Pjsip calling sip URI always congestion/busy

Asterisk Asterisk SIP
1

Upgrading from Asterisk 16 to 22 and converting our config/dialplan from mod_sip to mod_pjsip and so far I can’t get outbound calls to connect at all.

I created a minimal endpoint for outbound:

[outbound]
type=endpoint
disallow=all
allow=ulaw

and in the dialplan:

Dial(PJSIP/outbound/sip:music@iptel.org)

and with verbose and debug and pjsip logging all I get is:

Called PJSIP/outbound/sip:music@iptel.org
== Everyone is busy/congested at this time (1:0/1/0)
– Auto fallthrough, channel ‘Motif/test\40singpolyma-beefy.lan-e4dc’ status is ‘CONGESTION’

no logs from pjsip even with pjsip set logger onand I even did a tcpdump on 5060 and saw nothing, I don’t think it’s actually getting to the part where it tries SIP. But I don’t see any local logs about why it’s not completing either.

2

Oh my, I finally found it. The default pjsip config didn’t come with any transports defined, so I added this:

[transport]
type=transport
protocol=udp
bind=0.0.0.0

now it works