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@anveo_sip:1] Dial("PJSIP/demo-alice-00000005",
"PJSIP/blah@mytrunk") in new stack
-- Called PJSIP/blah@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 :)
--
-- -- --
[email protected]
https://dev1ce.com/john.gpg
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
