asterisk users - Jan 2007 - Re: asterisk-users Digest, Vol 30, Issue 79

>
>
>  Hi,
>
>
>  I checked by changing to from-zaptel, but no luck yet. Pls guide me on
> this.
>
> Regards,
> vudura senadeera
>
>
> ------------------------------
> >
> > Message: 9
> > Date: Fri, 19 Jan 2007 16:47:18 -0000
> > From: "Robert Jenkins" < raj@jrw.co.uk>
> > Subject: RE: [asterisk-users] Integrating asterisk with Toshiba
> >        Astrata DK380
> > To: "'Asterisk Users Mailing List - Non-Commercial
Discussion'"
> >        <asterisk-users@lists.digium.com>
> > Message-ID: <005701c73be9$7c907850$2800a8c0@office>
> > Content-Type: text/plain; charset="us-ascii"
> >
> > Hi,
> >
> > your zapata.con has 'context=from-pstn'
> >
> > Try changing this to 'context=from-zaptel'
> >
> > Robert Jenkins.
> >
> >
> >
> > _____
> >
> > From: asterisk-users-bounces@lists.digium.com
> > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Vidura
> > Senadeera
> > Sent: 19 January 2007 15:19
> > To: asterisk-users@lists.digium.com
> > Cc: support@digium.com
> > Subject: [asterisk-users] Integrating asterisk with Toshiba Astrata
> > DK380
> >
> >
> >
> > Deat all,
> >
> > I am in middle of integrate Asterisk with Toshiba astrata legacy pbx.
> >
> > Following is my setup
> >
> > Asterisk <-> Digium TE110P <-> E1 card in toshiba pbx
<-> Toshiba PBX
> >
> > A =============================================> B
> > C <============================================ D
> >
> > Asterisk PBX and strata PBX connected using back to back E1 cross
cable.
> > Physicall connectivity is OK. The digium TE110p
> > LED state green. zttool also OK.
> >
> > Toshiba stata configured to make outbound call via E1 link with
pressing
> > 9
> > and then the out side number.
> >
> > I was able to make call from soft phone to analog extension at toshiba
> > pbx.
> > A==> B way as shown above. But when trying to dial from
> > Toshiba PBX analog extension to asterisk extension, by pressing 9 the
> > call
> > rejected.
> >
> > In the asterisk command prompt I'm having following error message.
> >
> > -- Extension '' in context 'from-pstn' from ''
does not
> > exist.  Rejecting
> > call on channel 0/31, span 1
> >
> > Is there any wrong in my setup. dial plan??, additional configuration
if
> > i
> > required to put please guide me.
> >
> > I will be greately appreciated your feedback on this regard.
> >
> > configuration details
> >
> > /etc/zaptel.conf
> > # Span 1: WCT1/0 "Digium Wildcard TE110P T1/E1 Card 0"
> >
> > span=1,1,0,ccs,hdb3,crc4
> > bchan=1-15,17-31
> > dchan=16
> >
> > /etc/asterisk/zapata.conf
> >
> > signalling=pri_net ; pri_cpe = PRI slave ; pri_net = PRI master
> > switchtype=euroisdn
> > ;switchtype=national
> > echocancel=yes ; You can set this to 32, 64, or 128, tweak to your
> > needs.
> > echocancelwhenbridged=yes
> > echotraining=400 ; Asterisk trains to the beginning of the call,
number
> > is
> > in milliseconds
> > callerid=asreceived
> > overlapdial=no
> > pridialplan=unknown
> > immediate=no
> > ;rxwink=300
> > callprogress=no
> > loadzone=au
> > context=from-pstn ; Points to the default context of your
> > extensions.conf
> > group=2
> > channel=>1-15
> > channel=>17-31 ;PRI/E1 link
> >
> >
> > [trunkgroups]
> > trunkgroup=>2,16
> > spanmap=1,2,1
> >
> >
> >
> > /etc/asterisk/extension.conf
> >
> > [from-zaptel]
> > exten => _X.,1,Set(DID=${EXTEN})
> > exten => _X.,n,Goto(s,1)
> > exten => s,1,NoOp(Entering from-zaptel with DID == ${DID})
> > ; If ($did == "") { $did = "s"; }
> > exten => s,n,Set(DID=${IF($["${DID}"=
""]?s:${DID})})
> > exten => s,n,NoOp(DID is now ${DID})
> > exten =>
s,n,GotoIf($["${CHANNEL:0:3}"="Zap"]?zapok:notzap)
> > exten => s,n(notzap),Goto(ext-did,${DID},1)
> > ; If there's no ext-did,s,1, that means there's not a no
did/no cid
> > route.
> > Hangup.
> > exten => s,n,Macro(hangup)
> > exten => s,n(zapok),NoOp(Is a Zaptel Channel)
> > exten => s,n,Set(CHAN=${CHANNEL:4})
> > exten => s,n,Set(CHAN=${CUT(CHAN,-,1)})
> > exten => s,n,Macro(from-zaptel-${CHAN},${DID},1)
> > ; If nothing there, then treat it as a DID
> > exten => s,n,NoOp(Returned from Macro from-zaptel-${CHAN})
> > exten => s,n,Goto(ext-did,${DID},1)
> >
> >
> >
> >
> > --
> > Thanks & Regards,
> > Vidura B. Senadeera.
> >
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> >
> > ------------------------------
> >
> > Message: 10
> > Date: Fri, 19 Jan 2007 11:46:57 -0500
> > From: "Chris Earle \(CBL\)" < cearle@cbltech.ca>
> > Subject: [asterisk-users] Disconnect Supervision UK / BT solution?
> > To: <asterisk-users@lists.digium.com >
> > Message-ID: <013b01c73be9$6ec87c40$6b01a8c0@Chrisdev>
> > Content-Type: text/plain;       charset="iso-8859-1"
> >
> > Hi all
> >
> > I'm using sangoma a200 cards in the UK and have the ongoing, often
noted
> >
> > problem of disconnect supervision with BT POTS lines.
> >
> > Just noticed this post on
> > http://www.voip-info.org/wiki/view/UK+Asterisk+Details
> > stating that potentially someone's got a solution :
> >
> > "TDM400P &amp; Not Detecting Hangups:
> >
> > Got a TDM400P installed and having problems with Asterisk not
detecting
> > hangups? Using BT? If so, contact BT and ask what the "Disconnect
Clear
> > Time" setting is for your phone line. Odds are it's probably
100.
> > Increasing
> > it to 800 fixed the issue for me.
> >
> > "Disconnect Clear Time" is BT's name for CPC. "
> >
> >
> > Does anyone have any thoughts/confirmation about this finally being a
> > viable
> > solution?  This disconnect supervision problem has plagued TDM and
> > Sangoma
> > cards for a long time!
> >
> > Comments appreciated before I get on the phone with BT
> >
> >
> > --
> > Chris Earle
> > System Solutions Specialist
> >
> >
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Hi,
 
what do you now get in the way of error messages?
 
 
Robert Jenkins.
 


  _____  

From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Vidura
Senadeera
Sent: 19 January 2007 23:03
To: asterisk-users@lists.digium.com
Cc: support@digium.com
Subject: [asterisk-users] Re: asterisk-users Digest, Vol 30, Issue 79




Hi,

 
 I checked by changing to from-zaptel, but no luck yet. Pls guide me on
this.
 
Regards,
vudura senadeera

 


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

Message: 9
Date: Fri, 19 Jan 2007 16:47:18 -0000
From: "Robert Jenkins" < raj@jrw.co.uk>
Subject: RE: [asterisk-users] Integrating asterisk with Toshiba
       Astrata DK380
To: "'Asterisk Users Mailing List - Non-Commercial
Discussion'"
       <asterisk-users@lists.digium.com>
Message-ID: <005701c73be9$7c907850$2800a8c0@office >
Content-Type: text/plain; charset="us-ascii" 

Hi,

your zapata.con has 'context=from-pstn'

Try changing this to 'context=from-zaptel'

Robert Jenkins.



_____

From: asterisk-users-bounces@lists.digium.com
[mailto:  <mailto:asterisk-users-bounces@lists.digium.com>
asterisk-users-bounces@lists.digium.com] On Behalf Of Vidura
Senadeera
Sent: 19 January 2007 15:19
To: asterisk-users@lists.digium.com
Cc: support@digium.com
Subject: [asterisk-users] Integrating asterisk with Toshiba Astrata DK380 



Deat all,

I am in middle of integrate Asterisk with Toshiba astrata legacy pbx.

Following is my setup

Asterisk <-> Digium TE110P <-> E1 card in toshiba pbx <->
Toshiba PBX

A =============================================> B
C <============================================ D

Asterisk PBX and strata PBX connected using back to back E1 cross cable.
Physicall connectivity is OK. The digium TE110p 
LED state green. zttool also OK.

Toshiba stata configured to make outbound call via E1 link with pressing 9
and then the out side number.

I was able to make call from soft phone to analog extension at toshiba pbx. 
A==> B way as shown above. But when trying to dial from
Toshiba PBX analog extension to asterisk extension, by pressing 9 the call
rejected.

In the asterisk command prompt I'm having following error message. 

-- Extension '' in context 'from-pstn' from '' does not
exist.  Rejecting
call on channel 0/31, span 1

Is there any wrong in my setup. dial plan??, additional configuration if i
required to put please guide me.

I will be greately appreciated your feedback on this regard.

configuration details

/etc/zaptel.conf
# Span 1: WCT1/0 "Digium Wildcard TE110P T1/E1 Card 0" 

span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16

/etc/asterisk/zapata.conf

signalling=pri_net ; pri_cpe = PRI slave ; pri_net = PRI master
switchtype=euroisdn
;switchtype=national
echocancel=yes ; You can set this to 32, 64, or 128, tweak to your needs. 
echocancelwhenbridged=yes
echotraining=400 ; Asterisk trains to the beginning of the call, number is
in milliseconds
callerid=asreceived
overlapdial=no
pridialplan=unknown
immediate=no
;rxwink=300 
callprogress=no
loadzone=au
context=from-pstn ; Points to the default context of your extensions.conf
group=2
channel=>1-15
channel=>17-31 ;PRI/E1 link


[trunkgroups]
trunkgroup=>2,16 
spanmap=1,2,1



/etc/asterisk/extension.conf

[from-zaptel]
exten => _X.,1,Set(DID=${EXTEN})
exten => _X.,n,Goto(s,1)
exten => s,1,NoOp(Entering from-zaptel with DID == ${DID})
; If ($did == "") { $did = "s"; }
exten => s,n,Set(DID=${IF($["${DID}"= ""]?s:${DID})})
exten => s,n,NoOp(DID is now ${DID})
exten =>
s,n,GotoIf($["${CHANNEL:0:3}"="Zap"]?zapok:notzap)
exten => s,n(notzap),Goto(ext-did,${DID},1)
; If there's no ext-did,s,1, that means there's not a no did/no cid
route.
Hangup.
exten => s,n,Macro(hangup)
exten => s,n(zapok),NoOp(Is a Zaptel Channel) 
exten => s,n,Set(CHAN=${CHANNEL:4})
exten => s,n,Set(CHAN=${CUT(CHAN,-,1)})
exten => s,n,Macro(from-zaptel-${CHAN},${DID},1)
; If nothing there, then treat it as a DID
exten => s,n,NoOp(Returned from Macro from-zaptel-${CHAN}) 
exten => s,n,Goto(ext-did,${DID},1)




--
Thanks & Regards,
Vidura B. Senadeera.

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

Message: 10
Date: Fri, 19 Jan 2007 11:46:57 -0500
From: "Chris Earle \(CBL\)" < cearle@cbltech.ca>
Subject: [asterisk-users] Disconnect Supervision UK / BT solution?
To: <  <mailto:asterisk-users@lists.digium.com>
asterisk-users@lists.digium.com >
Message-ID: <013b01c73be9$6ec87c40$6b01a8c0@Chrisdev>
Content-Type: text/plain;       charset="iso-8859-1"

Hi all

I'm using sangoma a200 cards in the UK and have the ongoing, often noted 
problem of disconnect supervision with BT POTS lines.

Just noticed this post on
http://www.voip-info.org/wiki/view/UK+Asterisk+Details
stating that potentially someone's got a solution :

"TDM400P &amp; Not Detecting Hangups:

Got a TDM400P installed and having problems with Asterisk not detecting 
hangups? Using BT? If so, contact BT and ask what the "Disconnect Clear 
Time" setting is for your phone line. Odds are it's probably 100.
Increasing
it to 800 fixed the issue for me.

"Disconnect Clear Time" is BT's name for CPC. " 


Does anyone have any thoughts/confirmation about this finally being a viable

solution?  This disconnect supervision problem has plagued TDM and Sangoma
cards for a long time!

Comments appreciated before I get on the phone with BT 


--
Chris Earle
System Solutions Specialist 



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