One-way audio or not audio at all

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Revision as of 09:27, 11 May 2011 by Mindaugas (talk | contribs)
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  • /etc/init.d/iptables stop



Device is SIP

  • rtp debug, watch to see whether or not audio is coming/leaving
  • Make sure sip.conf is correctly configured if Asterisk is behind NAT
  • Make sure you are using correct codecs (same codecs in all the path of the call). Some devices does not support some codecs. Make tests using 1 same codec across the call path. Disable all others.
  • Make sure NAT settings are correct (nat=yes) (in provider/device settings)
  • Make sure you are using the same RTP range as your provider
  • Make canreinvite = no in device settings
  • Remove T.38 Support from device settings
  • In /etc/asterisk/sip.conf find lines:
t38pt_udptl = yes
t38pt_rtp=no
t38pt_tcp=no

and change to:

;t38pt_udptl = yes
;t38pt_rtp=no
;t38pt_tcp=no

Reload Asterisk.




Device is H323

  • If while connecting with an H.323 client, you get no audio or garbled audio and messages like this on the Asterisk console: rtp.c:489 ast_rtp_read: Unknown RTP codec 107 received, try to disable Speex or some other codec on Asterisk and/or the client side.
  • If there is no audio in H323 calls, make sure in /etc/asterisk/h323.conf that you have binaddr set to your server's real IP address.
    • H323 binaddr.png
    • After this, restart Asterisk.




See also