I’ve been having this issue for a while where when someone makes a call there is no progress tone or any audio, but the call still goes through and the other side picks up, but I can’t hear them. This happens on occasion.
Below is a diagram that I got of one of the calls where I couldn’t hear anything when I called someone.
This was calling a number that is on the same SIP Trunk and FreePBX server.
Just to add, when I play the call from the pcap file (Which is where I got the diagram from) I can hear audio from both sides, so the number I was calling has an announcement message which I did not hear.
IMO, the call flow posted is not meaningful. The rightmost address is clearly not Asterisk, because Asterisk (by default) uses only ports 10000-20000 for RTP. So, the middle address must be Asterisk, which is consistent with RTP port 13434. It’s also consistent with the left address being the provider on port 5060 with Asterisk on port 5164.
So the INVITE is one sent from the provider to Asterisk, i.e. this is an incoming call. Now the OP did state that he called one of his own numbers, but if his problem is “no audio on some outbound calls”, why did he only post info about the inbound leg??
Could you please tell me what would be meaningful and helpful for assistance purposes, as opposed to the call flow logs that I provided?
Additionally, I can confirm that both of these lines are through the same asterisk system, just with different outbound CIDs. You do raise a good point that it is technically internal, as it is on the same Asterisk server. However, this call was made using a full CID and not an internal extension, so I can confirm that the call would have been sent to the carrier before reaching the destination. What further information could I provide that would be useful for additional troubleshooting?
So the IP on the right is the providers and yes the middle one is the FreePBX server and the left one is also the providers.