CUPC digit manipulation

With the LDAP sync between AD and CUCM and CUPS, dialing straight from CUPC by right clicking your contacts, will result in dialing the extension as entered in AD under telephone number (or business phone).
Now, many people in our organisation have entered E.164 numbers, have added brackets, spaces etc. in other words, patterns that are not recognised by CUCM. As CUPC is essentially a CTI device, all digit manipulation will have to take place on CUCM. which does not have the flexibility of SIP patterns.
As we have about 4000 users on presence, having these people to change their contact number via our AD front end is a daunting task
Have other people in this forum dealt with this issue?
i know CUCM 7 at least handles the + sign in patterns but that will only partially fix the issue

Check out this doc for the click the call. The application dial rules are perfect for CUPS also. Uses the same premise.
http://www.cisco.com/en/US/docs/voice_ip_comm/cupa/click_to_call/7.0/english/install/guide/C2C2chapter.html
Also search the forum for Application Dial Rules. You will find many comments on the + symbol. In CUCM 6, it ignores the + all together and just looks at the digits. In CUCM, it uses the plus as a digit, etc. So adjust your strings accordingly.

Similar Messages

  • Digit Manipulation on Calling Number

    Dear Netpro,
    I would like to know more on the Digit Manipulations on the Calling Number.I have a Cisco 3640 box passing the Voice Calls and I intend to send a valid but different Caling Numbers ( within our Numbering Range ) to the Interconnected Operators.Pls, can anyone assist with this and suggest some docs, URLs or specific config.I have tried making translation rule but the multiple calls uses the same/single Calling number and I want to have the a different Calling Number everytime a new call is setup ie No two simultaneous calls shd have a same Calling Party Numbers.Pls,assist.
    Thanks in Anticipation.
    Best Regards,

    Chck the below link for digit manipulation on calling no
    http://www.cisco.com/warp/public/788/voip/translation_rule_acd.html
    http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122tcr/122tvr/vrg_r1.htm#2036716

  • Translation Pattern " Inbound Digit Manipulation"

    Hey Everybody ,
    I would like know how to translate all incoming call on 842000 to 2987135. The setup as the following
    1. I have Voice Gateway 2821 with E1 interface
    2. 500 DID ( 2987000 to 2987499 ) are mapped on the E1 line from the exchange
    3. The exchange pointing 842000 to first DID (2987000)then it will hunt 2987001 ---to-- 2987499.
    what i would like to do , it to map all incoming calls on 842000 to ring on 2987135 instead of 2987000.
    It would be appreciated to feed me back with the optimum solution.
    Regards,

    If I understand you correctly here is my thoughts.
    The when someone calls 842000 telco is remapping this to 2987000 and hunting across the other numbers. The problem is that the number getting passed to the circuit is 2987000 not 842000. With that said if you can't get the telco involved I don't think anything can be done. If you can then do this have them not do a roll on the number 842 and included that in your did block. That way you'll see 842000 come in on the circuit at that point you can add a translation partern in cm to send it where every it needs to go.

  • Gatekeeper Digit rewrite

    Hi,
    Does the Cisco Gatekeper any digit rewrite capability to be able to rewrite any incomming call with another prefix before redirecting to another gateway or gatekeeper?
    Thanks in advance
    Hooman Parta

    The current Multimedia Conference Manager (MCM) does not support digit manipulation without the use of an external route server such as NAM. The next generation IOS gatekeeper will support digit manipulation when and IOS H.323 to H.323 gateway is used.
    Digit manipulation is support in the Cisco CallManager and Voice gateways.

  • How to block alternate match on a voicegateway?

    Hello,
    i need configuration support for a voicegateway.
    we have two mainnumbers, which are connected by a BRI-interface and a E1-interface.
    When they call outside with accesscode = 0, they reache the phone provider and get connected,
    But in my example, we have two matching destination-pattern = 0T and .T
    When they forget to dial the accesscode, the BRI-Interface with destination-pattern .T is still working and the number manipulation is wrong.
    We use H323 to CUCM. MGCP is currently not an option for us.
    How can i prevent that on the voicegateway easily?
    voice-port 0/0/0:15
    description *** E1 ***
    input gain 3
    echo-cancel coverage 32
    no comfort-noise
    cptone DE
    bearer-cap 3100Hz
    voice-port 0/1/0
    description *** BRI 1 ***
    no vad
    compand-type a-law
    cptone DE
    dial-peer voice 1 pots
    description *** E1 ***
    tone ringback alert-no-PI
    progress_ind setup enable 3
    destination-pattern 0T
    direct-inward-dial
    port 0/0/0:15
    dial-peer voice 2 pots
    description *** BRI 1 ***
    destination-pattern .T
    direct-inward-dial
    port 0/1/0

    I tried it with a prefix, but this is not helpfull. As soon as the setup from callmanager reaches the voicegateway, i only can use a matching destination pattern, what activates a translationrule. My thinking was to seperate both mainnumbers, but for incoming calls this is not a problem, only for outgoing.
    Peer my understanding, the call is coming from callmanager. This is matching an incoming dialpeer, where i can use translationrules. The same behavior will apply for outgoing dialpeer to phoneprovider.
    My thinking and still my hope, is to use a prefix for all incoming calls from callmanager, which matches the proper outgoing dialpeer.
    In your translation-profiles you can apply your translation-rules to either calling, called, or redirect called party numbers.  You can do your digit manipulation at that point to remove your prefix.
    It also sounds like you're dealing with hunting issues.  In addition to the preference commands, you can also use the huntstop and even permission commands to limit hunting and add hard incoming/outgoing limitations to your dial-peers.
    Although I may have completely misunderstood the issue in which case please ignore. 
    thanks,
    will

  • How to append calling and called number with translation rules?

    Hello,
    I have one question about digit manipulations.
    How to append calling number and called number with IOS commands?
    For example, when 123 dials 45678, translations have to be performed and the new called number to be 12345678.
    Thank you,
    I will vote this conversation.

    It is not possible with translation rules.
    However, you can do that with a TCL/IVR script.

  • Suggestion needed urgently !!!!!

    Hi Experts,
    Our system configuration is
    Usage: Central User Administration
    Component Version: Web AS 620 ABAP
    Operating System: AIX 5.2
    Database: 9.2.0.6
    DB Size: 110 GB
    We also have GRC products installed and as per OSS notes there are limitations of using CUA with GRC products. (Note 1099011 - Limitations of using the CUA with the GRC products)
    We are planning to upgrade the current CUA to NW AS 700.
    We have following types of child systems are connected with CUA
    APO 3.0A, SAP Solution Manager 4.0, CRM 5.0, SAP SCM 4.0, SCM 4.10, SCM 5.00, SAP NetWeaver 2004s, Netweaver 04, SAP ECC 6.0, SAP R/3 Enterprise, R/3 release 4.6C
    Please suggest what will be impact of CUA upgrade to NW AS 700.
    What will be features and benefits of the same?
    Please suggest urgently.

    Fixed it with the help from a very clever colleague, thanks again to him :)
    We put two partitions (in my case) PT_Transformation_Called and PT_Transformation_Calling in, also two CSS with the respective Partitions within them in the system. Afterwards removed all manipulations elsewhere, for example routepatterns or routelist, set these to "Cisco Callmanager".
    Then put in some Calling and Called Transformation Patterns, on exactly these partitions, where i do my needed Digit manipulation and setting to TON and Numbering Plan.
    Then i used these new CSSs on the endpoint itself, thats their only purpose, doing the transformations THERE.
    So my "Called Party Transformation CSS" is now "CSS_Transformation_Called", and the "Calling Party Transformation CSS" is now "CSS_Transformation_Calling", under "Call Routing Information - Outbound Calls". Thats it. I can do any manipulations now there, quite comfortable.
    Thats the lesson from my colleague i got yesterday, solved my problem, everything regarding that works now fine.
    More time left for my other problems now, for example a CUPS installation that stops after a while now everytime, with an unrecoverable error...nice... :(

  • Voice Routing: Normalization Rule vs. Route

    We're just testing some Enterprise Voice stuff within our Lync Environment but there's still one Thing not clear to me: On the specific user tab, we're able to assign a "Dial plan policy" and a "Voice policy"
    In the Dial plan policy we're able to enter the normalization rules for Digit Manipulation.
    In the Voice policy we can assign routes the call will take (e.g trunk).
    But for example if there is a normalization rule which translates +402221111 to 1111 and on the route there are just numbers allowed starting with 1, why the call will fail? In my opinion the number should be translated to 1111 right after dialing and
    this should match the proper route.
    Can someone describe how the callflow is working in Detail. Will it maybe check for the specific route before there is any Digit Manipulation process?
    Thanks in advance

    Typically you'd do it the other way around, you'd translate 1111 to +402221111 so that it's in proper E.164 format, and you'd have your routes match that. 
    What you described should work though, at a very high non-mechanical level:
    A number is typed into Lync
    The number is normalized by the users dial plan (typically to a standard such as E.165)
    The normalized number is compared against the voice policy, which will match if it finds a route that matches that's tied to the policy through a PSTN usage.
    An INVITE is sent through the trunk using the path determined. 
    What kind of errors are you getting and do you get a "pass" in the Lync Control Panel -> Voice Routing -> Test Voice Routing when you enter the information?  If not, you've got a typo or misconfiguration somewhere and giving more detail
    will help.
    Please remember, if you see a post that helped you please click "Vote As Helpful" and if it answered your question please click "Mark As Answer".
    SWC Unified Communications
    This forum post is based upon my personal experience and does not necessarily reflect the opinion or view of Microsoft, its employees, or other MVPs.

  • Callmanager handling +1 inbound calls from Webex

    I have callmanager version 9.1.2 and on-prem Webex Meeting Server (1.5.1.323A). The "callme" function works but webex tags on a +1 to any number it calls. I can strip off the +1 with a translation pattern which works for Cisco phones but am not sure how to handle numbers that need to go to PSTN. Do I need transformation pattern to strip +1 and add 9? Can this be done at the sip trunk level?

    Hi Mate,
    First of all, for internal calls CWMS should not prefix + and a country code. There's an option on the call me - something like call internal number. If a user dials a number using this option it will ring internal extension without requiring any digit manipulation. Otherwise, for PSTN calls you select the country flag for the destination you are calling and it adds + and the respective country code.
    You can simply setup a translation pattern stripping + and prefixing an access code 9 or 0 or whatever and your existing call routing will take care of rest.
    I had done a CWMS implementation few months back and it worked fine for me. Not remember exactly if I used TP or RP but I didn't had any issue.
    -Terry
    Sent from Cisco Technical Support iPhone App

  • Looking for TCL/Scripting Resource

    Hey guys,
    I'm looking to contract with someone that can write scripts (TCL/Csh/etc) on the Cisco gateways to do things like custom CDR, digit manipulation, IVR, etc. Can anyone point me to some good people?
    Thanx,
    -Tim

    Contact Berbee.com or cdw.com they have good resources. Andy (Adignan) my buddy can guide you to a good contact as well.
    Baseer.

  • How to remove (0) from the CUCM corporate directory

    Hi,
    My users are using the CUCM corporate directory to look for contacts. The CUCM is currently synchronized with Active Directory server using the "Telephone Number" to synchronized the CUCM field "Phone Number".
    My problem is about directory numbers like : +49 (0) 4034563 20-32
    Currently when users try to dial this number, the CUCM automatically removes round brackets et dash, so the number dialed is : +49040345632032
    I would like to remove (0) in place of just remove round bracket and keep 0. Is it possible to modify the CUCM behavior to obtain +4940345632032 ?
    I'm currently using CUCM 8.6

    No, this is not possible using the native corporate directory application. The options are:
    1. Use another AD attribute (e.g. ipPhone) and populate that attribute the numbers you want to present to Cisco phones
    2. Keep the digit presentation the same and see if the CUCM dial plan can be modified to deal with the extra zero (this may or may not jive with your requirements. So, this option may be useless)
    3. Build (or have someone build) a custom corporate directory that deals with the digit manipulation so you present exactly what you want to present to the IP phone
    4. Purchase an off the shelf corporate directory that provides digit manipulation functionality
    I have employed #1 and #3 (sometimes together) for my customers. For #1 it is relatively simple to export telephoneNumber attribute, run it through a filter, and update the AD user object with the ipPhone attribute. It is a little more challenging operationally because you have to ensure this is followed by admins (or automation?). For #3, I use a corporate directory toolkit I developed. If you are savvy with coding then this is do-able or you can hire someone who can do it for you.
    Since my company has their own custom directory solution, I have not shopped around too heavily for an "off the shelf" solution. So, I can't make a specific recommendation there.
    HTH.
    Regards,
    Bill
    http://ucguerrilla.com

  • Connection between FXS/DID to PBX Trunk

    Hello
    I am trying to find how to setup a trunk between FXS/DID ports on a Cisco Router to a PBX.
    Can anyone who has set this up before share some some documents / links which shows how this can be setup?
    Many Thanks

    Well you can find info about creating trunk groups with voice ports (FXO or FXS) with google definitely :)
    But here is some example of real config about that:
    trunk group FXS-PBX
     hunt-scheme round-robin - "THIS IS TRUNK GROUP CONFIG"
    voice-port 0/1/0
     trunk-group FXS-PBX
     cptone SI
     description ===FXS PBX 1===
    voice-port 0/1/1
     trunk-group FXS-PBX
     cptone SI
     description ===FXS PBX 2===
    voice-port 0/1/2
     trunk-group FXS-PBX
     cptone SI
     description ===FXS PBX 3=== - "PUT 3 VOICE FXS PORTS IN TRUNK GROUP"
    dial-peer voice 100 pots
     trunkgroup FXS-PBX
     description ===FXS PBX===
     destination-pattern XXX - "CREATE PATTERN TO SEND CALLS TO PBX"
    And that would be your start point...then configure your PBX digit manipulation in opposite direction (can't help on that because I'm in IP world definitely :) )
    HTH,
    Dragan

  • IPGW and the ASA

    I want to install a 2651XM gateway/Gatekeeper into a DMZ of my firewall so that Internal Polycom devices can register and communicate to both internal Polycom devices and external video conferencing devices. Trouble finding out how to do this? What ports need to be open and can this be done with one gateway/gatekeeper? Documentation I have read suggests two are needed, one acting as a proxy. I only have one unit.

    A Cisco Multiservice IP-IP gateway and Gatekeeper (or MCM Proxy and Gatekeeper) can co-exist on the same box, but a via-zone aware GK is required for an IP-IP Gateway. The IP-IP Gateway is certainly more flexible, particularly for digit manipulation (through the configuration of dial-peers) and integration with Cisco CallManager.
    Have all you endpoints register with a local zone on the GK, and then configure remote zones for external gatekeepers.
    Then, for interoperability with an ASA box perform a static NAT translation of the GK/IPIPGW box to a real-world address, and allow H.323 Gatekeeper RAS (1719/udp) and H.323 H.225 call setup (1720/tcp) and the application inspection on the ASA will open the required ports for the RTP streams.
    Hope this helps. Please rate useful posts!

  • Call forwarding offnet doesn't insert 1 for non-local calls

    Hello,
    When users at a remote site (long distance) call over the WAN to a central site phone that is call forwarded to a number local to the central site.  The remote site phone is sent through its voice gateway and the calls fails because the CallManager or gateway doesn't insert a 1.
    Right now a special calling search space is configured on the central site phones that send call forwarded calls to the central site gateways.
    What other ways could we accomplish the same thing.  Voice translation patterns on the gateways, etc?

    There are multiple places you can do digit manipulation, like you mentioned you can do it on voice translation patterns applied on dial peers on the gateway, if you have CUCM, you can also do digit manipulation on route patterns or also on route lists.
    Not sure if this is what you were looking for.

  • Suggestion urgently needed, DigitManipulation for forwarded calls

    Hello community,
    i have an urgent problem with the display of the original calling number when the call is forwarded.
    I have CUCM 10.5 and E1/MGCP gateways, in germany.
    Heres my output of "debug isdn q931", think it gets quite clear what i want to do or what my problem is then.
    Dec  4 16:58:06.260: ISDN Se0/0/0:15 Q931: RX <- SETUP pd = 8  callref = 0x0293
                Bearer Capability i = 0x9090A3
                           Standard = CCITT
                           Transfer Capability = 3.1kHz Audio
                           Transfer Mode = Circuit
                           Transfer Rate = 64 kbit/s
                Channel ID i = 0xA98381
                           Exclusive, Channel 1
                Progress Ind i = 0x8A81 - Call not end-to-end ISDN, may have in-band info
                Calling Party Number i = 0x1180, '86139XXXXXXXXX'
                           Plan:ISDN, Type:International
                Calling Party Number i = 0x1183, '86139XXXXXXXXX'
                           Plan:ISDN, Type:International
                Called Party Number i = 0xC1, '998XXX123''
                           Plan:ISDN, Type:Subscriber(local)
    Dec  4 16:58:06.284: ISDN Se0/0/0:15 Q931: TX -> CALL_PROC pd = 8  callref = 0x8293
                Channel ID i = 0xA98381
                           Exclusive, Channel 1
    Dec  4 16:58:06.288: ISDN Se0/0/0:15 Q931: TX -> SETUP pd = 8  callref = 0x0005
                Sending Complete
                Bearer Capability i = 0x8090A3
                           Standard = CCITT
                           Transfer Capability = Speech
                           Transfer Mode = Circuit
                           Transfer Rate = 64 kbit/s
                Channel ID i = 0xA98382
                           Exclusive, Channel 2
                Progress Ind i = 0x8A81 - Call not end-to-end ISDN, may have in-band info
                Calling Party Number i = 0x2183, '0086139XXXXXXXXX'
                           Plan:ISDN, Type:National
                Called Party Number i = 0xA1, '170XXXXXX'
                           Plan:ISDN, Type:National
    Dec  4 16:58:06.332: ISDN Se0/0/0:15 Q931: RX <- CALL_PROC pd = 8  callref = 0x8005
                Channel ID i = 0xA98382
                           Exclusive, Channel 2
    Dec  4 16:58:07.896: ISDN Se0/0/0:15 Q931: RX <- ALERTING pd = 8  callref = 0x8005
                Progress Ind i = 0x8288 - In-band info or appropriate now available
                Progress Ind i = 0x8282 - Destination address is non-ISDN
    Dec  4 16:58:07.900: ISDN Se0/0/0:15 Q931: TX -> ALERTING pd = 8  callref = 0x8293
                Progress Ind i = 0x8288 - In-band info or appropriate now available
    What u see on the debug is a call originating from a chinese number (+86, which is also recognized as it should as international call) to the 998XXX123, and the DN 123 has forwarded the call to a german cellphone, the +49170XXXXXX. As u can see in the debug the type of the forwarded number switches to NATIONAL, which gets the +49 (instead keeping the +86 only!) prepended,which means on the target display i will see the +490086139XXXXXXXXX.
    What is total crap, because in case the called person misses that call no way to directly call back because of the shitty number display. On the gateway endpoint forwarding the originating caller number is activated. Plan and Type will be set on the routepattern, not on the endpoint. I played around a bit with the digits, but maybe i just don´t get it...where can i set that the original INTERNATIONAL calling number gets forwarded, with NO +49 prepended?
    BTW we have here clip-no-screening. And i don´t think its a law thing here, regarding not being able to display on outbound direction a countrycode from abroad. Because this worked on the 14 years old former PBX...and i don´t get it working with the brand new callmanager.
    Any suggestions very welcome, where and how i could do this digitmanipulation, many thanks in advance!

    Fixed it with the help from a very clever colleague, thanks again to him :)
    We put two partitions (in my case) PT_Transformation_Called and PT_Transformation_Calling in, also two CSS with the respective Partitions within them in the system. Afterwards removed all manipulations elsewhere, for example routepatterns or routelist, set these to "Cisco Callmanager".
    Then put in some Calling and Called Transformation Patterns, on exactly these partitions, where i do my needed Digit manipulation and setting to TON and Numbering Plan.
    Then i used these new CSSs on the endpoint itself, thats their only purpose, doing the transformations THERE.
    So my "Called Party Transformation CSS" is now "CSS_Transformation_Called", and the "Calling Party Transformation CSS" is now "CSS_Transformation_Calling", under "Call Routing Information - Outbound Calls". Thats it. I can do any manipulations now there, quite comfortable.
    Thats the lesson from my colleague i got yesterday, solved my problem, everything regarding that works now fine.
    More time left for my other problems now, for example a CUPS installation that stops after a while now everytime, with an unrecoverable error...nice... :(

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