Dial plans

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

• 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.

• 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

• 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

• TMSPE Dial Plan Configuration in TMS

  I work on the migration of TMS Agent legacy at TMSPE, and I noticed the following in the commissioning of TMS options:

  Note: Video Address Pattern and Device Address Pattern must be configured for  provisioning and FindMe to function properly.
  

  I am use to the legacy fill method only in the device URI to create the video address SIP that we use.  However, how can I go of this unique method of configuration to the new address TMSPE, and that each of the new address types are if they were separate or the same?

  Thank you

  Address of the video is the FindMe address, the address of device model is, well, just the address of the device; that is the address of JabberVideo (Movi) users. The address of video template is used only if you use FindMe, otherwise not.

  See: https://supportforums.cisco.com/thread/2179099?tstart=120

  /Jens

• spa112 telemarketing plan dial composed after hangup

  I have an intercom system connected to my SPA112 and use the following dial plan:

  (S0<:752>)

  He composed as expected when I hit the button on the intercom, however, when the call is completed, another call is initiated immediately by the same extension.  I don't know if the intercom is the origin of the problem, or if I can adjust a setting on the SPA112 to solve this problem

  If I use the following dial plan the problem goes away:

  (S1<:752>)

  but then I heard 1 second tone over the intercom before the call is placed.  I'd rather not do this way.

  I use freepbx distro as my PBX.

  I'm out of my lab, so we rely only on my memory now - and Google. Then read it please Configure the values of timer control in the locale of voice on SPA112 and SPA122

  It describes not only how do I deactivate CPC (CPC duration to 0 set), but he see the definition of tone on regional tab. To turn off the tone set length to 0.

• Numbering of the SPA9000 Plans?

  Hello

  Can someone please explain on the spa9000 dial plan

  We have too many fields to fill...

  Explain the

  Call routing rule: where I put the lines and also some numbering plan?

  Then the numbering on the FXS 1 plan

  (xx).

  And then plan on each line numbering...

  Why son much dial plans and how to give a plant correctly.

  Thanks adavance.

  GrilloVillegas

  Visit http://linksysbycisco.com/kb for the SPA9000 dial plans or click on the link below.

  http://Linksys.custhelp.com/cgi-bin/Linksys.cfg/php/enduser/std_adp.php?p_faqid=5179&p_created=11686...

• SPA3000 dial of Voip to PSTN problem

  
  

  Do it with the dial plan.  Set the PSTN Caller default DP a number available and configure the dial plan:

  (S0<:[email protected]>) where [email protected] is the sip uri when you want to transfer the call.

  If it's a PSTN number you want to transfer the call to you must configure tab PSTN line with the credentials of voip to voip account, you want to use to transfer the call.

  Of course, the line 1 tab you set enable component IP: Yes, if you send to a sip uri.

• How to dial directly to an IP address (point to point a.k.?) of the Codec C40

  We are looking to replace old Polycomm and Tandberg units with a telepresence C40 faithful.  On old Polycomm and Tandberg units, we can simply enter the endpoint IP address and connect.

  This unit of C40 has been attached to a CUCM and worked through their guardian.  We have done a factory reset the device, so we can move our network IP "Naked" in an accessible IP address.  However, we cannot make calls: it simply says 'Connection' and is there, doing nothing.

  Troubleshooting-> Diagnostics Web page, said "the call by default protocol is set to Auto, but the system is not registered on this Protocol".

  I guess the C40 can direct an IP address as our previous Polycomm and Tandberg units?

  Should what settings I use as the DefaultCall Configuration?  For the moment, it is 'auto '.

  In Configuration H323 I;

  NAT
  Set to Off.

  Profile 1
  CallSetup in direct mode
  PortAllocation the dynamic value

  Authentication
  LoginName is empty
  Set to Off
  Password is empty

  On the System Configuration in SystemUnit page I ContactInfo Type set to Auto.

  I don't know what I should put in E323Alias.  I guess it can be left blank?

  Help set up for direct IP point to point my calls really appreciated.

  Hi juddernaut,

  These forums seem a little foires right now and you have posted double https://supportforums.cisco.com/discussion/12137686/how-dial-direct-ip-a... (made it myself, because they take an age to the answer). I would like to delete the other post.

  Set the configuration of appeal to force H.323.

  The E.164 alias is a number as an extension system, but unless you have a doorman with dial plan of this won't be necessary and can be left blank. However, its value put something in the name of the system box so that the other side can see appealed.

  You should be able to dial via direct IP - I suspect his tent to dial via SIP and failing. Forcing them to use H.323 should do the trick.

  Chris

• Numbering plan of TMS actions required when changing the VCS

  Hi all

  I will be changing my exclusively H323 to SIP client only VCS dial plan. This will be done to activate the route through VCS - E and also prepare for the migration of their VC on CUCM endpoints at a later date. When I change the dial plan I'll change the end point URIs, aka Multiway and digital beaches assigned to scheduled meetings.

  Can you please help me understand what I need to do to make sure that:

  1. MSD directories are updated with new endpoint aliases; and
  2. Existing meetings are moved to the new number ranges

  Thanks in advance for your advice.

  Telephone directories TMS should update automatically as long as they do not use a source directory manually.  You should see the telephone directory changes some time after you migrate the H323 to SIP endpoints points, depending on how often you have telephone directories to update.

  A regular conference will have to be updated to reflect the change dial plan, you can simply change the Protocol that uses H323 to SIP endpoint in the settings to connect to the Conference.

• Everyone sees $19.99 for the plan of students and teachers

  Who else sees this price $ 19.99? The only option I get trying to change plans is for $29.99. I am a student.

  See screenshot.

  

  Dial plans can you not give right to additional discounts since you already have "exhausted their ' on your previous plan. Contact sales support by chat or phone for a detailed investigation.

  Mylenium

• Assistance in the selection of plan

  Hello!

  I'm trying to choose a plan - I work in a group of marketing to a larger company. We have several licenses CS6 since before and probably will not completely migrate to CC for a year or more. But we will need an AE install now for a specific project, which will stretch for a few months and AE is no not part of our CS6 we probably Design Premium.

  While I was looking for on this site, the dial plan for businesses:

  http://www.Adobe.com/creativecloud/business/teams.html?promoid=KTMBN

  What seems to be the best solution for us right now would be the one on the left. My reasoning is that it doesn't require to register for a full year. But moving forward on the link buy now brings me to the purchase for individuals page. Basically, I'm wondering if it would be good to make the purchase because in reality, it is our business, to purchase and this license, in particular, is not listed on the company tab.

  Thanks in advance,

  Nicklas

  You can use a license 'personal' use 'business'... an idea/caveat is that you should think about creating a professional email address then the Adobe ID associated with your purchase belongs to the company, not an employee who can leave the company, and then you have more access to any software related to this "private" e-mail