Problem with RTP packets


I’ve setup freepbx 15 and asterisk 16 on a debian 10.

My phone and freepbx are on the same LAN (
The freepbx deetect wan IP address correctly to 90.49.A.B / the local network is in the advanced sip setting.

The trunk SIP with OVH is OK.

When i call from external phone it’s ok, i’ve the sound.
When i call from local extension to the asterisk like *97 i’ve have sound.

And the probleme is when i call between local extension. for exemple 4003 (cisco 7942G) to 4002 (softphone on my computer: twinkle).

I see with wireshark, the SIP signaling is OK : From (asterisk) and (my computer) -> Trying / Ringing.

in the SIP INVITE i see that:
Transport: UDP
Send-by address: 90.49.A.B
Send-by Port: 5060

After all RTP paquet are send by computer ( to the WAN IP (90.49.A.B). And no sound between extension. And afteer 30 seconds, the asterisk send SIP BYE with reason Q.580 Cause 44
ANd in the asterisk log i see no activity RTP since 30seconds (timeout).

Thanks for help.

In the Settings/SIP Settings page, in the NAT section double check that you have your LAN specified in the Local Networks section as
Also, are you using PJSIP or SIP channels?

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My config:
So i use only pjsip on internal. Sip driver is only for the trunk with OVH

After i’ve checked, and with show advanced setting to yes.

I change that: 2021-04-12_20-51

to that:2021-04-12_20-51_1

and it’s works. So i don’t understand why i need to change this. In my setup i’ve made in the past i don’t need to change those parameters.

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