@amartin13 wrote:
So, I'm testing out Asterisk 13 / FreePBX 13 latest build everything up to date.
I can register with both SIP_CHAN and PJSIP no issues. I can also dial an the PBX answers.
I have a PBX on a 10.0.0.0/24 network.
I have a laptop with softphone on a 192.168.1.0/24 network
I have I firewall forwarding from an external ip of say 1.2.3.4I ran tcpdump and get 10.0.0.2 -> 1.2.3.4 for RTP
With SIP_CHAN everything works fine and I get audio and can send audio.
RTP Packets Look as follows in wireshark.192.168.1.x -> 1.2.3.4 to PBX
1.2.3.4 -> 192.168.1.x from PBX
192.168.1.x -> 1.2.3.4 to PBX
1.2.3.4 -> 192.168.1.x from
PBX192.168.1.x -> 1.2.3.4 to PBX
1.2.3.4 -> 192.168.1.x from
PBX192.168.1.x -> 1.2.3.4 to PBX
1.2.3.4 -> 192.168.1.x from PBXbut
When I use PJSIP I get from wireshark
192.168.1.x -> 10.0.0.2
192.168.1.x -> 10.0.0.2
192.168.1.x -> 10.0.0.2I've tried entering my external IP, my network ob both the General SIP and PJSIP Pages.
I also changed setting default SIP type to "YES" under Advanced settings. I also tried turning off CHAN_SIP.I see in the documentation that Asterisk recommends settings of local_net, external_signaling_address, and external_signaling_port. Are these being set somewhere I can't seem to find it.
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