Bug#842878: Asterisk crashes with pjproject 2.5.5

[Gedalya]

On 11/09/2016 04:51 AM, Bernhard Schmidt wrote:

> I have now uploaded Asterisk 13.12.1~dfsg-1 into sid, should be built
> and available within a few hours. This works well for me with pjproject
> from sid. Maybe it really just needed a recompile, but you had already
> tested that in the beginning. Would it be possible that something went
> wrong there?
I don't see what or how. I definitely rebuilt against the new pjproject.
I'm getting the same problems now, with the new packages in the debian archive.

Note that I can make outbound calls just fine. The problem is perhaps triggered by something specific in my inward dialing plan.

This is with the new packages:

    -- Executing *****desk-out:1] Goto("PJSIP/*****-00000000", "desk-out,*****,1") in new stack
    -- Goto (desk-out,*****,1)
    -- Executing [*****@desk-out:1] Set("PJSIP/*****-00000000", "CALLERID(num)=*****") in new stack
    -- Executing [*****@desk-out:2] Goto("PJSIP/*****-00000000", "out-vitel,*****,1") in new stack
    -- Goto (out-vitel,*****,1)
    -- Executing [*****@out-vitel:1] MixMonitor("PJSIP/*****-00000000", "/var/local/callrec/2016-11/1478733136.0.wav") in new stack
  == Begin MixMonitor Recording PJSIP/*****-00000000
    -- Executing [*****@out-vitel:2] Set("PJSIP/*****-00000000", "CDR(peername)=*****") in new stack
    -- Executing [*****@out-vitel:3] Dial("PJSIP/*****-00000000", "PJSIP/*****@trunk-didlogic") in new stack
    -- Called PJSIP/*****@trunk-didlogic
    -- Executing [*****@from-external:1] Goto("PJSIP/trunk-vitelity-in-00000002", "internal,gedalya-all,1")
    -- Goto (internal,gedalya-all,1)
    -- Executing [gedalya-all at internal:1] Set("PJSIP/trunk-vitelity-in-00000002", "CDR(peername)=trunk-vitelity-in") in new stack
    -- Executing [gedalya-all at internal:2] MixMonitor("PJSIP/trunk-vitelity-in-00000002", "/var/local/callrec/2016-11/1478733137.2.wav") in new stack
  == Begin MixMonitor Recording PJSIP/trunk-vitelity-in-00000002
    -- Executing [gedalya-all at internal:3] Dial("PJSIP/trunk-vitelity-in-00000002", "PJSIP/*****&PJSIP/*****&PJSIP/*****&PJSIP/*****&PJSIP/*****,30") in new stack
asterisk: ../src/pjsip/sip_auth_client.c:507: pjsip_auth_clt_deinit: Assertion `sess && sess->endpt' failed.
Aborted

This is me calling myself via my outbound trunk and coming back in via my inbound trunk.
The problem comes up only on the inbound calls, at the point where I dial them to the phones and other devices.
If I just call e.g. my mobile phone it works fine, I can complete the call and asterisk continues fine.

I'll try patching pjproject now and see how that goes.