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 :) -- -- -- -- j...@dev1ce.com 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