2 Replies Latest reply on Oct 20, 2016 12:24 PM by michael56

    call forwarded to adtran have no audio

    garabedy New Member

      I am having an issue when an outside line is forwarding calls to a number on the 908e PRI. When the call is answered I get no audio. I am seeing a 89 Redir reason:DTE OUT OF ORDER when i run debug idsn-l2 formatted. 

       

       

      18:24:54.564 ISDN.L2_FMT PRI  1  ==============================================

      18:24:54.564 ISDN.L2_FMT PRI  1  T Sapi:00 C/R:C Tei:00 INFO Ns:33  Nr:33  P:0

      18:24:54.564 ISDN.L2_FMT PRI  1    Prot:08  CRL:2  CRV:029B

      18:24:54.564 ISDN.L2_FMT PRI  1    M - 05 SETUP

      18:24:54.564 ISDN.L2_FMT PRI  1     IE - 04 BEARER CAPABILITY   Len=3

      18:24:54.564 ISDN.L2_FMT PRI  1          80 Xfer Cap.:SPEECH

      18:24:54.565 ISDN.L2_FMT PRI  1          90 Xfer Rate:64k

      18:24:54.565 ISDN.L2_FMT PRI  1          A2 Layer 1:G.711 U-Law

      18:24:54.565 ISDN.L2_FMT PRI  1     IE - 18 CHANNEL ID          Len=3

      18:24:54.565 ISDN.L2_FMT PRI  1          A1 Primary Rate

      18:24:54.565 ISDN.L2_FMT PRI  1             Intfc ID:IMPLICIT

      18:24:54.565 ISDN.L2_FMT PRI  1             Pref/Excl:PREFERRED

      18:24:54.565 ISDN.L2_FMT PRI  1             D-Chan Indicated:NO

      18:24:54.565 ISDN.L2_FMT PRI  1             Chan. Sel:FOLLOWS

      18:24:54.565 ISDN.L2_FMT PRI  1          83 Numb/Map:NUMBER

      18:24:54.566 ISDN.L2_FMT PRI  1          81 Channel:1

      18:24:54.566 ISDN.L2_FMT PRI  1     IE - 6C CALLING PARTY #     Len=12

      18:24:54.566 ISDN.L2_FMT PRI  1          00 Numb. Type:UNKNOWN

      18:24:54.566 ISDN.L2_FMT PRI  1             Numb. Plan:UNKNOWN

      18:24:54.566 ISDN.L2_FMT PRI  1          80 Presentation:ALLOWED

      18:24:54.566 ISDN.L2_FMT PRI  1             Screening:USER PROVIDED

      18:24:54.566 ISDN.L2_FMT PRI  1             Ph.# 16265551212

      18:24:54.566 ISDN.L2_FMT PRI  1     IE - 70 CALLED PARTY #      Len=5

      18:24:54.566 ISDN.L2_FMT PRI  1          80 Numb. Type:UNKNOWN

      18:24:54.566 ISDN.L2_FMT PRI  1             Numb. Plan:UNKNOWN

      18:24:54.566 ISDN.L2_FMT PRI  1             Ph.# 7700

      18:24:54.567 ISDN.L2_FMT PRI  1     IE - 74 REDIRECTING #       Len=14

      18:24:54.567 ISDN.L2_FMT PRI  1          00 Numb. Type:UNKNOWN

      18:24:54.567 ISDN.L2_FMT PRI  1             Numb. Plan:UNKNOWN

      18:24:54.567 ISDN.L2_FMT PRI  1          00 Presentation:ALLOWED

      18:24:54.567 ISDN.L2_FMT PRI  1             Screening:USER PROVIDED

      18:24:54.567 ISDN.L2_FMT PRI  1          89 Redir reason:DTE OUT OF ORDER

      18:24:54.567 ISDN.L2_FMT PRI  1             Ph.# 16265557700

      18:24:54.585 ISDN.L2_FMT PRI  1  ==============================================

        • Re: call forwarded to adtran have no audio
          jay Employee

          Garabed, I wouldn't expect that message to have anything to do with audio connection. If anything, the setup of the call might fail, but that would disconnect the call all together. I would suggest opening a ticket with support if you haven't already for a more in depth troubleshooting. Thanks

          • Re: call forwarded to adtran have no audio
            michael56 New Member

            Garabedy,


            The Problem:
            When making audio calls using SIP the phone rings but when it is answered there is only one way audio or no way audio.

             

            What Cause One Way Audio:
            The cause of one way audio is a combination of NAT and STUN

             

            How to Achieve Two Way Audio:
            The solution is far simpler than you might think. Do is the following:

             

            On your VoIP switch reduce the SIP port range. How many SIP calls do you think you will have going at any one time max?
            Most of you reading this will be 10 at a guess, maybe 20. Reduce the max range to 49162 giving me 10 ports.
            On your NAT device set up port forwarding for the 10 ports to your VoIP switch.
            The reason this works is because when the VoIP learns it is behind a symmetric NAT via STUN it will instead tell the remote
            switch to send audio to it’s local (non NATted) ports. Since we reduced the port range to 10 and have now opened these ports
            manually on the NAT it will allow the audio to come in. This will eliminate one way audio.