Difference between revisions of "One-way audio or not audio at all"

From Kolmisoft Wiki
Jump to navigationJump to search
Line 39: Line 39:


= See also =  
= See also =  
* [[One-way audio problems]]
* [[One way audio problems | One-way audio problems]]

Revision as of 12:28, 3 May 2010

  • /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 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

Retrieved from "https://wiki.kolmisoft.com/index.php?title=One-way_audio_or_not_audio_at_all&oldid=6846"