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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.
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.
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.
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.
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:firstname.lastname@example.org> on line configuration.Is there anything I am missing or do I need to do any additional configuration.
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 ?
------------------------------Subhash SrivastavaShinsei Bank, LimitedOriginal Message:Sent: 10-13-2020 04:13From: Cenk GunduzSubject: Office Telephony on PureConnectDear 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,RegardsCenk------------------------------Cenk GunduzOriginal Message:Sent: 10-12-2020 22:01From: Subhash SrivastavaSubject: Office Telephony on PureConnectDear 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------------------------------
------------------------------Regards,Subhash SrivastavaOriginal Message:Sent: 12-02-2020 06:41From: Cenk GunduzSubject: Office Telephony on PureConnectHi 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-TLSYou 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 GunduzOriginal Message:Sent: 12-01-2020 02:06From: Subhash SrivastavaSubject: Office Telephony on PureConnect
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 DIDCSIPCallInfo (1104343199):To=[sip:email@example.com]/From=[sip:firstname.lastname@example.org]/Leg 2 : IC connects to SIP Station extension mapped with DIDCSIPCallInfo (1104343200):To=[sip:email@example.com:5060]/From=[sip:firstname.lastname@example.org:8060]/[Interaction Center]in Leg 2 it passes identity defined on line used to connect to station.
how i can change that.
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).
Kindly confirm when we can connect over remote session to troubleshoot it further.
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.
I am 6 hours ahead of your time.
Just ping me when you are available, i will connect.
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