Hi there, I deployed my first FreePBX server and calls from Ip Phones works fine, but with softphones calls disconnect after 32 Sec and additionally RTP packets don’t receive to the destination. I’ve captured network traffic and wireshark shows that all call setup packets are sent to server internal ip address and both RTP packets and ACK to Answer are sent to server external ip address.
Can anyone help me, why and how can solve it?
Thanks in advanced
I’ve written a blog post about this[1]. If you are using chan_sip then there are different options for this. Generally though your problem is that NAT settings likely aren’t set.
[1] https://blogs.asterisk.org/2020/01/01/sip-and-rtp-routing/
check this site
https://hsunryou.blog.me/221706370385?Redirect=Log&from=postView
Thank you @jcolp for your post, I read it and set “RTP Symmetric” and reset “Force rport”, but the issue still. RTP Packets and ACK to Answer are sent to server external IP address.
If the problem is on the server receiving packets then you’d need to ensure that the firewall is open and forwarding packets to Asterisk.
Wow, fast reply, Thanks. IpPhones and SoftPhones are both only extensions and in a building (in same LAN) with different ip ranges. Ip Phones are in server ip ranges (10.12.110.xxx) but softphones (10.12.30.xxx), and I’ve access form 10.12.30.xxx to asterisk server (10.12.110.xxx) . pings reply successfully.
Additionally I opened every port on server ip address, but the issue still.
Then I’d suggest doing a “pjsip set logger on” on the Asterisk server and seeing if you’re getting the traffic. If you aren’t, then the traffic is getting blocked/stopped outside of Asterisk.
Thank you @jcolp, great help. There was some IPs that must be add to NAT setting.
This topic was automatically closed 30 days after the last reply. New replies are no longer allowed.
