IP Office not sending PAI over SIP trunk to Cisco UCM

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  • flemig
    Aspiring Member
    • Sep 2013
    • 1

    IP Office not sending PAI over SIP trunk to Cisco UCM

    Hey guys. I have a SIP trunk between a Cisco UCM cluster and an Avaya IP Office 500 system.

    Everything works great except when calling from the Cisco phones to the Avaya phones, the called party name (destination name) does not show up on the Cisco phone. When calling from Avaya to the Cisco phones, the Avaya phone shows the name of the person you are calling. This is done via the P-Asserted-Identity header which is being passed from the Cisco UCM to the Avaya system and being displayed on the Avaya phones. (As far as I am aware, at least).

    I cannot figure out how to get the Avaya system to send the PAI header over the SIP trunk for inbound calls. Can anyone please help me out. Below are two snippets of SIP traces from either system for inbound calls.

    Calling from Avaya to Cisco UCM (Inbound to Cisco, Cisco replies with PAI in 180):
    Code:
    SIP/2.0 180 Ringing
    Via: SIP/2.0/UDP 10.2.10.189:56082;rport;branch=z9hG4bKPj3SuuV7rduceI20EeEbCRnkr4LXpd5mjI
    From: "260" <sip:[email protected]>;tag=3Ovxj7AHdRERMRJPLk98i1yxRskDhU4m
    To: <sip:[email protected]>;tag=fd298c1515c002a3
    Call-ID: Odi1wNclBwyS75WWx9tT4FRjC8pnp4zl
    CSeq: 29327 INVITE
    Contact: "821558" <sip:[email protected]:5060;transport=udp>
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, SUBSCRIBE, REGISTER, PUBLISH, UPDATE
    P-Asserted-Identity: "UserX" <sip:[email protected]:5060>
    Supported: timer
    Content-Length: 0
    (That P-Asserted-Identity field, "UserX" is what gets displayed on the Avaya phone when calling out to Cisco).

    Calling from Cisco UCM to Avaya (Inbound to Avaya, no PAI info):
    Code:
    SIP/2.0 180 Ringing
    Via: SIP/2.0/TCP 10.2.10.3:5060;branch=z9hG4bK458d1604ff9c0
    From: "UserX" <sip:[email protected]>;tag=676646~81ca2ef4-b149-42e4-8577-2d9582cea9ed-31036730
    To: <sip:[email protected]>;tag=97d82c787a1a102e
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Contact: "UserY" <sip:[email protected]:5060;transport=tcp>
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
    Supported: timer
    Content-Length: 0
    (No PAI field information. No Called Name party display on the Cisco phones).

    Please help! Thank you..
  • rmcdou
    Hot Shot
    • Nov 2013
    • 11

    #2
    Greetings to All,

    I'm having the same issue. I can establish calls between an Avaya IP Office and a Cisco CME, the only problem being is that when the Cisco side call they are getting a display of QSIG-GATEWAY on their phones.


    I'm being told that I need to have Send Caller ID set to P Asserted ID which I have but the problem still continues.

    Can some please shed some light on the subject.


    Thanks in advance

    Comment

    • rmcdou
      Hot Shot
      • Nov 2013
      • 11

      #3
      Good afternoon Flemig,

      What settings do you have your "Call Routing Method" & "send Caller ID" set to?

      Thanks

      Comment

      • markgallagher
        Legend
        .
        • May 2010
        • 613

        #4
        In the SIP URI you can define what information the IP Office should use for the PAI - including not using PAI (the default).

        Comment

        • rmcdou
          Hot Shot
          • Nov 2013
          • 11

          #5
          Thanks for your reply.

          Looking at the SIP URI under the SIP line I see:
          Local URI *
          Contact*
          Display Name*
          PAI Use Internal Data

          Please advise.

          Regards

          Comment

          • pitcherj
            Member
            • Jan 2011
            • 4

            #6
            If you have PAI set to "Use Internal Data", I believe it will use what is set in the user SIP field. If there is nothing set there then ..... Try changing it from "Use Internal Data".

            Comment

            • rmcdou
              Hot Shot
              • Nov 2013
              • 11

              #7
              Thanks for your input.

              Comment

              • markgallagher
                Legend
                .
                • May 2010
                • 613

                #8
                Is that "Thanks, I'll check" or "Thanks, that fixed it"?

                Comment

                • rmcdou
                  Hot Shot
                  • Nov 2013
                  • 11

                  #9
                  Sorry about that. That is thanks I'll check. Well, I've checked and and changed but it's a no go. We are working to see if can make it work by using Transparent Tunneling of QSIG on the CUCME, added a couple of lines to the dial peer configuration.

                  Waiting on sytem reboot on our side since we can't bring down the IPO during normal business hours.

                  Regards,...

                  Comment

                  • ali127
                    Aspiring Member
                    • Nov 2013
                    • 1

                    #10
                    Avaya ip office 500 and Cisco CME

                    Hi Flemig, could you send send the document how to configure H.323 trunk between Cisco CME and Avaya ip office 500. We need call between them in local extension.

                    Comment

                    • rmcdou
                      Hot Shot
                      • Nov 2013
                      • 11

                      #11
                      Greetings,

                      I have not found a document for the CME but I used this one.http://www.microlana.ru/sites/defaul...aya/ipo-cm.pdf

                      Regards,

                      Comment

                      • sconnell72376
                        Member
                        • Nov 2013
                        • 7

                        #12
                        Originally posted by rmcdou View Post
                        Sorry about that. That is thanks I'll check. Well, I've checked and and changed but it's a no go. We are working to see if can make it work by using Transparent Tunneling of QSIG on the CUCME, added a couple of lines to the dial peer configuration.

                        Waiting on sytem reboot on our side since we can't bring down the IPO during normal business hours.

                        Regards,...
                        I'm running into an issue where on the Cisco side they get an asterisk on calls from the IPO, but caller ID receives fine on the IPO from the Cisco. I just set PAI to Use Internal Data, but after reading this not optimistic that will work. I had * in there before so it made sense if that's what the Cisco received. I'd be curious as to how you made out with this and what your solution was.

                        Comment

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