On Sat, Mar 14, 2020 at 2:02 PM John Roman <j...@dev1ce.com> wrote: > 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 :) >
The remote side eventually terminated the call. You'd need to grab a SIP trace (pjsip set logger on) and provide/look at the actual traffic to see what is going on. Based on your version string I also don't believe you are on Asterisk 17, you appear to be on master which will become Asterisk 18. -- Joshua C. Colp Asterisk Technical Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org
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