Difference between revisions of "One-way audio or not audio at all"
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= 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.
- Restart Asterisk after that
