
When you call someone, there is only one way audio or worst, no audio at all.There are lot of reasons why you are facing this issue. Most of the time, there are few commons issue that you might want to check before taking any further action. Two common issues that I found all the time are : NAT and codec issue.
Problem : Asterisk only one way audio issue or no sound incoming call
1. Make sure that you enable NAT is enable in your configuration and also open 10000-20000 UDP ports on your router / firewalls. Those port are by default but if you want to change those ports you have to change also your your PBX server. Of course, you have to also enable NAT on trunks configuration that your using on your PBX
For example, you have to put nat to yes in your configuration if you are running your server on a private IP address.
[myTrunk]
type=peer
host=xxx.xxx.xxx.xxx
nat=yes
canreinvite=no
disallow=all
allow=ulaw
allow=g729
context=from-trunk
dtmfmode=auto
2. Sometime, you are using some codec that your provider does not support or vice-versa. So, before configuring your trunk, you have to find a common codec that your server support as well as your provider. After that, put the common codec. For example, let’s say, our provider support ULAW and as well as our server, we gonna setup codec as ULAW.
[myTrunk]
type=peer
host=xxx.xxx.xxx.xxx
nat=yes
canreinvite=no
disallow=all
allow=ulaw
context=from-trunk
dtmfmode=auto
After that, do not forget to reload your Asterisk or FreePBX server before start doing the test.
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