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Barry O'Donovan

2007-May-23 11:34 UTC

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

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@pbx:1] 
Set("Zap/48-1",
"RECORDFILE=/srv/recordings/live/1179858572.0") in new stack
    -- Executing [0868017669@pbx:2] 
MixMonitor("Zap/48-1",
"/srv/recordings/live/1179858572.0.wav|b") in new
stack
    -- Executing [0868017669@pbx:3] SetCallerPres("Zap/48-1",
"allowed") in
new stack
    -- Executing [0868017669@pbx:4] SetCallerID("Zap/48-1",
"5400") in new
stack
    -- Executing [0868017669@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/

C F

2007-May-24 20:27 UTC

head link

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

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.

On 5/23/07, Barry O'Donovan <barry+asterisk-users@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@pbx:1]
> Set("Zap/48-1",
"RECORDFILE=/srv/recordings/live/1179858572.0") in new stack
>     -- Executing [0868017669@pbx:2]
> MixMonitor("Zap/48-1",
"/srv/recordings/live/1179858572.0.wav|b") in new
> stack
>     -- Executing [0868017669@pbx:3] SetCallerPres("Zap/48-1",
"allowed") in
> new stack
>     -- Executing [0868017669@pbx:4] SetCallerID("Zap/48-1",
"5400") in new
> stack
>     -- Executing [0868017669@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/
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>

C F

2007-May-28 15:30 UTC

head link

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

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@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@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@pbx:1]
> > > Set("Zap/48-1",
"RECORDFILE=/srv/recordings/live/1179858572.0") in new
> > > stack -- Executing [0868017669@pbx:2]
> > > MixMonitor("Zap/48-1",
"/srv/recordings/live/1179858572.0.wav|b") in new
> > > stack
> > >     -- Executing [0868017669@pbx:3]
SetCallerPres("Zap/48-1", "allowed")
> > > in new stack
> > >     -- Executing [0868017669@pbx:4]
SetCallerID("Zap/48-1", "5400") in
> > > new stack
> > >     -- Executing [0868017669@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/
> > >
> > > _______________________________________________
> > > --Bandwidth and Colocation provided by Easynews.com --
> > >
> > > asterisk-users mailing list
> > > To UNSUBSCRIBE or update options visit:
> > >    http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> > _______________________________________________
> > --Bandwidth and Colocation provided by Easynews.com --
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >    http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> --
> Kind regards,
> Barry O'Donovan
> +353 86 801 7669
>
> http://www.barryodonovan.com/
>

Steve Totaro

2007-May-28 18:21 UTC

head link

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

According to your CLI output, the channel is being torn down.  

Is there a lag on the CLI between the inside channel and the outfacing
channel getting the hangup request?

Does this only happen on mobile phones?  I know if I call my cell and
hangup, it will continue to ring a couple or even a few more times.

Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
 
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-
> bounces@lists.digium.com] On Behalf Of C F
> Sent: Monday, May 28, 2007 6:30 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] TE205P, E1, Panasonic PBX and hang-up
issues> 
> 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@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@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@pbx:1]
> > > > Set("Zap/48-1",
"RECORDFILE=/srv/recordings/live/1179858572.0")
in> new
> > > > stack -- Executing [0868017669@pbx:2]
> > > > MixMonitor("Zap/48-1",
"/srv/recordings/live/1179858572.0.wav|b") in> new
> > > > stack
> > > >     -- Executing [0868017669@pbx:3]
SetCallerPres("Zap/48-1",
> "allowed")
> > > > in new stack
> > > >     -- Executing [0868017669@pbx:4]
SetCallerID("Zap/48-1",
"5400")> in
> > > > new stack
> > > >     -- Executing [0868017669@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/
> > > >
> > > > _______________________________________________
> > > > --Bandwidth and Colocation provided by Easynews.com --
> > > >
> > > > asterisk-users mailing list
> > > > To UNSUBSCRIBE or update options visit:
> > > >    http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> > > _______________________________________________
> > > --Bandwidth and Colocation provided by Easynews.com --
> > >
> > > asterisk-users mailing list
> > > To UNSUBSCRIBE or update options visit:
> > >    http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >
> > --
> > Kind regards,
> > Barry O'Donovan
> > +353 86 801 7669
> >
> > http://www.barryodonovan.com/
> >
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users

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