One-way audio (cannot hear recipient) on outbound calls

Asterisk Asterisk SIP
1

Hi.

I’m having an Issue with outbound calls (inbound calls are fine, two way audio). The remote party here’s me but I cannot hear them.

I don’t see any specific error when debugging is on.

I have:
qualify=yes
nat=yes
directmedia=no

In general settings.

The phones are connected directly to a server running asterisk.

I can provide any output necessary, Just not sure what I should provide :slight_smile:

I haven’t had issues with other setups using the same configuration / server so I’m a bit confused as to what Is going on.

2

Issue was with the wrong codec on the phone settings just for any extreme noobs like me out there.

3

In current version of Asterisk nat=yes is deprecated, use nat=force_rport,comedia

4

Use the nat options, if any, that you actually need. I think the reason for deprecating it was to stop people always turning everything on.

Note that nat= is primarily for cases where the peer is inside and Asterisk outside. It is not the primary NAT setting for when Asterisk is inside.

5

It is known that nat= is only needed for cases where the peer is inside and Asterisk outside, but my comment was only just as an aside about a deprecated option, in case he need it on the future, but thanks for adding extra information about the correct use of nat

6

It’s not generally known that that is the case. My impression is that most people use it because they are inside NAT and see this nat= parameter and assume they have to set it (unfortunately, they often assume it is the only thing to set).

7

My Issue was the codec on the phone for some reason needed to be G729 to work correctly even though no codec was disallowed in the users.conf

8

You need a paid for codec if one side is G.729 and the other isn’t.