I have two Asterisk 18.15 at two branches (ipsec connected) talking with eachother through iax. Each of them has some PJSIP endpoints and people call each other.
Just to say in advance, since I also use iaxmodem for fax, I have jitterbuffer=no in general section of my iax.conf, since jitterbuffer was not a very good thing for fax transmission.
The peer definition in iax.conf is quite simple:
[pbxconnection]
type=friend
username=pbxconnection
host=other-host
fromdomain=other-host
secret=secret-password
context=default
notransfer=yes
qualify=yes
qualifysmoothing=yes
qualifyfreqok=60000
qualifyfreqnotok=10000
allow=alaw
Everything works as expected. Only problem is that users often complain about short audio cuts, like audio is away for some milliseconds.
So I thought, why not play a bit with jitterbuffer and added jitterbuffer=yes to the peer definition.
What happened then is that users reported that audio is much better now. But, I don’t have any ringback anymore. So if I call an endpoint at the other side, I don’t hear any ringing (or busy) indication. It’s just silence until the other party picks up.
And this is even the case if I put answer/ringing in dialplan before dialing to the endpoint at the other side.
Anyone faced this issue?