[Asterisk-Users] How to Order Disconnect Supervision from SBC using Adit 600?

2004-01-13 Thread Jonathan Moore
Can anyone help me with the term that SBC uses to refer to disconnect
supervision?  I have an Adit 600 channel bank which has helped improve the
disconnect detection time down to about 8 seconds. This is still causing some
issues in particular with call progress enabled in * we are having a few
disconnects while calls are in session (about 2 reported in first 5 days of use).

I have talked both to a local phone contractor and SBC directly and no one
seems to know what I am talking about. The phone contractor knew about the issue
with other phone systems in the area but didn't know there was a way to fix it
and SBC reps seem to never have heard of disconnect sup or calling party disconnect.

The * Handbook refers to loop start with call sup as kewlstart are
there other names for this protocol? One of the local contractors thought that
SBC automatically drops line voltage on remote hangup, in which case I need to
know what signalling to program into the ADIT 600's fxo channels. I also have
the option of going to groundstart signalling if this would fix the problem, but
it would cause some line downtime so it is not my preferred method.

The Adit 600 manual lists the following options for mapping FXO ports to the T1 DSO.

DPT = Dial Pulse Termination
EMDW = E&M Delayed Wink start
EMI = E&M Immediate start
EMICPD = E&M Immediate Start with Calling Party Disconnect
EMW = E&M Wink start
GS = Ground Start
GSRB = Ground Start with Reverse Battery
LS = Loop Start
LSCPD = Loop Start Calling Party Disconnect
LSRB = Loop Start with Reverse Battery
VoIP = Voice over IP (CMG only)

I believe I currently have the lines set to LSCPD which improved the hangup
situation, but hasn't completely fixed it.

I don't know if this has any relevance but I am also originating the clock
source from the * side with Wildcard T1 card.
-- 
Jonathan Moore
Director of Technology
Winfield Public Schools
Office 620.221.5100
Fax 620.221.0508





Visit Winfield Public Schools at http://usd465.com
-
This mail sent through IMP: http://horde.org/imp/
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] How to Order Disconnect Supervision from SBC using Adit 600?

2004-01-13 Thread Joel Maslak
On Tue, 13 Jan 2004, Jonathan Moore wrote:

> LSRB = Loop Start with Reverse Battery
> I believe I currently have the lines set to LSCPD which improved the hangup
> situation, but hasn't completely fixed it.

Try LSRB - it may work.

-- 
Joel
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] How to Order Disconnect Supervision from SBC using Adit 600?

2004-01-13 Thread James Sharp
> Can anyone help me with the term that SBC uses to refer to disconnect
> supervision?  I have an Adit 600 channel bank which has helped improve the
> disconnect detection time down to about 8 seconds. This is still causing
> some
> issues in particular with call progress enabled in * we are having a few
> disconnects while calls are in session (about 2 reported in first 5 days
> of use).
>
> I have talked both to a local phone contractor and SBC directly and no one
> seems to know what I am talking about. The phone contractor knew about the
> issue
> with other phone systems in the area but didn't know there was a way to
> fix it
> and SBC reps seem to never have heard of disconnect sup or calling party
> disconnect.

I've never seen a line from SBC that DIDN'T come with disconnect
supervision (some SBC line monkeys I know call it "battery drop
disconnect").


> The * Handbook refers to loop start with call sup as kewlstart are
> there other names for this protocol? One of the local contractors thought
> that
> SBC automatically drops line voltage on remote hangup, in which case I
> need to
> know what signalling to program into the ADIT 600's fxo channels. I also
> have
> the option of going to groundstart signalling if this would fix the
> problem, but
> it would cause some line downtime so it is not my preferred method.

Kewlstart is also an alias for battery drop disconnect.

> The Adit 600 manual lists the following options for mapping FXO ports to
> the T1 DSO.
>
> DPT = Dial Pulse Termination
> EMDW = E&M Delayed Wink start
> EMI = E&M Immediate start
> EMICPD = E&M Immediate Start with Calling Party Disconnect
> EMW = E&M Wink start
> GS = Ground Start
> GSRB = Ground Start with Reverse Battery
> LS = Loop Start
> LSCPD = Loop Start Calling Party Disconnect
> LSRB = Loop Start with Reverse Battery
> VoIP = Voice over IP (CMG only)
>
> I believe I currently have the lines set to LSCPD which improved the
> hangup
> situation, but hasn't completely fixed it.

That should be right.  If you're really interested in looking, take a
cheap voltmeter and put it across the line.  If everyone is on hook,
you'll see 48V.  If someone goes off hook, you'll see it drop to about 6V.
 If you see a quick drop to 0V when the far end hangs up, you've got
battery drop disconnect.

> I don't know if this has any relevance but I am also originating the clock
> source from the * side with Wildcard T1 card.

That's really the only way it'll work.  The channel bank can't generate
clocking.

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] How to Order Disconnect Supervision from SBC using Adit 600?

2004-01-13 Thread Jonathan Moore
I have a little more info on this. Following the suggestion of another post on
this topic I tracked down an analog phone with lighted buttons powered by the
phone connection. I directly connected the phone to one of my inbound lines and
called it with my cell phone. Picked up the analog phone, verified call
completion and then hung up my cell. I watched and waited for the lights to go
out. Sure enough they did, but it took 8 seconds from the time of the hangup.
After the flash more phone started emitting a dialtone sound. Is this correct? I
was under the impression the voltage drop would happen almost immediately.


-- 
Jonathan Moore
Director of Technology
Winfield Public Schools
Office 620.221.5100
Fax 620.221.0508


Quoting Joel Maslak <[EMAIL PROTECTED]>:

> On Tue, 13 Jan 2004, Jonathan Moore wrote:
> 
> > LSRB = Loop Start with Reverse Battery
> > I believe I currently have the lines set to LSCPD which improved the
> hangup
> > situation, but hasn't completely fixed it.
> 
> Try LSRB - it may work.
> 
> -- 
> Joel
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
> 


Visit Winfield Public Schools at http://usd465.com
-
This mail sent through IMP: http://horde.org/imp/
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


RE: [Asterisk-Users] How to Order Disconnect Supervision from SBC using Adit 600?

2004-01-13 Thread Scott Stingel
If you don't have a voltmeter to look at this, try just listening on the
line (using an analog telephone) when the far end hangs up.  You should hear
a distinct click-click on the line a second or two after they hang up.  If
you hear this, it's likely you are getting the required disconnect
supervision from the telco.  Note that many (most?) smaller private PBX's do
not drop loop current on an analog line when the far end disconnects - but
central office class switches usually do. 

It's not very scientific, but once you've heard one you can recognise it.

regards

Scott M. Stingel 
Emerging Voice Technology Inc.

Email:  [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>   
URL:www.evtmedia.com <http://www.evtmedia.com>   



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James Sharp
Sent: Tuesday, January 13, 2004 3:55 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] How to Order Disconnect Supervision from SBC
using Adit 600?


> Can anyone help me with the term that SBC uses to refer to disconnect
> supervision?  I have an Adit 600 channel bank which has helped improve the
> disconnect detection time down to about 8 seconds. This is still causing
> some
> issues in particular with call progress enabled in * we are having a few
> disconnects while calls are in session (about 2 reported in first 5 days
> of use).
>
> I have talked both to a local phone contractor and SBC directly and no one
> seems to know what I am talking about. The phone contractor knew about the
> issue
> with other phone systems in the area but didn't know there was a way to
> fix it
> and SBC reps seem to never have heard of disconnect sup or calling party
> disconnect.

I've never seen a line from SBC that DIDN'T come with disconnect
supervision (some SBC line monkeys I know call it "battery drop
disconnect").


> The * Handbook refers to loop start with call sup as kewlstart are
> there other names for this protocol? One of the local contractors thought
> that
> SBC automatically drops line voltage on remote hangup, in which case I
> need to
> know what signalling to program into the ADIT 600's fxo channels. I also
> have
> the option of going to groundstart signalling if this would fix the
> problem, but
> it would cause some line downtime so it is not my preferred method.

Kewlstart is also an alias for battery drop disconnect.

> The Adit 600 manual lists the following options for mapping FXO ports to
> the T1 DSO.
>
> DPT = Dial Pulse Termination
> EMDW = E&M Delayed Wink start
> EMI = E&M Immediate start
> EMICPD = E&M Immediate Start with Calling Party Disconnect
> EMW = E&M Wink start
> GS = Ground Start
> GSRB = Ground Start with Reverse Battery
> LS = Loop Start
> LSCPD = Loop Start Calling Party Disconnect
> LSRB = Loop Start with Reverse Battery
> VoIP = Voice over IP (CMG only)
>
> I believe I currently have the lines set to LSCPD which improved the
> hangup
> situation, but hasn't completely fixed it.

That should be right.  If you're really interested in looking, take a
cheap voltmeter and put it across the line.  If everyone is on hook,
you'll see 48V.  If someone goes off hook, you'll see it drop to about 6V.
 If you see a quick drop to 0V when the far end hangs up, you've got
battery drop disconnect.

> I don't know if this has any relevance but I am also originating the clock
> source from the * side with Wildcard T1 card.

That's really the only way it'll work.  The channel bank can't generate
clocking.

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] How to Order Disconnect Supervision from SBC using Adit 600?

2004-01-13 Thread TC

> > I don't know if this has any relevance but I am also originating the
clock
> > source from the * side with Wildcard T1 card.
>
> That's really the only way it'll work.  The channel bank can't generate
> clocking.
Is that a general statement of just confined to the ADIT 600
I thought any T1 interface could either receive timing or it can be a source
for timing
its just the interface may not be a high quality source eg stratum 4 not
stratum1


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] How to Order Disconnect Supervision from SBC using Adit 600?

2004-01-13 Thread Andrew Kohlsmith
> > I don't know if this has any relevance but I am also originating the
> > clock source from the * side with Wildcard T1 card.

> That's really the only way it'll work.  The channel bank can't generate
> clocking.

Are you sure about that?  I'm confident that I have both my Adit600 *and* my 
cheap-ass AB1 providing clocking *to* asterisk.  (not both at the same 
time, but when I was testing with both)

Regards,
Andrew
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] How to Order Disconnect Supervision from SBC using Adit 600?

2004-01-13 Thread Steven Critchfield
On Wed, 2004-01-14 at 11:44, TC wrote:
> > > I don't know if this has any relevance but I am also originating the
> clock
> > > source from the * side with Wildcard T1 card.
> >
> > That's really the only way it'll work.  The channel bank can't generate
> > clocking.
> Is that a general statement of just confined to the ADIT 600
> I thought any T1 interface could either receive timing or it can be a source
> for timing
> its just the interface may not be a high quality source eg stratum 4 not
> stratum1

I'm pretty sure most devices can produce timing, just that channel banks
tend to be setup to default to accepting timing. This would after all be
the normal config for most sites.

The ADIT I'm pretty sure can produce timing, as I think I had mine setup
that way once.
-- 
Steven Critchfield  <[EMAIL PROTECTED]>

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


RE: [Asterisk-Users] How to Order Disconnect Supervision from SBC using Adit 600?

2004-01-13 Thread Don Pobanz
On Wednesday, January 14, 2004 11:45 AM, TC [SMTP:[EMAIL PROTECTED] 
wrote:
>
> > > I don't know if this has any relevance but I am also originating
> > > the clock
> > > source from the * side with Wildcard T1 card.
> >
> > That's really the only way it'll work.  The channel bank can't
> > generate
> > clocking.
> Is that a general statement of just confined to the ADIT 600
> I thought any T1 interface could either receive timing or it can be a
> source for timing
> its just the interface may not be a high quality source eg stratum 4
> not stratum1
>

Any clock can be the source. Of course, you should select the highest 
available stratum clock and time everything else from this if possible. 

Another problem is if equipment at both ends of a T1 are trying to 
derive timing from the other end. This can cause an unstable clock and 
create synchronization problems.

Always, always for T1s, one end needs to provide timing and the other 
end derive timing. (The only exception is if the equipment on both ends 
are tied to stratum 1 clocks. However, I would guess that this does not 
apply to any of us on this list).

Don Pobanz
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] How to Order Disconnect Supervision from SBC using Adit 600?

2004-01-13 Thread Rich Adamson

> > > I don't know if this has any relevance but I am also originating the
> clock source from the * side with Wildcard T1 card.
> >
> > That's really the only way it'll work.  The channel bank can't generate
> > clocking.

be carefull with semantics

the transmit leg of the T1 leaving the channel bank is clocked by the 
channel bank, and therefore could be used by the device at the opposite
end for "sync". However, the clock within an end-node channel bank is 
(by design) syncing off the receive leg of the T1. Therefore, by design 
it is not an official "source" of clock sync. Words #?!%*

> Is that a general statement of just confined to the ADIT 600
> I thought any T1 interface could either receive timing or it can be a source
> for timing its just the interface may not be a high quality source 
> eg stratum 4 not stratum1

exactly.



___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] How to Order Disconnect Supervision from SBC using Adit 600?

2004-01-13 Thread James Sharp
> I have a little more info on this. Following the suggestion of another
> post on
> this topic I tracked down an analog phone with lighted buttons powered by
> the
> phone connection. I directly connected the phone to one of my inbound
> lines and
> called it with my cell phone. Picked up the analog phone, verified call
> completion and then hung up my cell. I watched and waited for the lights
> to go
> out. Sure enough they did, but it took 8 seconds from the time of the
> hangup.
> After the flash more phone started emitting a dialtone sound. Is this
> correct? I
> was under the impression the voltage drop would happen almost immediately.

Do you have another analog line that's on the same Central office as the
line in question?  The delay could be lag time betwee the time you hang up
your cell phone, the cell provider MTSO processes the hang up, passes it
on to their termination provider, who then passes it on to your
termination provider, who then passes it on to you.

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] How to Order Disconnect Supervision from SBC using Adit 600?

2004-01-13 Thread Jonathan Moore
On Tue, 13 Jan 2004, James Sharp wrote:

> > I have a little more info on this. Following the suggestion of another
> > post on
> > this topic I tracked down an analog phone with lighted buttons powered by
> > the
> > phone connection. I directly connected the phone to one of my inbound
> > lines and
> > called it with my cell phone. Picked up the analog phone, verified call
> > completion and then hung up my cell. I watched and waited for the lights
> > to go
> > out. Sure enough they did, but it took 8 seconds from the time of the
> > hangup.
> > After the flash more phone started emitting a dialtone sound. Is this
> > correct? I
> > was under the impression the voltage drop would happen almost immediately.
> 
> Do you have another analog line that's on the same Central office as the
> line in question?  The delay could be lag time betwee the time you hang up
> your cell phone, the cell provider MTSO processes the hang up, passes it
> on to their termination provider, who then passes it on to your
> termination provider, who then passes it on to you.

I will try that, but the results were consistent with what we are seeing
across the board. I will call from my pbx to my test line in the morning
and time the results.

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] How to Order Disconnect Supervision from SBC using Adit 600?

2004-01-14 Thread Jonathan Moore
Quoting Jonathan Moore <[EMAIL PROTECTED]>:

> On Tue, 13 Jan 2004, James Sharp wrote:
> 
> > > I have a little more info on this. Following the suggestion of another
> > > post on
> > > this topic I tracked down an analog phone with lighted buttons powered
> by
> > > the
> > > phone connection. I directly connected the phone to one of my inbound
> > > lines and
> > > called it with my cell phone. Picked up the analog phone, verified call
> > > completion and then hung up my cell. I watched and waited for the lights
> > > to go
> > > out. Sure enough they did, but it took 8 seconds from the time of the
> > > hangup.
> > > After the flash more phone started emitting a dialtone sound. Is this
> > > correct? I
> > > was under the impression the voltage drop would happen almost
> immediately.
> > 
> > Do you have another analog line that's on the same Central office as the
> > line in question?  The delay could be lag time betwee the time you hang up
> > your cell phone, the cell provider MTSO processes the hang up, passes it
> > on to their termination provider, who then passes it on to your
> > termination provider, who then passes it on to you.
> 
> I will try that, but the results were consistent with what we are seeing
> across the board. I will call from my pbx to my test line in the morning
> and time the results.
> 

I tried the test again today but using an analog line to call the test line
(plexar to plexar) and it was in the same range. I think my particular test call
was actually 11 seconds. So it doesn't seem to matter whether it is a cell call
or a land line call there is a significant delay before the voltage drops.

I also met with our SBC rep today and he brought a technical person with him.
The techie seemed to think the delay was longer than it should be. They are
going to check into it. They are also going to check to see if the delay for
ground start lines is lower if nothing can be done to speed up the loop start
signaling.



Visit Winfield Public Schools at http://usd465.com
-
This mail sent through IMP: http://horde.org/imp/
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users