Difference between revisions of "One-way audio or not audio at all"
From Kolmisoft Wiki
Jump to navigationJump to search
| Line 1: | Line 1: | ||
* Make sure NAT settings are correct | * Make sure NAT settings are correct (nat=yes) | ||
* 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. | ||
| Line 6: | Line 6: | ||
** [[Image:h323_binaddr.png]] | ** [[Image:h323_binaddr.png]] | ||
** Restart Asterisk after that | ** Restart Asterisk after that | ||
* Make sure sip.conf are correctly configured if [[Asterisk under NAT | Asterisk is behind NAT]] | |||
Revision as of 12:34, 1 July 2009
- Make sure NAT settings are correct (nat=yes)
- 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.
- Restart Asterisk after that
- Make sure sip.conf are correctly configured if Asterisk is behind NAT
