One-way audio or not audio at all
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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.