Problem with audio in asterisk 18.6

Hi,
I am using asterisk 18.6 and CentOS 8.3.

I start an audio call, but I have problems with the audio.
I start a call from softphone 1 to a mobile, but from the softphone I can not hear nothing.

Both devices are in the same office, and using the same local net.

[siptrunk]
type=peer
disallow=all
allow=alaw
allow=ulaw
allow=g729
host=100.64.216.4
fromdomain=100.64.216.4
qualify=yes
dtmfmode=rfc2833
nat=forece_rport,comedia
context=from-trunk
sendrpid=yes
trustrpid=yes

You’ll need to provide logging, but meanwhile,

forece_rport is not a valid option! Could you explain why you need the nat option, as, if really needed, it indicates a significant, unusual network configuration. NB If Asterisk is behind NAT, you need to tell it how to find its public address and nat= does not do that.

The only devices you have identified are the soft phone and the mobile. I assume the mobile is connected through the mobile network, so I don’t understand why being in the same office is relevant.

Also, you haven’t provided any part of sip.conf that relates to devices in the office.

You should also note that chan_sip is now deprecated, so you should be trying to reproduce this on chan_pjsip.

Do you have a licence for the G.729 codec? If transcoding becomes necessary and you don’t, you could end up with no audio.

When i stop internet then audio is working properly and when internet start, at that time audio is not working

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