Problem with RTP packets

Hello,

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

My phone and freepbx are on the same LAN (192.168.10.0/24)
The freepbx deetect wan IP address correctly to 90.49.A.B / the local network is 192.168.10.3/24 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 192.168.10.3 (asterisk) and 192.168.10.140 (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 (192.168.10.140) 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 192.168.10.0/24
Also, are you using PJSIP or SIP channels?

1 Like

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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