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

From Kolmisoft Wiki
Jump to navigationJump to search
Line 3: Line 3:
= Device is SIP =
= Device is SIP =


* rtp debug, watch audio is comming/leaving or not
* Make sure sip.conf are correctly configured if [[Asterisk under NAT | Asterisk is behind NAT]]
* Make sure sip.conf are correctly configured if [[Asterisk under NAT | Asterisk is behind NAT]]
* Make sure NAT settings are correct (nat=yes)
* Make sure NAT settings are correct (nat=yes)

Revision as of 08:46, 6 July 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)
  • Make sure you are using same RTP range as your provider
  • Make canreinvite = no in device settings

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