[asterisk-users] congested/busy on trunk?

[Joshua C. Colp]

On Sat, Mar 14, 2020 at 2:02 PM John Roman <john at 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 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 :)
>

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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