[Asterisk-Users] Problem with Hangups

Tue Aug 30 07:56:12 MST 2005

Hiz;

I post you the debug, for seeing if anyone can help me.

-------------------------------------------
Aug 30 12:53:36 VERBOSE[3562]:     -- Accepting voice call from
'800245' to '800275' on channel 0/1, span 4
Aug 30 12:53:36 DEBUG[3562]: Enabled echo cancellation on channel 10
Aug 30 12:53:36 VERBOSE[3562]:     -- Executing Dial("Zap/10-1",
"SIP/275|60|tTr") in new stack
Aug 30 12:53:36 DEBUG[3562]: SIMPLE DIAL (NO URL)
Aug 30 12:53:36 DEBUG[3562]: Setting NAT on RTP to 0
Aug 30 12:53:36 DEBUG[3562]: Setting NAT on VRTP to 0
Aug 30 12:53:36 DEBUG[3562]: Outgoing Call for 275
Aug 30 12:53:36 DEBUG[3562]: Call from user '275' is 1 out of 0
Aug 30 12:53:36 VERBOSE[3562]:     -- Called 275
Aug 30 12:53:36 DEBUG[3562]: (Provisional) Stopping retransmission
(but retaining packet) on
'5243a87e0ef2d2ba0afdaca845fcf98e at 192.168.2.71' Request 102: Found
Aug 30 12:53:36 DEBUG[3562]: (Provisional) Stopping retransmission
(but retaining packet) on
'5243a87e0ef2d2ba0afdaca845fcf98e at 192.168.2.71' Request 102: Found
Aug 30 12:53:36 VERBOSE[3562]:     -- SIP/275-ca0f is ringing
Aug 30 12:53:42 VERBOSE[3562]:     -- Channel 0/1, span 4 got hangup
Aug 30 12:53:42 DEBUG[3562]: update_user_counter(275) - decrement outUse counter
Aug 30 12:53:42 DEBUG[3562]: Exiting with DIALSTATUS=CANCEL.
Aug 30 12:53:42 VERBOSE[3562]:   == Spawn extension (default, 800275,
1) exited non-zero on 'Zap/10-1'
Aug 30 12:53:42 DEBUG[3562]: Acked pending invite 102
Aug 30 12:53:42 DEBUG[3562]: Stopping retransmission on
'5243a87e0ef2d2ba0afdaca845fcf98e at 192.168.2.71' of Request 102: Found
Aug 30 12:53:42 DEBUG[3562]: Stopping retransmission on
'5243a87e0ef2d2ba0afdaca845fcf98e at 192.168.2.71' of Request 102: Found
Aug 30 12:53:42 DEBUG[3562]: Set option AUDIO MODE, value: ON(1) on Zap/10-1
Aug 30 12:53:42 DEBUG[3562]: Hangup: channel: 10 index = 0, normal =
26, callwait = -1, thirdcall = -1
Aug 30 12:53:42 DEBUG[3562]: Not yet hungup...  Calling hangup once
with icause, and clearing call
Aug 30 12:53:42 DEBUG[3562]: disabled echo cancellation on channel 10
Aug 30 12:53:42 DEBUG[3562]: Set option TDD MODE, value: OFF(0) on Zap/10-1
Aug 30 12:53:42 DEBUG[3562]: Updated conferencing on 10, with 0 conference users
Aug 30 12:53:42 DEBUG[3562]: Set option AUDIO MODE, value: OFF(0) on Zap/10-1
Aug 30 12:53:42 DEBUG[3562]: disabled echo cancellation on channel 10
Aug 30 12:53:42 VERBOSE[3562]:     -- Hungup 'Zap/10-1'

---------------------------------------------------------------------

Thanks you in advanced.

2005/8/30, Rich Adamson <radamson at routers.com>:
> There's not enough info in the posts below to help. The issue "sounds"
> like a problem with "disconnect supervision" (or whatever you want to
> call it).
> 
> In the case where a sip phone (or other asterisk phone) answers the
> call and then hangs up, the zap channel is being hung up properly due
> to the sip phone hangup. (That has nothing to do with disconnect
> supervision coming from the central office.)
> 
> When the call isn't answered by any phone, the zap channel depends on
> disconnect supervision coming from the central office. In the US, that
> signal is either: a) the dropping of tip/ring voltage to zero (as in
> open circuit), or, b) reversal of the tip/ring voltage. In all cases
> that I've seen in recent years, the open circuit for about 400 milliseconds
> is the most common in the US.
> 
> So, to diagnose the problem properly, place a voltmeter across the tip &
> ring pstn leads and watch for the open circuit when the pstn caller hangs
> up. If you "see" that open circuit, then you likely have either a bug in
> stable (which I don't use and can't comment on), or, your dialplan is
> arranged to do something odd and not recognize the hangup.
> 
> If you want help on this, then give us a clue as to whether you can or
> can't see the disconnect supervision, a copy/paste of the section of
> zapata.conf that pertains to the zap channel, some indication what the
> zap channel really is (eg, TDM, x100p, channel bank), CLI output when
> the hangup occurs, and copy/paste of the extensions.conf that is handling
> the inbound call.
> 
> ------------------------
> > Have you resolved the problem, I find the same problem
> >
> > Thanks
> >
> > 2005/8/23, Don Brearley <donbrearley at hibbing.edu>:
> > >
> > > Ohhhh come on now!   Nothing?  Not even a "No idea! Good Luck!" or anything?
> > >
> > > Weak :)
> > >
> > > Just kidding.  Thanks just the same.
> > >
> > > - Don
> > >
> > > >>> donbrearley at hibbing.edu 8/22/2005 11:09 AM >>>
> > >
> > > Hello,
> > >
> > > I am having an issue with hangups being handled within Asterisk.  Right now, when an
> inbound call hits
> > > the Asterisk box, Asterisk picks up the call just fine.
> > >
> > > When the caller enters an extension to call, the Asterisk dials out on Zap/3 and rings the
> extension with
> > > no problem.   If the extension is answered, there is no problem.
> > >
> > > If the caller hang's up before the phone is answered, by either a person, or by voicemail,
> the hangup
> > > is not detected, and the call continues as though the caller was still on the line.   It
> will continue to ring
> > > the extension until voicemail is picked up, and then is finally hungup after all of the
> timeouts.  If I dont
> > > have the extension setup to forward to the voicemail, it will just ring forever unless I
> force the hangup.
> > >
> > > I am using kewlstart signalling, and have busydetect=yes in my zapata.conf file.
> > >
> > > I am in the USA, and my indications.conf file seems to be in order.
> > >
> > > I am running Asterisk 1.0.9 with the zaptel-freebsd 0.10 ZapTel driver.
> > >
> > > Thank you for any insight into this issue!
> > >
> > > - Don Brearley
> > >  HCC Computer Services
> > >
> > > _______________________________________________
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> > > Asterisk-Users at lists.digium.com
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
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> > >
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> > >
> >
> >
> > --
> >       Jose Miguel Aracil
> >  Dpto. Seguridad y Redes
> >
> >        HARD I SOFT, S.A
> > Andorra La Vella - Andorra
> > _______________________________________________
> > --Bandwidth and Colocation sponsored by Easynews.com --
> >
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> 
> ---------------End of Original Message-----------------
> 
> 
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-- 
      Jose Miguel Aracil
 Dpto. Seguridad y Redes

       HARD I SOFT, S.A
Andorra La Vella - Andorra