I run a well functioning Asterisk server in version 15.2.1 with pjsip for SIP over all transports (UDP, TCP, TLS, WebSocket). I use it with a WebRTC WebPhone (based on SIP.js) as well as classic SIP clients and phones.
After a test upgrade to Asterisk 15.2.2 and 15.3.0 (both tested, same result) I have the following problems:
- I can not hear any sound (not even a ringing tone, no playback in ConfBridge)
- If the other party accepts the call, the call ends after one second
I found the following:
- After canceling the call, the following appears in the log:
== Everyone is busy/congested at this time (1:0/0/1) -- Executing [0123456789@sipuser-sipprovider-out:3] Hangup("PJSIP/asteriskaccount-00000002", "") in new stack
- In comparison with version 15.2.1 the following is missing in the log of a call:
> 0x7fd59c039190 -- Strict RTP learning after remote address set to: 192.168.0.2:8000
Does anyone have any idea what the problem may be?
Here’s the log of the call: https://pastebin.com/c00TTWSu
126.96.36.199 = Asterisk server
188.8.131.52 / 192.168.0.2 = My PC (external/internal)
345.345.345.345 / sipprovider.com = SIP provider