How to resolve one way or no audio issues

Its a common issue with PBX to have audio issues like one way audio or no audio. Sometime only caller can hear remote party or remote party only can hear the caller. This is mainly because of NAT issues. We recommend to use NAT with enabling 10000-20000 UDP ports on firewalls and also to enable natting on trunks

Example

[didforsale1]
type=peer
host=209.216.2.211
nat=yes
canreinvite=no
disallow=all
allow=ulaw&g729
context=from-trunk
dtmfmode=auto

[didforsale2]
type=peer
host=209.216.15.70
nat=yes
canreinvite=no
disallow=all
allow=ulaw&g729
context=from-trunk
dtmfmode=auto

Second reason for causing one way audio. Its nothing rather than Codec, This is happening when a calls comes with ULAW and system tries to accept with other codecs which can cause superflous codec negotiation. To avoid this we need to remove the unwanted codec on your switch. For example if you are using G729 then remove ULAW parameter from our trunks. If you are using ULAW then remove g729.

Example

[didforsale1]
type=peer
host=209.216.2.211
nat=yes
canreinvite=no
disallow=all
allow=g729 (Use either one of the codec you want)
context=from-trunk
dtmfmode=auto

If both of the setups dont work and if you in a local network, you need to add additional parameters in /etc/asterisk/sip.conf

Under the general context add your Public IP, NAT and local IP

Example

[general]

externip= (Your public ip)
localnet= (your local network address)
nat = yes

Example, Say your public IP is 208.54.15.6 and your local network is 192.168.1.0

externip= 208.54.15.6
localnet= 192.168.1.0/255.255.255.0
nat = yes

Hope this will solve the issue

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