop to the student plan all Apps for 15.99 a month. I use a Mac. I need specifically to Illustrator, InDesign, Photoshop, Bridge, and Lightroom

  To the link below, click on the still need help? option in the blue box below and choose the option to chat or by phone...

  Make sure that you are logged on the Adobe site, having cookies enabled, clearing your cookie cache.  If he continues to not try to use a different browser.

  https://helpx.Adobe.com/contact.html?step=ZNA_account-payment-orders_stillNeedHelp

  Concerning

  Stéphane

• Change dialing plan

  I would change my Creative student Cloud and the Professor plan the plan of photography edition.  Can someone please?  Thank you

  Hello Janet,.

  You must cancel your current plan as per the link cancel your creative cloud membership .

  Then, you can register on the different level you want.

  Kind regards

  Sheena

• SPA3102 PSTN to voip

  Hi all
  What am I supposed to hear when the SPA picks up the call?
  My VOIP works and my PTSN works. Ive plugged the socket "phone" on my Fax (which is on the same phone number as adsl, and is set to 'reply' mode) 'line' on my SPA3102 and implemented according to the instructions of JMG (www.jmgtechnology.com.au/spa_3102_guide.pdf). The fax receives faxes. If I dial my phone number of voip phone attached to the rings SPA and everything is good. So: I dial the fax number, the phone in the SPA rings, then it stops. Here, I'm guessing that the SPA should have picked up the call (after the series of 5 seconds). But the phone I cling will tone, that it seems Ive been hung up on. I had once a voice saying: 'Please enter pin followed by the key voice portal sharp code', but that only happened once and the PIN that I had, I put in the SPA did not work. Is that what I'm supposed to hear? What am I supposed to hear when the SPA picks up the call?
  Thank you
  T.

  On the SPA3102, if you 'Ring through line 1', Yes, an incoming call from the PSTN line rings the phone on the SPA3102 to the number of seconds defined in the PSTN answer delay setting.  After the gateway PSTN-to-voip the SPA3102 will get the call.  If you set the PIN then the gateway which will respond to the call and give you three "beeps" as a signal for you to enter a PIN.  If you have authentication set to None and the default dial plan configured to receive dialing number, you will hear a dial tone.  If you have authentication set to None, and the defined default dial plan to automatically dial a voip number that you will continue to hear ringing until the answer to the voip call.

  I don't know what the impact is a device of fax on the PSTN line.