In Asterisk SIP Settings, confirm that External Address and Local Networks are correctly set. If you change these, after Submit and Apply Config, you must restart Asterisk.
If you still have trouble, at the Asterisk command prompt type pjsip set logger on
or sip set debug on
according to whether the trunk is pjsip or chan_sip.
Make a failing call in, paste the Asterisk log (not the console output) for the call at pastebin.freepbx.org .
As you are too new to post links, just post the last eight hex characters of the link here. Also post firewall make/model and any VoIP related settings. Confirm that hardware firewall has a public IP on its WAN interface. Confirm that VM is using bridged networking (local IP of PBX is in the same subnet as the LAN presented by the hardware firewall).
We have uploaded the log file, see upload under tapmetags42
For our inbound call test, we used one of our Sipstation: DID Verification numbers in an attempt to rule out any setup we might have influenced. This was all standard FreePBX install and SIPSTATION activation.
Again, the same number works fine on our OLD Asterisk, very strange.
OLD Asterisk 13.12.1
NEW Asterisk 16.20.0
Both the OLD and NEW servers share the same firewall and network.
We physically turn off one server before starting the other server.
They are never both started at the same time.
we use the same Internal IP address on both the OLD and NEW, since we did not want two Asterisk servers attempting to connect to the one (same) SIPSTATION account.
Sorry, I am having trouble navigating 6800+ lines of logs without the SIP trace as a guide.
I suspect that the contact rewrite on line 3707 is related to the trouble, but I can’t find an analogous rewrite of the media address in the SDP.
As requested previously, at the Asterisk command prompt type pjsip set logger on
before making the test call. The log should now include the incoming INVITE and the 200 OK response given by Asterisk before it plays “you have reached …”
Line 79 shows audio requested to a private address, which obviously won’t work.
In Asterisk SIP Settings, confirm that External Address and Local Networks are correctly set. The pjsip-specific settings for the transport should have External IP Address and Local network left blank. If you change these, after Submit and Apply Config you must restart Asterisk.
Confirm that ESXi is using bridged networking (PBX LAN address is in the same subnet as the ESXi host).
Confirm that your hardware firewall has SIP ALG (or other SIP ‘helper’) turned off and allows traffic to the Bandwidth media server (188.8.131.52/20).
Confirm that the trunk settings has Rewrite Contact: No and RTP Symmetric: Yes.
If you still have trouble, please post:
Contents of /etc/asterisk/pjsip.transports.conf
Contents of /etc/asterisk/pjsip.transports_custom.conf or /etc/asterisk/pjsip.transports_custom_post.conf, if present
Make/model of hardware firewall and any VoIP-related settings.