[asterisk-users] congested/busy on trunk?

[John Roman]

greetings asterisk users :)
ive just deployed version 17 and migrated as best I can to pjsip.  I can
receive calls, and get to my mailbox prompt, however placing calls seems
impossible with the following error on dial:

Connected to Asterisk GIT-master-0cde95ec89 currently running on dunkel (pid = 517890)
dunkel*CLI>
dunkel*CLI>
  == Setting global variable 'SIPDOMAIN' to 'ringythingy.dev1ce.com'
    -- Executing [blah at anveo_sip:1] Dial("PJSIP/demo-alice-00000005", "PJSIP/blah at mytrunk") in new stack
    -- Called PJSIP/blah at mytrunk
    -- PJSIP/mytrunk-00000006 is ringing
    -- PJSIP/mytrunk-00000006 is ringing
    -- PJSIP/mytrunk-00000006 is making progress passing it to PJSIP/demo-alice-00000005
       > 0x7ff39839e360 -- Strict RTP learning after remote address set to: 72.9.156.128:52642
       > 0x7ff3983994c0 -- Strict RTP learning after remote address set to: [2605:e000:130a:fb:517d:7894:9482:c2bd]:54006
    -- PJSIP/mytrunk-00000006 is making progress passing it to PJSIP/demo-alice-00000005
  == Everyone is busy/congested at this time (1:1/0/0)
    -- Auto fallthrough, channel 'PJSIP/demo-alice-00000005' status is 'BUSY'

Any idea what im doing wrong?  Thanks :)



-- 
-- -- --
john at dev1ce.com
https://dev1ce.com/john.gpg