PJSIP Trunk for FXO Behind Nat

Hi All,

We recently converted a server from the chan_sip driver to only pjsip.

Everything is working, but we’re having one way audio issues with an FXO, an Obi 110.

We have this device setup via a Trunk entry. Here’s what we had on chan_sip:

username=8880
type=friend
transport=tcp
secret=somepassword
qualifyfreq=25
qualify=yes
nat=yes
host=dynamic
dtmfmode=auto
context=from-pstn
canreinvite=no

Note the nat and dynamic parts.

I cannot figure out how to replicate this on the pjsip trunk, and therefore we’re having one way audio issues.

I read on this wiki that I need to set the following settings to replicate chan_sip’s “nat=yes”, but I can’t find them in the FreePBX UI under Trunks, only under Extensions, and this device needs to be a Trunk.

https://wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip

rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes

Anyone know how I can have a NAT’d dynamic host on pjsip?

Anybody?

You mean these settings in the Advanced tab under PJSIP settings in the trunk? Which, BTW, are set to “yes” as their default.

2019-10-12%2012_02_58-Window

You need to describe this issue a bit better. Which side is getting the one-way audio? Who can’t hear who? Where is this FXO device compared to the PBX? Etc. etc. etc…

Thank you.

We do not have those setting under the advance tab. Here’s what we have:


We’re on FreePBX 13.0.197 and Asterisk 13.23.1

Perhaps we just need to upgrade?

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