Difference between revisions of "One-way audio or not audio at all"
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* Make sure you are using same RTP range as your provider | * 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. | * 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 | |||
Revision as of 19:24, 8 May 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
