Difference between revisions of "One-way audio or not audio at all"
From Kolmisoft Wiki
Jump to navigationJump to search
| Line 6: | Line 6: | ||
* rtp debug, watch audio is comming/leaving or not | * 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) ('''in provider/device settings''') | ||
* Make sure you are using same RTP range as your provider | * Make sure you are using same RTP range as your provider | ||
* Make '''canreinvite = no''' in device settings | * Make '''canreinvite = no''' in device settings | ||
Revision as of 16:04, 12 October 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) (in provider/device settings)
- Make sure you are using 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
- 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
