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Integration to Cisco CUCM SIP Trunks

  • 1.  Integration to Cisco CUCM SIP Trunks

    Posted 07-07-2021 11:29
    Looking to see if anyone has successfully integrated their Cisco CUCM to Genesys Cloud.  I can't find any integration notes online and wanting to make sure this is possible.
    #Ask Me Anything (AMA)
    #Integrations
    #SIP/VolP
    #Telephony

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    Martin Bunting
    i3Vision Technologies Inc.
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  • 2.  RE: Integration to Cisco CUCM SIP Trunks

    Posted 07-07-2021 15:34
    Yes, many times.  Essentially, you need a BYOC SIP trunk to the CUCM and route calls back and forth with a dial plans on each side.  You can pass UUIData back and forth for screen pops and the like.

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    Robert Wakefield-Carl
    Avtex Solutions, LLC
    Contact Center Innovation Architect
    robertwc@avtex.com
    https://www.Avtex.com
    https://RobertWC.Blogspot.com
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  • 3.  RE: Integration to Cisco CUCM SIP Trunks

    Posted 07-07-2021 23:56
    The other thing to consider here is how your calls will be routing to/from the CUCM publisher and subscriber nodes.  From an inbound CUCM=>Genesys perspective you can get away with having a single BYOC SIP Trunk configured and in the SIP trunk configuration in CUCM configure the destination(s) to be the IPs for each of your Edges and on the trunk configuration in Genesys Cloud add the IPs for each node to the SIP Server/Proxies and SIP Access Control lists.

    You might want to also consider creating a separate trunk pointing to each CUCM node so you can test specific paths.


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    Vaun Mccarthy
    NTT New Zealand Limited
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  • 4.  RE: Integration to Cisco CUCM SIP Trunks

    Posted 07-08-2021 00:07
    Thanks, Vaun and Robert.  To add this would be a new installation of Genesys Cloud and as such we would be using BYOC-C Edges.   Genesys is not currently selling local edges unless there is a strong compelling reason.

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    Martin Bunting
    i3Vision Technologies Inc.
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  • 5.  RE: Integration to Cisco CUCM SIP Trunks

    Posted 07-08-2021 01:34
    Yes, that is true, so the alternative to local Edges is to use a local SBC that talks to all your nodes and then use that to connect into Genesys Cloud through one or two SIP trunks.

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    Robert Wakefield-Carl
    Avtex Solutions, LLC
    Contact Center Innovation Architect
    robertwc@avtex.com
    https://www.Avtex.com
    https://RobertWC.Blogspot.com
    ------------------------------



  • 6.  RE: Integration to Cisco CUCM SIP Trunks

    Posted 12-14-2021 01:49
    Is there any guide or documentation that where to configure in CUCM in order to get the integration working between GC and Cisco CUCM?

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    JIMMY KWAN
    ITApps Sdn Bhd
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  • 7.  RE: Integration to Cisco CUCM SIP Trunks

    Posted 12-14-2021 01:55
    HI Jimmy

    As per the above comments, it's relatively straight forward and isn't really any different to setting up any other type of SIP trunk really.  Once the trunk(s) is/are configured in CUCM and Genesys Cloud, providing firewalls etc allow it and you've got numbering plans/route patterns setup then you should be away.

    As per above though it does depend on what your Edge deployment model is though.

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    Vaun McCarthy
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  • 8.  RE: Integration to Cisco CUCM SIP Trunks

    Posted 12-14-2021 02:15
    Edited by Matt Lawson 17 days ago
    Hi Vaun,

    I am working with the Cisco CUCM engineer now. He has created the SIP trunk and the SIP trunk status is UP.
    below is the SIP Trunk config.
    Below is the forwarding

     

    He has configured a DID that point to this trunk. unfortunately when i call the DID, number not in service message announced to me. Any idea?

     

     

    With Regards, 

     

    Jimmy Kwan

    Senior Consultant

     






  • 9.  RE: Integration to Cisco CUCM SIP Trunks

    GENESYS
    Posted 12-15-2021 11:49
    Jimmy,

    That configuration doesn't look correct to me.

    You shouldn't be targeting the lb01.byoc.ap-northeast-1.mypurecloud.jp address directly and you should instead use the fully qualified domain name (FQDN) that you have setup in the SIP Trunk on the Genesys side.  That FQDN of your Genesys SIP Trunk should probably look something like <you_company_name>.byoc.mypurecloud.jp.  Sending to that endpoint will then load balance across the lb0x.* instances.

    On the Genesys side you also need to create a DID and associate that DID to an Architect callflow or an agent.

    If you do all of that, then on the CUCM side your SIP INVITEs will go to <The_DID>@<FQDN_of_Genesys_SIP_Trunk>.  If you link that DID to an Architect call flow then you should hear the callflow answer the call.

    Here is an unofficial, quick and dirty guide that my team created a fe years back (old screen shots) for setting up a BYOC SIP Trunk to Twilio.  Maybe you can use it to understand all the parts necessary to setup a trunk to CUCM: https://contentmanagement.mypurecloud.com/s/#/1/4h6d5abwyndkrbbqjnlsglhx7q

    Good luck,

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    Jim Crespino
    Senior Director, Developer Evangelism
    Genesys
    https://developer.genesys.com
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  • 10.  RE: Integration to Cisco CUCM SIP Trunks

    Posted 12-15-2021 23:05
    Jim,

    Thanks for the guide.

    Rather than pointing to a lb0x, we have changed it to point to <you_company_name>.byoc.mypurecloud.jp.

    As a start, for inbound call, we manager to get the call established without voice. We found out that fo Cisco, the SIP and RTP is processed on different host/IP. The Cisco vendor is working on the normalization script to manipulate the SIP message so it can point to the right destination. At the same time, I have get the firewall vendor to make necessary configuration to have the public IP to achieve single public IP NATing to multiple host.

    After the firewall vendor configure the firewall, I have few test call and reviewed the PCAP trace. Since the call can be established, I have shift my focus on the SDP packet. The IP on SDP is point to CISCO internal SIP server which I think it should be the public IP. I am getting the Cisco vendor to make necessary changes to the normalization script, hopefully my guess is correct.

    Is there anyone here tried using Cisco CUCM to route call to GC (Cloud Edge) and successfully make it work?

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    JIMMY KWAN
    ITApps Sdn Bhd
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  • 11.  RE: Integration to Cisco CUCM SIP Trunks

    Posted 12-16-2021 12:29
    Hi Jimmy,

    You may ask your Cisco vendor to configure a Trusted Relay Point (TRP) for the SIP trunk and limit the RTP ports to a specific range, for example 10000-11999; then ask firewall vendor to configure port forwarding for this port range to the TRP IP. The Cisco vendor also needs to have LUA script to change connection IP in SDP to the public IP configured port forwarding. This is for RTP part.

    About SIP signaling part, I believe your Cisco vendor already configured LUA script for sip message manipulation, just make sure the host part in Via and Contact should be the public IP or FQDN.

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    Wen Gu
    Expert
    SITA
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  • 12.  RE: Integration to Cisco CUCM SIP Trunks

    Posted 01-13-2022 06:06
    Edited by Matt Lawson 17 days ago
    Hi All,

    I manage to make inbound and outbound call with both end voice but outbound call will automatically disconnect in 18 seconds based on GC.

    below is the invite message from GC



    and below is the 200 ok from cisco

    i notice there are multiple 200 ok being send out and there is no ACK from GC for the 200 OK.
    I noticed there is something not right in the SDP where by GC use OPUS and Cisco use PCMU. is this because of these? I have tried to configure G711 ulaw in GC but not able to do so. Only OPUS is allowed.

    ***EDIT***
    I just found an article that have sharing on missing ACK after 200 OK.
    Troubleshooting missing ACK in SIP
    Drops of wisdom, knowledge and news from OpenSIPS remove preview
    Troubleshooting missing ACK in SIP
    We all experienced calls getting self disconnected after 5-10 seconds - usually disconnected by the callee side via a BYE request - but a BYE which was not triggered by the party behind the phone, but by the SIP stack/layer itself. This is one of the most common issues we get in SIP and one...
    View this on Drops of wisdom, knowledge and news from OpenSIPS >

    Based on article, the 200 OK Contact should be in Public IP rather then private IP (customer internal IP).

    should it be the public IP?

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    JIMMY KWAN
    ITApps Sdn Bhd
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  • 13.  RE: Integration to Cisco CUCM SIP Trunks

    Posted 01-13-2022 21:11
    Hi All,

    Some gd news here. Resolved the auto disconnect issue by replacing the Contact and P-Preferred-Identity to public IP address.

    Now, encountered a new problem, when I trigger DTMF from GC, Cisco end cannot detect. I am routing the GC to an IVR to test on the DTMF input.

    Any Idea?

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    JIMMY KWAN
    ITApps Sdn Bhd
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  • 14.  RE: Integration to Cisco CUCM SIP Trunks

    Posted 01-14-2022 04:12
    Hi Jimmy,

    From the SDP bodies of the SIP INVITE and 200OK messages you shared initially, it looks like that there is a mismacth in the DTMF codec used (media type "telephone-event"):
    GC is offering DTMF with clock rates 8000 and 48000 Hz with payload types 101 and 102 repectively.
    Cisco is accepting DTMF payload type 102, but declaring it to have clock rate 8000 Hz.

    I assume GC is encoding with 48Khz and Cisco cannot decode it correctly.
    This may be a bug on the Cisco side, as it should select the payload type 101 with 8 Khz offered by GC instead of the 102.

    But not sure why GC is offering 2 different payload types for DTMF in your case.
    Normally it is possible to configure only 1 DTMF payload type in the "media" section of the external trunk configuration.

    Regards,
    Marcelo

    Regards,
    Marcelo





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    Marcelo Heil França
    InfinIT.cx GmbH
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