no audio

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Asterisk Users 5 Comments

Any reason why I don’t get audio on the channel after it rings and the
end user picks up.
Here are my files.

CONSOLE=Console/dsp ; Console interface for demo
OUTBOUNDTRUNK=SIP/callwithus

[default]
include => stdexten
exten => s,1,Answer()
exten => s,n,Wait(1)
exten => s,n,Dial(SIP/callwithus/1111444444,120,A,(demo-thanks))
exten => s,n,Wait(2)
exten => s,n,Hangup()

my sip.conf file

[general]
context=default
allowoverlap=no
bindport=5060
port=5060
bindaddr=0.0.0.0
canreinvite=no ;if your asterisk box is behind a NAT ro

;register => xxx:yyy@carrier.callwithus.com
register => xxx:yyy@sip.callwithus.com

[callwithus]
type=friend
host=sip.callwithus.com
username=xxx
secret=yyy
qualify=no
insecure=invite

5 thoughts on - no audio

  • Steve,
    thanks for your note

    negative. no joy.
    removed the line to make is very basic. see below.

    [globals]
    CONSOLE=Console/dsp ; Console interface for demo
    OUTBOUNDTRUNK=SIP/callwithus

    ;[general]

    [default]
    include => stdexten
    exten => s,1,Answer()
    exten => s,n,Wait(1)
    exten => s,n,Dial(SIP/callwithus/1111444444)
    exten => s,n,Wait(2)
    exten => s,n,Hangup()
    ~

    href=”mailto:sedwards@sedwards.com”>sedwards@sedwards.com      Voice: +1-760-468-3867 PST

  • Un-top-posting…

    Crank up the verbosity and debugging levels, check the codecs, etc.

    Does ‘sip set debug on’ give any clues?

  • Hi,

    I have a server running at more than two years with Asterisk 1.6, and began presenting problem seedlings links in external SIP extensions on some links.

    By doing “rtp set debug on” discovered the problem, he is trying to deliver the audio directly to internal IP Extension. And sometimes shown correctly on the external IP, where this time the link works correctly. These extensions are at Stake sip configured with nat = yes and = no canreivinte.

    In the “sip show settings” I have my ip correctly list “externip: MY_IP:
    5060” Server is not behind NAT, the IP is directly on it, it has only one network card, the SIM remote extensions are behind NAT.

    What may be occurring in some links for it to work correctly and not others?


    Atenciosamente Daviramos Roussenq Fortunato

  • I don’t know if it will resolve, but try add the “localnet” option in sip.conf.