Difference between revisions of "One-way audio or not audio at all"
From Kolmisoft Wiki
Jump to navigationJump to search
| Line 30: | Line 30: | ||
= Device is H323 = | = 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. | ||
** [[Image:h323_binaddr.png]] | ** [[Image:h323_binaddr.png]] | ||
** | ** After this, restart Asterisk. | ||
<br><br> | <br><br> | ||
= See also = | = See also = | ||
* [[One way audio problems]] | * [[One way audio problems]] | ||
Revision as of 10:57, 27 April 2010
- /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 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.
- After this, restart Asterisk.
