Hello everyone!
first, let me say that i’m a noob on the whole PBX argument; anyway, i’m experiencing some problems…
sometimes i get one-way-audio for external calls (we can hear them but they cant hear us) and this is totally inconsistent, as per following picture! (red dots are calls where audio was one-way, green is where it works fine)
Sounds like the same issue, in that you are having one-way audio. Why are you having one way audio? There is likely a port/firewall issue. RTP ports are defaulted to 10000-20000 UDP. I assume this is behind a firewall?
Google: “One way audio freepbx” for many (many!) examples and resolutions.
edited: 12:45 italy, 26/7/2023
I realize that i’m stupid and i forgot to mention one important thing: last week i had the whole system running correctly, for reasons, i had to made a new installation on a new console, i copy-pasted manually all the infos on the new pbx from the existing pbx, so im 99% sure that the problem is on the new BPX configuration (i’m worried i forgot some specs, expecially on the trunk)
Not gonna lie, i have no clue how to be sure that those settings are correctly configured
thats what i feel too
obviously i already tired (like 15 different pages, most of them from this forum)
Asterisk SIP Settings are under Settings → Asterisk SIP Settings. Both External Address and the Local Networks section need to be correctly filled out with your information.
Hello, someone that would like to spend so time reading call logs and try to find out what is wrong and why the problem is totalyl inconsistent/random?
This isn’t going to be a log reading problem, it is more likely that you are having a firewall/port issue. If the calls are having no trouble connecting, then your signaling port is likely fine. You are going to need to make sure that the right ports are flowing in from the right places, especially your RTP ports. You’ll likely need Wireshark to capture network traffic.
thats what i’m asking, if someone can read those data from me, because i can get them but i’m clueless on what to search/read…
the problem is very random, as per following screenshot, so i think the best way is just to read what is actually going on during the calls and try to understand what are the differences between a good call and a broken on!
Hello!
i called the provider of the trunk and they confirm there are no problem from theyr side
in the meantime, i did:
asterisk -rvvv and sip debug on
i copy-pasted results from a broken call and a good call
reading the results (ngl, i’m clueless about what the hell i’m reading…) i see differences [i will keep updating]
Got SDP version 20070 and unique parts [- 20070 IN IP4 192.168.0.133]
Got SDP version 20069 and unique parts [- 20069 IN IP4 192.168.0.133]
The provided logs don’t show anything wrong. As already mentioned by others the best way would be to capture the network traffic of a bad call and open the captured file with the tool Wireshark. In Wireshark you can inspect the rtp traffic of a bad call and maybe see if the rtp packets are flowing in just one direction.
You can capture the network traffic by running tcpdump -i eth0 -w some-file-name.pcap on your pbx.
In Wireshark you can use the option Telephony > VoIP Calls to inspect the calls and playback the audio of calls.
