Difference between revisions of "One-way audio or not audio at all"
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* 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. | * 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. | ||
* Make '''canreinvite = no''' in device settings | * Make '''canreinvite = no''' in device settings | ||
* When no audio in H323 calls, make sure in '''/etc/asterisk/h323.conf''' you have '''binaddr''' set to your servers real IP address. | |||
Revision as of 12:24, 31 July 2008
- Make sure NAT settings are correct
- Make sure you are using same RTP range as your provider
- 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.
- Make canreinvite = no in device settings
- When no audio in H323 calls, make sure in /etc/asterisk/h323.conf you have binaddr set to your servers real IP address.
