6 Replies Latest reply on Dec 5, 2013 10:16 AM by matt

    Call on hold (SIP Trunk)

    onlysim New Member

      Hi Guys,

       

      I am facing some issue on NetVanta 71000 IP-PBX , let me describer my setup and scenario below

       

      The Adtran IP Phone 712 is connected with PBX 7100

      The PBX is configured with a trunk account.

       

       

      Scenarios Call on Hold.

       

       

      When we put call on hold , the IP phone sends send a new invite of in dialog with SDP parameter sendonly to PBX , but PBX does not forward same SDP parameter to trunk.Instead PBX originate new invite (in dialog with sendrecv).

       

      This new invite is strange to our Application and more like PBX doing RTP manipulation instead of originating SDP sendonly parameter.

       

      Regards,

      A Sidd

        • Re: Call on hold (SIP Trunk)
          Employee

          A Sidd,

           

          Thank you for posting your question.  Can you upload your current configuration along with the output from a debug voice verbose and debug sip stack messages (both enabled at the same time) while you recreate the issue to our FTP server with the instructions below?

           

          Open Internet Explorer web browser on their PC
          Type the following URL:  ftp://ftp.adtran.com

          Press the Alt key, click View, and then click Open FTP Site in Windows Explorer

          Double-click the "Incoming" folder
          Drag and drop files from PC into the Internet Explorer window

          Reply to this post with the exact filenames used so we can retrieve the files


          Thanks,

          Matt

          • Re: Call on hold (SIP Trunk)
            jayh Hall_of_Fame

            onlysim wrote:

             

            Scenarios Call on Hold.

             

            When we put call on hold , the IP phone sends send a new invite of in dialog with SDP parameter sendonly to PBX , but PBX does not forward same SDP parameter to trunk.Instead PBX originate new invite (in dialog with sendrecv).

             

            This new invite is strange to our Application and more like PBX doing RTP manipulation instead of originating SDP sendonly parameter.

            This sounds like normal behavior depending on how (and if) you are sourcing music-on-hold. 

             

            The IP phone is sending an in-dialog reinvite telling the PBX to stop the outbound RTP stream from the phone and invite the MOH source to now send audio to the remote party.  It's sendonly because the audio coming from the remote party isn't going anywhere useful as far as the IP phone is concerned. 

             

            If the external trunk is SIP, the PBX will then reinvite the remote party away from the phone and to the MOH source which may be internal to the PBX.  It's possible that the sendrecv is used to allow the remote party to escape from hold by connecting the receive stream to a DTMF receiver, or it may just be convenient to do so in all cases regardless if there is anything actually listening to the audio coming from the on-hold party. 

            1 of 1 people found this helpful
              • Re: Call on hold (SIP Trunk)
                onlysim New Member

                Hi Jayh,

                 

                Thanks for details , it is possible I avoid this normal behavior.

                In our setup we have IMS Application server and we want call hold and call transfer should be handled by that server.(PBX transparently forward SIP to SBC and ).

                 

                Regards,

                Sim