[asterisk-users] TE205P, E1, Panasonic PBX and hang-up issues

[C F]

Yes that makes more sense. Now to the problem, please post your
zapata.conf as well as your zaptel.conf. Also if you don't mind
downloading the config file from the Panasonic TD1232 and email to me
off list so I can take a look at it and make sure the settings are ok
on the panasonic side.

Thank you

On 5/28/07, Barry O'Donovan <barry at opensolutions.ie> wrote:
> On Fri 25 May 2007, C F wrote:
> > Are you sure the panasonic is TVP 100? I have installed over 50
> > Panasonic systems in my life, and service many more, I have never
> > heard of that system, and a quick google shows it's just a VoiceMail
> > system and not a PBX.
>
> Thanks for the reply. Does D1232 Digital Super Hybrid System make more sense?
>
> Thanks,
> Barry
>
> >
> > On 5/23/07, Barry O'Donovan <barry+asterisk-users at opensolutions.ie> wrote:
> > > Hey folks,
> > >
> > > I have a Digium TE205P working as a man in the middle:
> > >
> > > PRI line -------- Asterisk/TE205P -------- PBX
> > >
> > > The PBX is a Panasonic KX - TVP 100.
> > >
> > > Everything is working great except for one little issue. Asterisk isn't
> > > hanging up the PRI B channel when the PBX channel is hung up.
> > >
> > > I don't want to overload you with information but please ask if more is
> > > needed. I suspect I'm really hoping someone who had a similar problem
> > > with just say "ah, I know what that is!".
> > >
> > > Versions in use for Zaptel, LibPRI and Asterisk are all the SVN 1.4
> > > branch.
> > >
> > > To replicate:
> > >
> > > 1. dial a mobile (say) from one of the PBX phones;
> > > 2. when you here a ring tone, hang up the PBX phone;
> > > 3. the mobile continues to ring.
> > >
> > > The verbose output is:
> > >
> > >     -- Accepting overlap call from '' to '<unspecified>' on channel 0/17,
> > > span 2
> > >     -- Starting simple switch on 'Zap/48-1'
> > >     -- Executing [0868017669 at pbx:1]
> > > Set("Zap/48-1", "RECORDFILE=/srv/recordings/live/1179858572.0") in new
> > > stack -- Executing [0868017669 at pbx:2]
> > > MixMonitor("Zap/48-1", "/srv/recordings/live/1179858572.0.wav|b") in new
> > > stack
> > >     -- Executing [0868017669 at pbx:3] SetCallerPres("Zap/48-1", "allowed")
> > > in new stack
> > >     -- Executing [0868017669 at pbx:4] SetCallerID("Zap/48-1", "5400") in
> > > new stack
> > >     -- Executing [0868017669 at pbx:5] Dial("Zap/48-1", "Zap/g0/0868017669")
> > > in new stack
> > >     -- Requested transfer capability: 0x00 - SPEECH
> > >     -- Called g0/0868017669
> > >   == Begin MixMonitor Recording Zap/48-1
> > >     -- Zap/1-1 is ringing
> > >     -- Channel 0/17, span 2 got hangup request, cause 16
> > >     -- Zap/1-1 answered Zap/48-1
> > >     -- Channel 0/1, span 1 got hangup request, cause 0
> > >     -- Hungup 'Zap/1-1'
> > >   == Spawn extension (pbx, 0868017669, 5) exited non-zero on 'Zap/48-1'
> > >   == End MixMonitor Recording Zap/48-1
> > >     -- Hungup 'Zap/48-1'
> > > asterisk1*CLI>
> > >
> > >
> > > Any suggestions or fixes that you might have from prior instances would
> > > be greatly appreciated.
> > >
> > > Thanks a million,
> > >
> > > Barry O'Donovan
> > > http://www.barryodonovan.com/
> > >
> > > _______________________________________________
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>
>
> --
> Kind regards,
> Barry O'Donovan
> +353 86 801 7669
>
> http://www.barryodonovan.com/
>