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astuser at braingia.org
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 Posted: Thu Jun 26, 2008 8:11 pm Post subject: [asterisk-users] Asterisk, POTS and plain handsets |
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Hello,
I've spent a couple days searching and posted into the forum with no luck, apologies
to anyone who reads the Digium forums for the cross-post.
I'm having a problem with an asterisk set up where I have a TDM402B connected to a POTS
line. Also connected to the POTS line are plain telephones, non SIP, just plain
old telephones. When one of the normal handsets goes off-hook, asterisk
reads it as an incoming call and starts handling it accordingly, running through the
context for that channel as if an incoming call was detected. I'd like asterisk to act
like just another handset on the line or an answering machine and not do anything if a handset is used
on that same line.
It seems like maybe it's a voltage issue where asterisk or the zaptel module is sensing a
voltage change on the line and so is doing what it thinks it should do. I'd like to know
how to dumb it down or make it less sensitive to the changes, if indeed that is the cause.
I've tried various combinations of asterisk versions and zap module versions and various combinations of
phone lines to the card and, well, everything else that I can think of. So I'm hoping that
someone out there has used an asterisk set up like this and maybe encountered the same
things that I'm seeing.
I'd be happy to post any configs that someone might find relevant.
Steve |
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stotaro at totarotechn...
Guest
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| Posted: Thu Jun 26, 2008 9:35 pm Post subject: [asterisk-users] Asterisk, POTS and plain handsets |
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On Thu, Jun 26, 2008 at 9:11 PM, Steve <astuser at braingia.org> wrote:
| Quote: | Hello,
I've spent a couple days searching and posted into the forum with no luck, apologies
to anyone who reads the Digium forums for the cross-post.
I'm having a problem with an asterisk set up where I have a TDM402B connected to a POTS
line. Also connected to the POTS line are plain telephones, non SIP, just plain
old telephones. When one of the normal handsets goes off-hook, asterisk
reads it as an incoming call and starts handling it accordingly, running through the
context for that channel as if an incoming call was detected. I'd like asterisk to act
like just another handset on the line or an answering machine and not do anything if a handset is used
on that same line.
It seems like maybe it's a voltage issue where asterisk or the zaptel module is sensing a
voltage change on the line and so is doing what it thinks it should do. I'd like to know
how to dumb it down or make it less sensitive to the changes, if indeed that is the cause.
I've tried various combinations of asterisk versions and zap module versions and various combinations of
phone lines to the card and, well, everything else that I can think of. So I'm hoping that
someone out there has used an asterisk set up like this and maybe encountered the same
things that I'm seeing.
I'd be happy to post any configs that someone might find relevant.
Steve
|
Post the output from Asterisk's CLI. I think maybe your contexts are
overlapping or are the same. It should say something to the effect of
"Starting simple switch"
Check what context your FXO channels are in, something like
context=from-verizon and then check the context of your FXS (plain
telephones), they should be in a different context such as
context=to-phones.
Also, make sure you have immediate=no
Then check your dialplan and make sure those contexts do what you want
and you are not accidentally including a context where it should not
be.
Thanks,
Steve Totaro |
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astuser at braingia.org
Guest
|
| Posted: Fri Jun 27, 2008 8:49 am Post subject: [asterisk-users] Asterisk, POTS and plain handsets |
|
|
On Thu, Jun 26, 2008 at 10:35:13PM -0400, Steve Totaro wrote:
| Quote: | Post the output from Asterisk's CLI. I think maybe your contexts are
overlapping or are the same. It should say something to the effect of
"Starting simple switch"
|
When I take one of the plain phones off-hook, just lifting the
receiver, here's the CLI output. Zap3/1 is an FXO port which is
connected to the POTS line. Note that sometimes taking the phone
off-hook doesn't do it but only when the receiver is hung up or put back
on-hook does asterisk start to detect a ring.
The context home-incoming has one thing to do, dial a SIP phone for 20
seconds. On normal incoming calls this works great.
-- Starting simple switch on 'Zap/3-1'
[Jun 27 08:31:16] ERROR[8889]: callerid.c:540 callerid_feed: No start
bit found in fsk data.
-- Executing [s at home-incoming:1] Dial("Zap/3-1", "SIP/gxp20001,20")
in new stack
== Using SIP RTP CoS mark 5
-- Called gxp20001
-- SIP/gxp20001-081d50c0 is ringing
== Spawn extension (home-incoming, s, 1) exited non-zero on 'Zap/3-1'
-- Hungup 'Zap/3-1'
| Quote: | Check what context your FXO channels are in, something like
context=from-verizon and then check the context of your FXS (plain
telephones), they should be in a different context such as
context=to-phones.
|
The FXO ports come from the telco drop to the asterisk server. The card
doesn't have any FXS ports on it rather these phones are also connected
directly to the telco drop. So /something/ is happening where when one
of these phones is taken off-hook (or sometimes put back on-hook),
asterisk is catching it as being interesting event.
| Quote: | Also, make sure you have immediate=no
|
Didn't have it directly in the channel config, added it and rebooted and
no luck.
| Quote: | Then check your dialplan and make sure those contexts do what you want
and you are not accidentally including a context where it should not
be.
|
The contexts themselves do what I want but for some reason asterisk is
trying to handle an off-hook situation anywhere else on the copper as an
incoming call and handling it through the home-incoming context.
My hope is to be able to have asterisk act as an answering machine would
on this line, sharing the line with other regular phones. When a phone
call comes in, I'd like it to ring the two SIP phones on the network and
then if no one picks up after N seconds, answer the call. Obviously,
the regular phones would ring normally on an incoming call because
they're still connected to the telco drop.
Thanks for any further assistance.
Steve |
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stotaro at totarotechn...
Guest
|
| Posted: Fri Jun 27, 2008 10:25 am Post subject: [asterisk-users] Asterisk, POTS and plain handsets |
|
|
On Fri, Jun 27, 2008 at 9:49 AM, Steve <astuser at braingia.org> wrote:
| Quote: | On Thu, Jun 26, 2008 at 10:35:13PM -0400, Steve Totaro wrote:
| Quote: | Post the output from Asterisk's CLI. I think maybe your contexts are
overlapping or are the same. It should say something to the effect of
"Starting simple switch"
|
When I take one of the plain phones off-hook, just lifting the
receiver, here's the CLI output. Zap3/1 is an FXO port which is
connected to the POTS line. Note that sometimes taking the phone
off-hook doesn't do it but only when the receiver is hung up or put back
on-hook does asterisk start to detect a ring.
The context home-incoming has one thing to do, dial a SIP phone for 20
seconds. On normal incoming calls this works great.
-- Starting simple switch on 'Zap/3-1'
[Jun 27 08:31:16] ERROR[8889]: callerid.c:540 callerid_feed: No start
bit found in fsk data.
-- Executing [s at home-incoming:1] Dial("Zap/3-1", "SIP/gxp20001,20")
in new stack
== Using SIP RTP CoS mark 5
-- Called gxp20001
-- SIP/gxp20001-081d50c0 is ringing
== Spawn extension (home-incoming, s, 1) exited non-zero on 'Zap/3-1'
-- Hungup 'Zap/3-1'
| Quote: | Check what context your FXO channels are in, something like
context=from-verizon and then check the context of your FXS (plain
telephones), they should be in a different context such as
context=to-phones.
|
The FXO ports come from the telco drop to the asterisk server. The card
doesn't have any FXS ports on it rather these phones are also connected
directly to the telco drop. So /something/ is happening where when one
of these phones is taken off-hook (or sometimes put back on-hook),
asterisk is catching it as being interesting event.
| Quote: | Also, make sure you have immediate=no
|
Didn't have it directly in the channel config, added it and rebooted and
no luck.
| Quote: | Then check your dialplan and make sure those contexts do what you want
and you are not accidentally including a context where it should not
be.
|
The contexts themselves do what I want but for some reason asterisk is
trying to handle an off-hook situation anywhere else on the copper as an
incoming call and handling it through the home-incoming context.
My hope is to be able to have asterisk act as an answering machine would
on this line, sharing the line with other regular phones. When a phone
call comes in, I'd like it to ring the two SIP phones on the network and
then if no one picks up after N seconds, answer the call. Obviously,
the regular phones would ring normally on an incoming call because
they're still connected to the telco drop.
Thanks for any further assistance.
Steve
|
I tried the same thing about six or seven years ago. It isn't going
to work (or at least not that I am aware of). How many lines are you
talking about. If it is just a few, you could look at ATAs (they are
relatively cheap), a card with FXS ports, or SIP phone.
Thanks,
Steve Totaro |
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astuser at braingia.org
Guest
|
| Posted: Fri Jun 27, 2008 12:27 pm Post subject: [asterisk-users] Asterisk, POTS and plain handsets |
|
|
On Fri, Jun 27, 2008 at 11:25:04AM -0400, Steve Totaro wrote:
| Quote: | I tried the same thing about six or seven years ago. It isn't going
to work (or at least not that I am aware of). How many lines are you
talking about. If it is just a few, you could look at ATAs (they are
relatively cheap), a card with FXS ports, or SIP phone.
|
That's weird. This _was_ a working setup until a few weeks ago when an
FXO port on an existing TDM422B stopped working. I ordered a new card,
a TDM402B; I don't need the FXS ports since nothing was hooked into them
anyway. With the new card the previously working setup no longer works.
The new card seems to be a newer revision. The physical layout of the
modules is different on the card itself and I believe it uses a
different kernel module, wctdm24xxp now.
Is it possible that the new card is more sensitive and less configurable
and can't be adjusted to work like the old card? Or is it possible that
both modules on the new card are faulty?
Steve |
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astuser at braingia.org
Guest
|
| Posted: Fri Jun 27, 2008 3:59 pm Post subject: [asterisk-users] Asterisk, POTS and plain handsets |
|
|
Upon further digging, this is seems almost certainly related to the card
and kernel module being used for the card. I loaded up the old TDM22B
card and I'm no longer having the issue. I even took one of the FXO
(red) modules from the new card and put it into the old card and it
still works.
When the new card loads it uses kernel module wctdm24xxp for its kernel
module whereas the old card used wctdm for its module. So there must be
some flaw in the wctdm24xxp module -- or some flaw in wctdm was
fixed but it was really a _feature_ to me. 
I'm wondering if this isn't worth a bug report though -- or maybe
there's something wrong with the new card?
Steve
On Fri, Jun 27, 2008 at 08:49:48AM -0500, Steve wrote:
| Quote: | On Thu, Jun 26, 2008 at 10:35:13PM -0400, Steve Totaro wrote:
| Quote: | Post the output from Asterisk's CLI. I think maybe your contexts are
overlapping or are the same. It should say something to the effect of
"Starting simple switch"
|
When I take one of the plain phones off-hook, just lifting the
receiver, here's the CLI output. Zap3/1 is an FXO port which is
connected to the POTS line. Note that sometimes taking the phone
off-hook doesn't do it but only when the receiver is hung up or put back
on-hook does asterisk start to detect a ring.
The context home-incoming has one thing to do, dial a SIP phone for 20
seconds. On normal incoming calls this works great.
-- Starting simple switch on 'Zap/3-1'
[Jun 27 08:31:16] ERROR[8889]: callerid.c:540 callerid_feed: No start
bit found in fsk data.
-- Executing [s at home-incoming:1] Dial("Zap/3-1", "SIP/gxp20001,20")
in new stack
== Using SIP RTP CoS mark 5
-- Called gxp20001
-- SIP/gxp20001-081d50c0 is ringing
== Spawn extension (home-incoming, s, 1) exited non-zero on 'Zap/3-1'
-- Hungup 'Zap/3-1'
| Quote: | Check what context your FXO channels are in, something like
context=from-verizon and then check the context of your FXS (plain
telephones), they should be in a different context such as
context=to-phones.
|
The FXO ports come from the telco drop to the asterisk server. The card
doesn't have any FXS ports on it rather these phones are also connected
directly to the telco drop. So /something/ is happening where when one
of these phones is taken off-hook (or sometimes put back on-hook),
asterisk is catching it as being interesting event.
| Quote: | Also, make sure you have immediate=no
|
Didn't have it directly in the channel config, added it and rebooted and
no luck.
| Quote: | Then check your dialplan and make sure those contexts do what you want
and you are not accidentally including a context where it should not
be.
|
The contexts themselves do what I want but for some reason asterisk is
trying to handle an off-hook situation anywhere else on the copper as an
incoming call and handling it through the home-incoming context.
My hope is to be able to have asterisk act as an answering machine would
on this line, sharing the line with other regular phones. When a phone
call comes in, I'd like it to ring the two SIP phones on the network and
then if no one picks up after N seconds, answer the call. Obviously,
the regular phones would ring normally on an incoming call because
they're still connected to the telco drop.
Thanks for any further assistance.
Steve
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
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stotaro at totarotechn...
Guest
|
| Posted: Fri Jun 27, 2008 4:34 pm Post subject: [asterisk-users] Asterisk, POTS and plain handsets |
|
|
Ah, I am not sure why what you want to accomplish cannot be done or
why it shouldn't be. Like you said, it worked with wctdm.
When I tried it you had to modprobe wcfxo and wcfxs, back in the ole
days, and it was confusing which to load first........ The behavior
was the same as you describe with your new card. I never tried it
with wctdm.
Thanks,
Steve T
On Fri, Jun 27, 2008 at 4:59 PM, Steve <astuser at braingia.org> wrote:
| Quote: |
Upon further digging, this is seems almost certainly related to the card
and kernel module being used for the card. I loaded up the old TDM22B
card and I'm no longer having the issue. I even took one of the FXO
(red) modules from the new card and put it into the old card and it
still works.
When the new card loads it uses kernel module wctdm24xxp for its kernel
module whereas the old card used wctdm for its module. So there must be
some flaw in the wctdm24xxp module -- or some flaw in wctdm was
fixed but it was really a _feature_ to me.
I'm wondering if this isn't worth a bug report though -- or maybe
there's something wrong with the new card?
Steve
On Fri, Jun 27, 2008 at 08:49:48AM -0500, Steve wrote:
| Quote: | On Thu, Jun 26, 2008 at 10:35:13PM -0400, Steve Totaro wrote:
| Quote: | Post the output from Asterisk's CLI. I think maybe your contexts are
overlapping or are the same. It should say something to the effect of
"Starting simple switch"
|
When I take one of the plain phones off-hook, just lifting the
receiver, here's the CLI output. Zap3/1 is an FXO port which is
connected to the POTS line. Note that sometimes taking the phone
off-hook doesn't do it but only when the receiver is hung up or put back
on-hook does asterisk start to detect a ring.
The context home-incoming has one thing to do, dial a SIP phone for 20
seconds. On normal incoming calls this works great.
-- Starting simple switch on 'Zap/3-1'
[Jun 27 08:31:16] ERROR[8889]: callerid.c:540 callerid_feed: No start
bit found in fsk data.
-- Executing [s at home-incoming:1] Dial("Zap/3-1", "SIP/gxp20001,20")
in new stack
== Using SIP RTP CoS mark 5
-- Called gxp20001
-- SIP/gxp20001-081d50c0 is ringing
== Spawn extension (home-incoming, s, 1) exited non-zero on 'Zap/3-1'
-- Hungup 'Zap/3-1'
| Quote: | Check what context your FXO channels are in, something like
context=from-verizon and then check the context of your FXS (plain
telephones), they should be in a different context such as
context=to-phones.
|
The FXO ports come from the telco drop to the asterisk server. The card
doesn't have any FXS ports on it rather these phones are also connected
directly to the telco drop. So /something/ is happening where when one
of these phones is taken off-hook (or sometimes put back on-hook),
asterisk is catching it as being interesting event.
| Quote: | Also, make sure you have immediate=no
|
Didn't have it directly in the channel config, added it and rebooted and
no luck.
| Quote: | Then check your dialplan and make sure those contexts do what you want
and you are not accidentally including a context where it should not
be.
|
The contexts themselves do what I want but for some reason asterisk is
trying to handle an off-hook situation anywhere else on the copper as an
incoming call and handling it through the home-incoming context.
My hope is to be able to have asterisk act as an answering machine would
on this line, sharing the line with other regular phones. When a phone
call comes in, I'd like it to ring the two SIP phones on the network and
then if no one picks up after N seconds, answer the call. Obviously,
the regular phones would ring normally on an incoming call because
they're still connected to the telco drop.
Thanks for any further assistance.
Steve
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
|
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
|
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