Please help. Asterisk noob here. We had outbound calls working, and then all of a sudden we started getting this:
app_dial.c: Called PJSIP/9876541230@Livingston-Vitelity-543-210-4567
app_dial.c: Everyone is busy/congested at this time (1:0/0/1)
Executing [s@macro-dialout-trunk:34] NoOp("PJSIP/3325-00000061", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 21") in new stack
On the line it says, “All circuits are busy. Please try again later.”
Here’s a screenshot of my Outbound Route settings:
Trace the INVITE request not the REGISTER request, and assuming your provider require you need to be registered to allow you to call out, verify your SIP credentials,
The ‘401 Unauthorized’ is just the first step in the authorization dance. It’s a ‘challenge/response’ kind of thing. Note the ‘200 OK.’
I’m just a lowly ‘chan SIP’ guy, but ‘sip show registry’ (or the PJSIP equivalent) should show ‘Registered.’
As noted above, you should look at INVITEs. REGISTER tells your provider where you are. INVITE is the start of a call.
Personally, I like to use ‘sudo tcpdump -i any -n -s 0 -w call-issue.pcap port 5060’ to capture packets and ‘sngrep -I call-issue.pcap’ as my ‘secret decoder ring.’ The Asterisk console log with SIP debugging enabled can be useful as well. I just really like the way sngrep groups all the packets in a dialog and displays ladder diagrams. Comparing packets to identify differences can be a big help where needed.
