Users Can Not Hear The Audio Playback Sometimes

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Asterisk Users 6 Comments

It would be easier to comment if you would copy the relevant part of the Dialplan.

The only dial cmd I see is Dial(“SIP/1002-00000292”, “DAHDI/g1/9999999999,300,wW”)

At least where I live, 9999999999 can be part of a valid telephone number. You could increase the verbosity and see what Dial() is actually doing.

jg

6 thoughts on - Users Can Not Hear The Audio Playback Sometimes

  • Try calling Progress() just before the dial command. Without this Asterisk wont send the SIP/183 Session Progress and send the inband audio until the call is answered.

  • Thanks Gareth , Do I need to change something in asterisk dial plan ? I am using freepbx to mange asterisk graphically.

  • Yes you will see a section called [macro-dialout-trunk]
    Within that there will be a line line :-
    exten => s,n,Dial(something….)

    Just before that line add :-
    exten => s,n,Progress()

    You will then need to reload the dialplan (dialplan reload from the asterisk prompt) and you can give it a go.

  • Thank you Gareth,

    It worked like a charm.

    The only problem I am having is now, when I do some changes in my freepbx and reload it just rewrites my dial play , I will try to fix it though.

    Thanks again

    Cheers Shantanu

  • I did see in the console output it doing a GotoIf() and checking if a custom trunk was defined just before the original dial command. Maybe there is a way you can define a custom trunk and basically copy the standard part of the dialplan but with the extra line you added?
    Sorry I dont know the product so cant tell you how to do it.

  • No Problem, it was easy to fix. There is a file which is ignored by freepbx while doing the reload, so i copied my trunk into that file.

    thanks shantanu