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Office Telephony on PureConnect

  • 1.  Office Telephony on PureConnect

    Top 25 Contributor
    Posted 10-12-2020 22:01
    Edited by Subhash Srivastava 10-12-2020 23:55
    Dear Community Members,

    I am looking for some reference material/use cases for implementing Enterprise telephony on PureConnect.
    already browsed through many documents available online/Genesys Portal but couldn't find anything concrete.
    Please help if you have some inputs to share.

    Regards,
    #Telephony

    ------------------------------
    Subhash Srivastava
    ------------------------------


  • 2.  RE: Office Telephony on PureConnect

    Posted 10-13-2020 04:14
    Dear Subhash,

    What type of implementation are you planning to implement with PureConnect platform? Can you please provide more details?

    As per my experience you can use PureConnect with Cisco UCM and you can register Cisco IP Phones to PureConnect. Then you will have a chance to run Call Center and Business Users at the same time.

    Thanks,
    Regards
    Cenk

    ------------------------------
    Cenk Gunduz
    ------------------------------



  • 3.  RE: Office Telephony on PureConnect

    Top 25 Contributor
    Posted 10-13-2020 05:03

    ​Dear Cenk,

    Thanks for your response.

    CUCM and CIC integration is already in places using SIP trunk in between.
    Actually, I am looking for replacing CUCM with PureConnect and planning to migrate all users on PureConnect.





    ------------------------------
    Subhash Srivastava
    Shinsei Bank, Limited
    ------------------------------



  • 4.  RE: Office Telephony on PureConnect

    GCAP Member
    Posted 10-13-2020 12:08
    Subash,

    As you have noticed there isn't much documentation targeted towards Enterprise specific implementations.  You decide how you want to implement the user experience.

    Specific things to look for in the documentation are:

    Operator Add-On (https://help.genesys.com/pureconnect/mergedProjects/wh_qr/desktop/pdfs/client_desktop_operator_console_qr.pdf) , Call Coverage, and Monitored Appearances.  These are things that help facilitate the work that an executive assistant or receptionist might perform.
    Mini-Mode, I found that some business users prefer that view of the desktop.
    Create and curate custom contact lists and speed dial lists for use by different groups.
    While none of these have loads of documentation, you can read up and use them to decide what level of training to provide.


    Lessons that I learned:
    Make sure that the default user has the fewest security  right possible (none is best).  This allows you to assign entitlements with greater granularity (especially status messages allowed).
    Keep the instructions as simple as possible.  Most enterprise users don't use the application often enough to become proficient at using the many features in interaction desktop.
    Folks in the E/C-Suite just want a stand alone phone on the desk rather that having to login to the client to receive calls.

    ------------------------------
    Tim Cannon
    ------------------------------



  • 5.  RE: Office Telephony on PureConnect

    Top 25 Contributor
    Posted 10-14-2020 00:44

    ​Hi Tim,

    Thanks for your response.
    I will check the information as you suggested.

    Basically I don't have any concerns for the users who want to use it with Interaction Desktop login.

    I need some idea about configuring those phones as stand-alone phones & assigning DID's to them.

    one way I found it by using DID/DNIS configuration option in IA and then mapping it with station extension defined under IA -Stations.

    However, I am not sure, if this option would be able to display the DID on phone itself.
    May be I need to add description in SIP file for the phone to display the desired DID number.

    I will try various options and if you have anything more to suggest, please update.



    ------------------------------
    Regards,
    Subhash Srivastava
    ------------------------------



  • 6.  RE: Office Telephony on PureConnect

    Posted 10-14-2020 03:51
    Hi Subhash,

    You are on the right track which you can use DID/DNIS options to forward calls to the specific stations with specific DID numbers. And of course the caller number is displayed on the phone screen.

    I am also agree with Tim, especially majority of the enterprise users prefer to use the phone itself on their desk instead of using Interaction Desktop application on their PC. Actually there are some configurations for the clients to forward calls to the Desk Phones or mobile phones (Follow-me Option etc..) either PC/Laptop is or application is off.

    You should give it try some options and you will probably have some questions and come back here to discuss with us for the best practical way.

    Thanks,
    Regards

    ------------------------------
    Cenk Gunduz
    ------------------------------



  • 7.  RE: Office Telephony on PureConnect

    Top 25 Contributor
    Posted 11-23-2020 21:54

    Hi Cenk,

    As discussed, I had created DID 's and assigned them to Stations/workgroups/Users for various feature tests.

    I was able to get the inbound calls to DID on configured station, WG's & users, even outbound call work fine & shows proper ANI.
    However, I got stuck with one option & don't know how to address it, need your help.

    I have configured DID numbers which are mapped with Stations (configured as cisco phones registered as SIP phone with CIC).
    Inbound/Outbound works fine, but when inbound call comes to the station phone, I don't see actual ANI of the caller, rather the ANI defined on SIP tie line is displayed.

    Could you please guide how I can get the caller ANI displayed on phones.



    ------------------------------
    Regards,
    Subhash Srivastava
    ------------------------------



  • 8.  RE: Office Telephony on PureConnect

    Posted 11-24-2020 03:03
    Hi Subhash,

    HAve you tried to check ANI of the caller on ICBM screen? IS it displayed there correctly or still wrong?

    Actually probably you will see the correct ANI on ICBM screen otherwise if the wrong number passed by your GW or Telco your DID/DNIS mapping should also not work.

    As i see you are using CUCM and this should be configured on CUCM i believe rather than passing original ANI information via your SIP Line or not. Because phones are not registered on CIC they are still registered on CUCM as a station and this should be double checked on CUCM.

    Thanks,
    Regards

    ------------------------------
    Cenk Gunduz
    ------------------------------



  • 9.  RE: Office Telephony on PureConnect

    Top 25 Contributor
    Posted 11-24-2020 05:08

    Hi Cenk,

    Thanks for your revert.
    Yes, on ICBM screen I can see correct ANI is being passed from GWY/Telco to CIC.
    when I am login to Interaction Desktop using CIC user with my mobile number(using remote​ number option), correct ANI is displayed when Inbound call to mapped DID reaches to my user.

    On the other hand if I map same DID with one of the station extension defined in CIC (Station Extn is SIP phone registered to CIC), Inbound call get routed to correct station but ANI is not displayed.

    Just to add, though we are using Cisco 7960 phones, these phones  are not registered to CUCM, they are configured on CIC as SIP phone under unmanaged stations.

    As these phones used SIP Tie-Line defined on IA for SIP station connectivity, ANI of SIP-Tie line is displayed instead of actual caller ANI.

    There is no role for CUCM to play in between.




    ------------------------------
    Regards,
    Subhash Srivastava
    ------------------------------



  • 10.  RE: Office Telephony on PureConnect

    Posted 11-24-2020 07:13
    Hi Subhash,

    The method is different while using Remote Number option. Because once you use Remote Number as a station, CIC initiates an Outbound Call to your Cell phone then connect you to incoming call. If you see the correct ANI on your Cell Phone that means Outbound ANI is correctly configured and passed by CIC. This is done on the Line Settings under SIP line identity (out) options. (Line Value 1, Line Value 2...)

    For the standard SIP stations as you configured your Cisco IP Phones, this is an inbound call and also can be checked and configured on the Line Settings under SIP line identity (in) options.

    Can you please check the following parameter in there;

    - Use only numeric portion (this should be checked to dispaly numeric portion of the CallerID)

    Also you can play with the following setting;

    - Calling Address (this should be set to "Use 'P-Asserted-Identity' header only:" or also you can try "Use 'P-Asserted-Identity' header then 'From' header"' (default):"

    Note: All Inbound Identity settings affect the values of the Eic_RemoteAddress and Eic_RemoteId call attributes.

    I hope this will help you to find a fix your problem.

    Thanks,
    Regards

    ------------------------------
    Cenk Gunduz
    ------------------------------



  • 11.  RE: Office Telephony on PureConnect

    Top 25 Contributor
    Posted 12-01-2020 02:07
      |   view attached

    Hi Cenk, 

    Thanks for your continuous support.

    I have exhausted all the options as suggested, however, still I can't see original ANI of SIP station.

    IC still shows me the <Station-UDP> Line Value 1, which is defined as  "8888"<sip:8888@co.jp> on line configuration.

    Is there anything I am missing or do I need to do any additional configuration. 

    Please suggest.



    ------------------------------
    Regards,
    Subhash Srivastava
    ------------------------------



  • 12.  RE: Office Telephony on PureConnect

    Posted 12-02-2020 06:41
    Hi Subhash,

    May i ask why you prefer using Stations-UDP line for incoming calls?

    These following 3 lines especially used for registration purpoese not for calls;

    - Stations-UDP
    - Stations-TCP
    - Stations-TLS

    You can create a new line and go to access section of the line settings and configure your ip address and exceptions and leave Line Values as default and test it. 

    Thanks,
    Regards



    ------------------------------
    Cenk Gunduz
    ------------------------------



  • 13.  RE: Office Telephony on PureConnect

    Top 25 Contributor
    Posted 12-02-2020 21:18

    Hi Cenk, 


    Yes, these 3 lines are for registration purpose and our stations (SIP Phones) are registered using - Station-UDP line.
    Also when any call comes to these stations (call coming directly to these stations or incoming call assigned to agents, who are using these stations for login as workstation), IC uses Station-UDP line to connect the station.


    In my case, I am configuring DID number for each SIP Phone(station), as follows:
    under DID/DNIS menu, configuring a DID and mapping it to station extension.

    lets say DID is : 06-7731-0098 and Station extension is : 3002 

    now when I dial this DID externally, call reaches to IC platform and IC transfers it to extension mapped to this DID, i.e. 3002.
    To do so IC initiates another call using Station-UDP line to station: 3002 and patch it with Incoming call leg.

    As SIP Station is dialed via Station-UDP line, IC gives line level parameters as ANI for the call.

    --------

    Are you suggesting that, I keep Station-UDP line as it is for SIP station registration purpose and define another line for dialing out to these stations by updating dial plan ?



    ------------------------------
    Regards,
    Subhash Srivastava
    ------------------------------



  • 14.  RE: Office Telephony on PureConnect

    Posted 12-03-2020 01:03
    Hi Subhash,

    Normally if you are using DID/DNIS mapping CIC will not initiate a 2nd call to patch incoming call to the station. IC Platform directly patch the incoming call to that station but yes there are 2 legs as you say, one is incoming leg and the another is station leg. And you can only see the incoming leg in the CDR logs (In the InteractionSummary Table you can see only one row for each interactionid of each incoming call)

    Once you call that DID externally, it is going to be translated to DID (For example last or first 4 digit etc..) by the Telco or GW and you can catch it via DID/DNIS mapping in the Dial Plan and transfer it to your extension.

    And yes these 3 lines are used only for registration purpose. (This does not mean that these lines can not be used for calls but not recommended)

    You can create another line (for example: SIP_Line_Test) and configure its access option (Allow/Deny ip address) to receive all incoming calls from that line and see what happens? I do not mean that use the new line for dial-out purpose by configuring Dial Plan.

    You know for the incoming calls you do not need to play with Dial Plan, you only need to configure DID/DNIS mapping if needed. Therefore for the incoming calls you do not need to touch your dial plan you only need to configure the access options (Allow/Deny ip addresses) of the new line.

    Thanks,
    Regards

    ------------------------------
    Cenk Gunduz
    ------------------------------



  • 15.  RE: Office Telephony on PureConnect

    Top 25 Contributor
    Posted 12-03-2020 02:13

    Hi Cenk, 

    Need some more clarity on this :

    You can create another line (for example: SIP_Line_Test) and configure its access option (Allow/Deny ip address) to receive all incoming calls from that line and see what happens? I do not mean that use the new line for dial-out purpose by configuring Dial Plan.

    are you saying that I configure this line to receive all incoming calls on it ? 

    if so, my question still remains open , First leg of the call would connect using this line(which is external incoming call)

    however, 2nd leg which is for connecting to station, would still use same station-UDP Line and display same old values of ANI.

    I am sharing the Tsserver log snippet for your reference :

    Leg 1 :  INCOMING CALL fromm GWY to IC on DID

    CSIPCallInfo (1104343199):
    To=[sip:77310092@10.143.196.22]/[]
    From=[sip:08046539882@10.145.98.202]/[]

    Leg 2 :  IC connects to SIP Station extension mapped with DID

    CSIPCallInfo (1104343200):
    To=[sip:3002@10.0.59.105:5060]/[]
    From=[sip:8888@co.jp:8060]/[Interaction Center]

    in Leg 2 it passes identity defined on line used to connect to station.

    how i can change that.




    ------------------------------
    Regards,
    Subhash Srivastava
    ------------------------------



  • 16.  RE: Office Telephony on PureConnect

    Posted 12-04-2020 03:00
    Hi Subhash,

    Then please clear the Line Value 1 under calling address of Stations-UDP (IdentityOut) then please try again.

    It may help you.

    Thanks,
    Regards

    ------------------------------
    Cenk Gunduz
    ------------------------------



  • 17.  RE: Office Telephony on PureConnect

    Top 25 Contributor
    Posted 12-04-2020 03:46
    Hi Cenk,

    As per Calling address value for Line 1 can't be left blank.
    it says that "Line Value 1 must contain a non-empty address"

    ------------------------------
    Regards,
    Subhash Srivastava
    ------------------------------



  • 18.  RE: Office Telephony on PureConnect

    Posted 12-04-2020 04:51
    Hi Subhash,

    Can you please try the following steps?

    1.       In the Calling Address section, next to the Line Value 1 field or the Line Value 2 field, click ...   The Configure Line Value dialog box appears.
    2.       To display non-specific information for the outbound line identification, select the Use Anonymous values check box. This option displays: Address = "sip:anonymous@anonymous.invalid," Name = "Anonymous."
    Thanks,
    Regards

    ------------------------------
    Cenk Gunduz
    ------------------------------



  • 19.  RE: Office Telephony on PureConnect

    Posted 12-04-2020 06:31
    Hi Sunhash,

    I think we miss something but i don't know what is that. MAybe we can arrange a remote session and i may help you on this later, OK?

    Thanks,
    Regards

    ------------------------------
    Cenk Gunduz
    ------------------------------



  • 20.  RE: Office Telephony on PureConnect

    Top 25 Contributor
    Posted 12-06-2020 20:18

    Hi Cenk, 

    Thankyou so much  for your continued support.

    Lets meet on remote session tomorrow. if you are OK, please share your suitable timeslot.

    FYI, I am in JST time zone(GMT+9:00)  - can connect anytime between (11:00 JST to 18:00 JST).



    ------------------------------
    Regards,
    Subhash Srivastava
    ------------------------------



  • 21.  RE: Office Telephony on PureConnect

    Top 25 Contributor
    Posted 12-04-2020 06:14
    Hi Cenk,

    Tried this one as well & this option shows "anonymous..." as ANI.


    ------------------------------
    Regards,
    Subhash Srivastava
    ------------------------------



  • 22.  RE: Office Telephony on PureConnect

    Top 25 Contributor
    Posted 12-13-2020 21:19

    Hi Cenk, 

    Kindly confirm when we can connect over remote session to troubleshoot it further.



    ------------------------------
    Regards,
    Subhash Srivastava
    ------------------------------



  • 23.  RE: Office Telephony on PureConnect

    Posted 12-17-2020 02:58

    Dear Subhash,

    I'm sorry for late reply as we are in a very tight and busy schedule in Dec.

    I will try to be online on Monday according to my Turkey Local Time afternoon. I don't know how is the time difference but our time zone is GMT+3.

    Thanks,

    Regards



    ------------------------------
    Cenk Gunduz
    ------------------------------



  • 24.  RE: Office Telephony on PureConnect

    Top 25 Contributor
    Posted 12-17-2020 03:55

    Thanks Cenk.

    I am 6 hours ahead of your time.

    Just ping me when you are available, i will connect.



    ------------------------------
    Regards,
    Subhash Srivastava
    ------------------------------



  • 25.  RE: Office Telephony on PureConnect

    Top 25 Contributor
    Posted 10-15-2020 02:03
    Edited by Subhash Srivastava 10-15-2020 02:06
    ​Hi Tim & Cenk,

    Thanks a lot for your inputs and confirming the I am on right track to achieve the required functionality.
    As suggested, I will go ahead and try out these options in our development environment & get back with updates.

    ------------------------------
    Regards,
    Subhash Srivastava
    ------------------------------