I have a problem, if I call a reachable number everything is ok.
But if I call an unattainable or nonexistent number, I do not receive audio, silent for minutes. In the old version of freepbx this did not happen, which module or configuration blocks audio ringing ?
Do I understand correctly, trunk is chan_sip but extension is IAX? If not (more than one PBX involved, etc., please provide details).
Otherwise, does this also fail when using a chan_sip or pjsip extension?
Do you have the RTP port range forwarded in your router/firewall?
If you still have trouble, possible approaches:
At the Asterisk command prompt, type sip set debug on
and make a test call. The SIP trace will appear in the log and may have clues as to whether the provider was directed to send audio correctly to the PBX.
Or, run tcpdump on the PBX, make a test call, stop the capture, move the file to your PC and open it in Wireshark. You can see whether the provider is sending audio, whether it’s in the correct format and not silent and whether it’s being passed to the extension.
Or, configure an IP phone or softphone to connect to the provider directly, bypassing the PBX. Confirm that audio is indeed present when calling the unreachable number.
I rehearsed with the voip provider directly, and it works
although I use everything in chain_sip is the same, does not change anything.
I do not receive the audio “the number called is temporarily unavailable” or “the number called can be turned off or unreachable”
I tried a freepbx connection directly to the main router, but not working.
SIP Status 480 Temporarily not Available, the called number is off, but I do not receive the return audio from the external provider.
as a last resort, I will try to delete and redo the database asterisk, with the latest upgrades and backups of previous versions of freepbx I think something happened in the recovery configuration.
The trace you posted is essentially unreadable – the forum replaced multiple spaces with one and broke lines into multiple. If you post another, please paste it at https://pastebin.freepbx.org/ and post a link here.
We see only communications between PBX and trunk provider? Surely packets between extension and PBX should have appeared.
And every SIP request and response appears twice? Do you know what may be causing that? It’s not retransmissions, because the timestamps are identical.
However, we do see alaw audio packets in both directions. If the extension traffic can be shown, you can see whether audio from the trunk is being properly passed to the extension. If it is, you can capture the raw packets with tcpdump and open the capture in Wireshark. Then, play the incoming audio and confirm that it is not silent or corrupted.
We have a moving target here. In post 7, Eutelia sent a 183 with some audio, but your capture did not include enough info to know why it wasn’t sent on to the extension, or why it was not heard.
In post 9, Eutelia sent a 480 without any prior audio. In this case, it’s up to the extension to play a busy signal and/or indicate a failed call on its display.
If both of these test calls were to the same number, it’s possible that the provider was using different upstreams, or that the MNO treated the calls differently, e.g. in one case the mobile was registered on the network but did not respond to the page.
I installed on another freepbx machine and inserted a trunk and an extension, and everything works perfectly.
But as soon as I installed the Extension Settings module, I had the same problem on the new machine. Removed this form everything works fine.
I do not know why!
IAX is a more powerful protocol, I can use it on remote phones, softphones, and I do not need to open ports on the router to work.
In recent years, ISP operators block a lot of network traffic and reserve port 5060 for their VoIP router, I’m having a lot of difficulties on Freepbx systems that I have installed and that have been working for years, because of the Fibra ADSL. IAX has met me so far, and it works great, only I have this little problem that does it on the latest Freepbx releases even when it is installed, and I would like to solve it. Convenient to have an extension on the mobile phone so you have the office wherever you go!
That is highly debatable, however, IAX has not seen any real development or improvements for the past 6+ years or so. Anything that is done with it now is due to community members making those changes. There isn’t anything being actively done with it.
It is also a protocol that only exists if you have Asterisk or FreeSWITCH. This isn’t a globally used protocol and it has a lot of limitations compared to SIP.