"random" one-way audio problems

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)

another problem is that external IVR doesnt get our inputs, so we cant navigate other’s people IVR

i’m ready to give all the Logs/info you may need to better investigate, thanks in advance for any type of help you may give!

Are your Asterisk SIP settings External and Local network IPs correctly configured?

Ports used by clients might be an issue. So nat is prime one, If you have a port being blocked at firewall is another.

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.


i’m confident that they are correctly filled!

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.

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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!

should i post Wireshark stuff?

i seen you linked in another post to: Providing Great Debug - Support Services - Documentation
but when i use tail -f /var/log/asterisk/full i get a totally different output compared to the guide…

if i try grep command, it doesnt even do anything at all…
# grep 1690451053.304 /var/log/asterisk/full*

No audio (especially if you do have audio on other calls with the same trunk) = no RTP. That is the difference.

  1. You can go up and down your stack and find out where your RTP range is off (different range, NAT/Prot forwarding issues, etc.).

  2. Or run Wireshark for a while and review a bad call when it comes up.

  3. You can also turn on SIP debugging on the full log and that may provide some clues.

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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]
Got SDP version 20069 and unique parts [- 20069 IN IP4]

Peer audio RTP is at port
Peer audio RTP is at port

huge last difference i found is that the following data show up only in the good call, it smissing in the broken call!

no one can help me find the problem? :frowning:

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.

i will try to do that, thanks a lot for now!

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