rtp.conf updated and attached.
I restarted asterisk.
Re-tried tests–same results.
Test 1 - my landline to my flowrouteDID - no audio at all
Test 2 - my T-Mobile calling my flowraouteDID:
You need to describe your network configuration. Audio problems are often related to NAT problems. STUN and other NAT helpers are usually not necessary once you understand your network and have some control over your router. Also, since you mentioned Fedora, there’s always a good chance that SELinux will break things when it’s active.
I think it makes only sense to look at the Asterisk configuration, if it is explicitly excluded that the root of the problem is outside Asterisk. This usually means a couple of network tests.
So SELinux is active, but permissive. I realize it may be a problem and beg any help that can be provided on this issue–I seem to remember it is possible to “set it to off” with a mandatory reboot. Should I try that?
Also,I just turned the local firewall (on the PBX) off and tested to find both tests still failing in the same manner. I have restarted that firewall but can easily take it down, as needed.
You may be right, or not. You need to capture some VoIP traffic on the LAN and the WAN side of your router (or check the the firewall states, in case this is available). I know cases where the service provider reuses the ports from incoming traffic (to their servers) for signalling and then port forwarding is relatively pointless as it is assumed that the existing connection stays open all the time. In cases like that you need to configre something like “outbound NAT rules” (pfSense jargon) or so-called “1-1 NAT”, which is a bit more general.
STUN may help in some scenarios, but I tend to forget which cases can be handled, so I’d rather look at what is actuall going on.
I installed Asterisk using “dnf install asterisk”.
I have only been using the default installation and no purchased modules to the best of my knowledge given I trust the Fedora repo to have a legitimate installation.
Perhaps you can help me determine if something is wrong with regards to this. Here is what I have:
*CLI> module show like 729
Module Description Use Count Status Support Level
format_g729.so Raw G.729 data 0 Running core
res_format_attr_g729.so G.729 Format Attribute Module 1 Running core
When I place a call to flowroute I see:
*CLI> core show channels verbose
Channel Context Extension Prio State Application Data CallerID Duration Accountcode PeerAccount BridgeID
PJSIP/flowroute-0000 from-trunk myflowrouteDID 19 Up Read digit,custom/no-solicitor +myLandline 00:00:07
1 active channel
1 active call
3 calls processed
*CLI> core show codec 19
19 G.729A (g729)
Not counting my op (where I totally screwed the tcpdump captures), subsequent tcpdump-captures should contain all inbound and outbound packets from my PBX. Is it possible what you are discussing is in those “watch” files?
Additionally, and I was hoping to avoid discussing this because I want to focus on one issue at a time, I have my PBX connected to another Asterisk PBX as follows:
MyPBX (192.168.1.x) NAT
MySPEdgerouter to Internet
Internet to private router (192.168.10.x)
private router to remotePBX (192.168.10.x) NAT
I am able to call and receive calls over this connection with audio working normally.
That-said, it uses a different codec:
*CLI> core show channels verbose
Channel Context Extension Prio State Application Data CallerID Duration Accountcode PeerAccount BridgeID
PJSIP/6003-00000005 my-phone 2058 1 Up Dial PJSIP/xxx(remoteext)@RemotePBX myCIDdata 00:00:16 049e372d-4425-4a41-8
PJSIP/RemotePBX-0000 from-trunk 1 Up AppDial (Outgoing Line) xxxx 00:00:16 049e372d-4425-4a41-8
2 active channels
1 active call
5 calls processed
*CLI> core show codec 1
1 Codec 2 (codec2)
Perhaps that means I am having a problem with my g729 codec???
I am not sure what to ask.
I think what you are telling me is that (from my above osts) I am using g729 but don’t have the codec. I also thought my config (from op) showing:
disallow=all
allow=ulaw
means my Asterisk PBX shouldn’t use g729.
But I am getting no-audio because it actually is g729.
I would sincerely appreciate any information to help me figure this out.