On Tue, Dec 22, 2015 at 7:26 AM, Marcos Prates <mar...@ultrawave.com.br>
wrote:

> Hi,
>
> I'm having a strange problem with Asterisk 13 i can't seem to find out
> whats causing it.
> After a Dial call from one SIP peer to another, if the calling side hangs
> up, DIALSTATUS is not set, but when the called side hangs up, it does.
> The strangest thing is when debugging SIP, it sends/receives the BYE
> signal normaly on both situations.
> I'm using DIALSTATUS on my accounting/billing scripts, so when this
> happens it break the routine.
>
> Can anyone shed some light into this for me? i'm running out of ideas here.
>
> Thanks.
>
> Marcos O.
>
>
Works for me. Given the following dialplan, which has a hardcoded Dial to
PJSIP endpoint 'alice':

exten => _XXXX,1,NoOp()
 same => n,Dial(PJSIP/alice,15)
 same => n,Hangup()

exten => h,1,NoOp()
 same => n,Log(NOTICE, ${DIALSTATUS})


Calling party (bob) hangs up first:

   -- Executing [1000@default:1] NoOp("PJSIP/bob-00000001", "") in new stack
    -- Executing [1000@default:2] Dial("PJSIP/bob-00000001",
"PJSIP/alice,15") in new stack
    -- Called PJSIP/alice
    -- PJSIP/alice-00000002 is ringing
    -- PJSIP/alice-00000002 answered PJSIP/bob-00000001
    -- Channel PJSIP/alice-00000002 joined 'simple_bridge' basic-bridge
<593321e8-d105-4075-94f3-20480cae3c45>
    -- Channel PJSIP/bob-00000001 joined 'simple_bridge' basic-bridge
<593321e8-d105-4075-94f3-20480cae3c45>
    -- Channel PJSIP/bob-00000001 left 'native_rtp' basic-bridge
<593321e8-d105-4075-94f3-20480cae3c45>
  == Spawn extension (default, 1000, 2) exited non-zero on
'PJSIP/bob-00000001'
    -- Channel PJSIP/alice-00000002 left 'native_rtp' basic-bridge
<593321e8-d105-4075-94f3-20480cae3c45>
    -- Executing [h@default:1] NoOp("PJSIP/bob-00000001", "") in new stack
    -- Executing [h@default:2] Log("PJSIP/bob-00000001", "NOTICE, ANSWER")
in new stack
[Dec 22 16:32:47] NOTICE[9668][C-00000001]: Ext. h:2 @ default:  ANSWER

Called party (alice) hangs up first:

*CLI>     -- Executing [1000@default:1] NoOp("PJSIP/bob-00000000", "") in
new stack
    -- Executing [1000@default:2] Dial("PJSIP/bob-00000000",
"PJSIP/alice,15") in new stack
    -- Called PJSIP/alice
    -- PJSIP/alice-00000001 is ringing
    -- PJSIP/alice-00000001 answered PJSIP/bob-00000000
    -- Channel PJSIP/alice-00000001 joined 'simple_bridge' basic-bridge
<0be9ceeb-014b-477c-b993-a4600ad7f9b0>
    -- Channel PJSIP/bob-00000000 joined 'simple_bridge' basic-bridge
<0be9ceeb-014b-477c-b993-a4600ad7f9b0>
    -- Channel PJSIP/alice-00000001 left 'native_rtp' basic-bridge
<0be9ceeb-014b-477c-b993-a4600ad7f9b0>
    -- Channel PJSIP/bob-00000000 left 'native_rtp' basic-bridge
<0be9ceeb-014b-477c-b993-a4600ad7f9b0>
  == Spawn extension (default, 1000, 2) exited non-zero on
'PJSIP/bob-00000000'
    -- Executing [h@default:1] NoOp("PJSIP/bob-00000000", "") in new stack
    -- Executing [h@default:2] Log("PJSIP/bob-00000000", "NOTICE, ANSWER")
in new stack
[Dec 22 16:34:17] NOTICE[9740][C-00000000]: Ext. h:2 @ default:  ANSWER


-- 
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
-- 
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
               http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Reply via email to