One-way audio or not audio at all

• /etc/init.d/iptables stop

Device is SIP

• Check if your SIP/H.323 supports the codecs you use in MOR, for example eyeBeam does not support aLaw codec, so you will not have audio if you use aLaw as default codec for that device in MOR.
• 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 you are using correct codecs (same codecs in all the path of the call). Some devices does not support some codecs. Make tests using 1 same codec across the call path. Disable all others.
• 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.

Device connected to MOR is Eyebeam

Check if you are not using aLaw codec, as we have previously experienced problems when using Eyebeam and aLaw.

RTP debug

• To enable RTP debug, login into your server via SSH and open Asterisk CLI

• Next, enable RT debug by using this command

rtp debug

See also

• One-way audio problems