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

From Kolmisoft Wiki
Jump to navigationJump to search
Line 33: Line 33:
** [[Image:h323_binaddr.png]]
** [[Image:h323_binaddr.png]]
** Restart Asterisk after that
** Restart Asterisk after that
<br><br>
= See also =
* [[One way audio problems]]

Revision as of 15:21, 17 November 2009

  • /etc/init.d/iptables stop



Device is SIP

  • rtp debug, watch audio is comming/leaving or not
  • Make sure sip.conf are correctly configured if Asterisk is behind NAT
  • Make sure NAT settings are correct (nat=yes) (in provider/device settings)
  • Make sure you are using 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

  • While connecting with H.323 client you get no audio or garbled audio and messages like this on Asterisk console: rtp.c:489 ast_rtp_read: Unknown RTP codec 107 received. Try to disable Speex or some other codec on Asterisk or/and client side.
  • When no audio in H323 calls, make sure in /etc/asterisk/h323.conf you have binaddr set to your servers real IP address.
    • H323 binaddr.png
    • Restart Asterisk after that




See also

Retrieved from "https://wiki.kolmisoft.com/index.php?title=One-way_audio_or_not_audio_at_all&oldid=5683"