asterisk users - Sep 2003 - Hangups after voicemail

Hi,

Try as I might, I can't get hangups detected on a Zap channel with loop
start
lines. So, after someone leaves a voicemail and then hangs up, Asterisk 
doesn't know it, exits VoicemailMain2, and loops back to the corporate 
greeting, tying up the line even though the outside caller has hung up. 

Therefore, I've added the following hideous hack - er, code - to
voicemail2.c.
It starts right after the call to play_and_record() in leave_voicemail().

if (res != '#' && chan != NULL &&
!strncmp(chan->name, "Zap", 3)) {
	/* Hang up the Zap channel only */
	ast_softhangup(chan, AST_SOFTHANGUP_EXPLICIT);
}

Obviously, it hangs up the channel after the voicemail has been recorded, if 
the # key wasn't pressed, if the channel still exists, and if it's a Zap
channel. I couldn't see a way to do this with AGI.

Question: is this safe? I used a soft hangup because the channel is controlled 
by another thread. I also modified channel.c so that ast_channel_free() sets 
chan to NULL after it's freed, just in case. Is there anything else I should
be aware of? The code seems to work in my testing, resulting in a proper 
hangup right after the voicemail has been recorded. I'm not up on my
Asterisk
internals, so I'm not totally confident about this.

Thanks,

Chris

Have you tried using kewlstart instead?  Your loopstart lines might be
configured for kewlstart (forward disconnect supervision).

-wade
> -----Original Message-----
> From: Christian Hecimovic [mailto:checimovic@qworks.ca]
> Sent: Tuesday, September 16, 2003 7:48 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Hangups after voicemail
> 
> Hi,
> 
> Try as I might, I can't get hangups detected on a Zap channel with
loop
> start
> lines. So, after someone leaves a voicemail and then hangs up,
Asterisk
> doesn't know it, exits VoicemailMain2, and loops back to the corporate
> greeting, tying up the line even though the outside caller has hung
up.
> 
> Therefore, I've added the following hideous hack - er, code - to
> voicemail2.c.
> It starts right after the call to play_and_record() in
leave_voicemail().
> 
> if (res != '#' && chan != NULL &&
!strncmp(chan->name, "Zap", 3)) {
> 	/* Hang up the Zap channel only */
> 	ast_softhangup(chan, AST_SOFTHANGUP_EXPLICIT);
> }
> 
> Obviously, it hangs up the channel after the voicemail has been
recorded,
> if
> the # key wasn't pressed, if the channel still exists, and if it's
a
Zap
> channel. I couldn't see a way to do this with AGI.
> 
> Question: is this safe? I used a soft hangup because the channel is
> controlled
> by another thread. I also modified channel.c so that
ast_channel_free()
> sets
> chan to NULL after it's freed, just in case. Is there anything else I
> should
> be aware of? The code seems to work in my testing, resulting in a
proper
> hangup right after the voicemail has been recorded. I'm not up on my
> Asterisk
> internals, so I'm not totally confident about this.
> 
> Thanks,
> 
> Chris
> 
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users

> Have you tried using kewlstart instead?  Your loopstart
> lines might be configured for kewlstart (forward
> disconnect supervision).
I found there to be quite a delay there, around 20 seconds between the
calling party clearing down and the telco giving me loop current drop. Is
that something you could ask the telco to change? In the UK you get the loop
current drop immediately, followed by "Do-dah-dit The other person has
cleared.." announcement.

Paul

Do you have silence in the channel when the remote user hangs up or busy
tone ?

If you have silence you can use maxsilence=x_seconds in voicemail.conf
with
Voicemail2 application and that will make sure the calls are hanged up
after x_seconds of silence in the channel.

If you have busy tone then use the busydetect=yes in zapata.conf.
You can also limit the length of the voicemail message with
maxmessage=x_seconds in the voicemail.conf

regards
Martin

On Tue, 16 Sep 2003, Christian Hecimovic wrote:
> Hi,
>
> Try as I might, I can't get hangups detected on a Zap channel with loop
start
> lines. So, after someone leaves a voicemail and then hangs up, Asterisk
> doesn't know it, exits VoicemailMain2, and loops back to the corporate
> greeting, tying up the line even though the outside caller has hung up.
>
> Therefore, I've added the following hideous hack - er, code - to
voicemail2.c.
> It starts right after the call to play_and_record() in leave_voicemail().
>
> if (res != '#' && chan != NULL &&
!strncmp(chan->name, "Zap", 3)) {
> 	/* Hang up the Zap channel only */
> 	ast_softhangup(chan, AST_SOFTHANGUP_EXPLICIT);
> }
>
> Obviously, it hangs up the channel after the voicemail has been recorded,
if
> the # key wasn't pressed, if the channel still exists, and if it's
a Zap
> channel. I couldn't see a way to do this with AGI.
>
> Question: is this safe? I used a soft hangup because the channel is
controlled
> by another thread. I also modified channel.c so that ast_channel_free()
sets
> chan to NULL after it's freed, just in case. Is there anything else I
should
> be aware of? The code seems to work in my testing, resulting in a proper
> hangup right after the voicemail has been recorded. I'm not up on my
Asterisk
> internals, so I'm not totally confident about this.
>
> Thanks,
>
> Chris
>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>

Silencethreshold of 256 sounds a bit high...

You can also add a timeout extension to just hangup the line:

exten => t,1,Hangup

Without using Kewlstart, there isn't anyway for Asterisk to know that
the line has been disconnected, so you'll have to use the timeouts.

-wade

> -----Original Message-----
> From: Christian Hecimovic [mailto:checimovic@qworks.ca]
> Sent: Wednesday, September 17, 2003 11:59 AM
> To: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] Hangups after voicemail
> 
> Hi Martin,
> 
> Yes, silence detection in voicemail is working. I am using Voicemail2
with
> the
> silencethreshold set to 256. However, the line doesn't hang up after
the
> silence is detected; instead, Voicemail2 exits after recording the
> voicemail
> correctly, and Asterisk loops back into the main menu as if the # key
was
> pressed because the channel is still alive. Then it times out after 15
> seconds, as you can see below.
> 
> From extensions.conf:
> 
> [incoming]
> exten => s,1,Answer
> exten => s,2,DigitTimeout,5
> exten => s,3,ResponseTimeout,10
> exten => s,4,BackGround(corp_greeting)
> include => locals
> include => errors
> 
> The locals context consists of macros which look like this:
> 
> exten => s,1,Playback(transfer,skip)
> exten => s,2,Dial(${ARG2},20)
> exten => s,3,Voicemail2(u${ARG1})
> exten => s,4,Goto(incoming,s,1)
> exten => s,103,Voicemail2(b${ARG1})
> exten => s,104,Goto(incoming,s,1)
> 
> So after a voicemail is left, there is a Goto back into the incoming
> context.
> It all works great, except for when the line gets tied up by the
> DigitTimeout
> and ResponseTimeout bits when hangups aren't detected.
> 
> I've tried using BUSYDETECT_MARTIN with busydetect=yes and it
didn't
work.
> The
> channel stays up after the outside caller hangs up.
> 
> Since all of our inside phones are SIP lines, there is no problem
> detecting
> hangups when a voice conversation is taking place, since Asterisk
> obviously
> detects SIP hangups correctly whether it's SIP to SIP or SIP to
outside
> line.
> The problem is really only when outside callers leave voicemail.
> 
> Thanks,
> 
> Chris
> 
> On Wednesday 17 September 2003 08:09, Martin Pycko wrote:
> > Do you have silence in the channel when the remote user hangs up or
busy
> > tone ?
> >
> > If you have silence you can use maxsilence=x_seconds in
voicemail.conf
> > with
> > Voicemail2 application and that will make sure the calls are hanged
up
> > after x_seconds of silence in the channel.
> >
> > If you have busy tone then use the busydetect=yes in zapata.conf.
> > You can also limit the length of the voicemail message with
> > maxmessage=x_seconds in the voicemail.conf
> >
> > regards
> > Martin
> >
> > On Tue, 16 Sep 2003, Christian Hecimovic wrote:
> > > Hi,
> > >
> > > Try as I might, I can't get hangups detected on a Zap channel
with
> loop
> > > start lines. So, after someone leaves a voicemail and then hangs
up,
> > > Asterisk doesn't know it, exits VoicemailMain2, and loops
back to
the
> > > corporate greeting, tying up the line even though the outside
caller
> has
> > > hung up.
> > >
> > > Therefore, I've added the following hideous hack - er, code -
to
> > > voicemail2.c. It starts right after the call to play_and_record()
in
> > > leave_voicemail().
> > >
> > > if (res != '#' && chan != NULL &&
!strncmp(chan->name, "Zap", 3))
{
> > > 	/* Hang up the Zap channel only */
> > > 	ast_softhangup(chan, AST_SOFTHANGUP_EXPLICIT);
> > > }
> > >
> > > Obviously, it hangs up the channel after the voicemail has been
> recorded,
> > > if the # key wasn't pressed, if the channel still exists, and
if
it's
> a
> > > Zap channel. I couldn't see a way to do this with AGI.
> > >
> > > Question: is this safe? I used a soft hangup because the channel
is
> > > controlled by another thread. I also modified channel.c so that
> > > ast_channel_free() sets chan to NULL after it's freed, just
in
case.
> Is
> > > there anything else I should be aware of? The code seems to work
in my
> > > testing, resulting in a proper hangup right after the voicemail
has
> been
> > > recorded. I'm not up on my Asterisk internals, so I'm not
totally
> > > confident about this.
> > >
> > > Thanks,
> > >
> > > Chris
> > >
> > >
> > > _______________________________________________
> > > Asterisk-Users mailing list
> > > Asterisk-Users@lists.digium.com
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users

Then it sounds like your Telco isn't giving you kewlstart signaling.
This is by far the most reliable method of telling asterisk that the
line has been disconnected.  Trying asking your Telco if they can supply
you with Kewlstart or Forward Disconnect Supervision on your line.

Basically, all this does is momentarily reverse the polarity on the line
to indicate that the line has been disconnected.  The Zaptel FXO devices
detect this condition to indicate to Asterisk that the line has been
disconnected.

-wade
> -----Original Message-----
> From: Christian Hecimovic [mailto:checimovic@qworks.ca]
> Sent: Wednesday, September 17, 2003 12:01 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] Hangups after voicemail
> 
> Hi Wade,
> 
> Yes, my zapata.conf looks like this:
> 
> group = 1
> context = incoming
> signalling = fxs_ks
> echocancel = yes
> echocancelwhenbridged = yes
> channel => 1-2
> 
> So they are configured as kewlstart.
> 
> Thanks,
> 
> Chris
> 
> On Tuesday 16 September 2003 16:53, Wade J. Weppler wrote:
> > Have you tried using kewlstart instead?  Your loopstart lines might
be
> > configured for kewlstart (forward disconnect supervision).
> >
> > -wade
> >
> > > -----Original Message-----
> > > From: Christian Hecimovic [mailto:checimovic@qworks.ca]
> > > Sent: Tuesday, September 16, 2003 7:48 PM
> > > To: asterisk-users@lists.digium.com
> > > Subject: [Asterisk-Users] Hangups after voicemail
> > >
> > > Hi,
> > >
> > > Try as I might, I can't get hangups detected on a Zap channel
with
> >
> > loop
> >
> > > start
> > > lines. So, after someone leaves a voicemail and then hangs up,
> >
> > Asterisk
> >
> > > doesn't know it, exits VoicemailMain2, and loops back to the
corporate
> > > greeting, tying up the line even though the outside caller has
hung
> >
> > up.
> >
> > > Therefore, I've added the following hideous hack - er, code -
to
> > > voicemail2.c.
> > > It starts right after the call to play_and_record() in
> >
> > leave_voicemail().
> >
> > > if (res != '#' && chan != NULL &&
!strncmp(chan->name, "Zap", 3))
{
> > > 	/* Hang up the Zap channel only */
> > > 	ast_softhangup(chan, AST_SOFTHANGUP_EXPLICIT);
> > > }
> > >
> > > Obviously, it hangs up the channel after the voicemail has been
> >
> > recorded,
> >
> > > if
> > > the # key wasn't pressed, if the channel still exists, and if
it's
a
> >
> > Zap
> >
> > > channel. I couldn't see a way to do this with AGI.
> > >
> > > Question: is this safe? I used a soft hangup because the channel
is
> > > controlled
> > > by another thread. I also modified channel.c so that
> >
> > ast_channel_free()
> >
> > > sets
> > > chan to NULL after it's freed, just in case. Is there
anything
else I
> > > should
> > > be aware of? The code seems to work in my testing, resulting in a
> >
> > proper
> >
> > > hangup right after the voicemail has been recorded. I'm not
up on
my
> > > Asterisk
> > > internals, so I'm not totally confident about this.
> > >
> > > Thanks,
> > >
> > > Chris
> > >
> > >
> > > _______________________________________________
> > > Asterisk-Users mailing list
> > > Asterisk-Users@lists.digium.com
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users

Since I see so many questions about this, and could not find a concise 
answer when I was looking for the same thing...

Here is an example showing how to configure communications between
Asterisk and a Cisco 2600 router with an FXO card in it.

http://www.tape.net/~gerry/asterisk/cisco26x0.html

Comments and suggestions are welcome.

You can still use timeouts.  The responsetimeout would only be active in
the main menu.  If they don't respond within the timeout, then hangup on
them.

The alternative is to loop the menu a set number of times before hanging
up.  This would require some logic.

-wade
> -----Original Message-----
> From: Christian Hecimovic [mailto:checimovic@qworks.ca]
> Sent: Wednesday, September 17, 2003 12:41 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] Hangups after voicemail
> 
> Yes, I had to set it high, otherwise it didn't work right...I'll
fiddle
> with
> it a bit.
> 
> Timeouts aren't really an option, because if the caller presses #
after
> leaving a voicemail then they should be popped back into the main
menu. If
> I
> could check DTMF signals from extensions.conf, then this would indeed
> work.
> Basically, the logic after Voicemail2 exits should be
> 
> if (last key was not # and the channel is still alive)
>     hangup
> else
>     goto the main menu
> 
> The only way I could do this was by modifying voicemail2.c. So, back
to my
> main question: are there any problems with this? I'm most concerned
about
> memory issues. Should I be freeing something first, making another
cleanup
> function call, etc.?
> 
> On Wednesday 17 September 2003 09:08, Wade J. Weppler wrote:
> > Silencethreshold of 256 sounds a bit high...
> >
> > You can also add a timeout extension to just hangup the line:
> >
> > exten => t,1,Hangup
> >
> > Without using Kewlstart, there isn't anyway for Asterisk to know
that
> > the line has been disconnected, so you'll have to use the
timeouts.
> >
> > -wade
> >
> > > -----Original Message-----
> > > From: Christian Hecimovic [mailto:checimovic@qworks.ca]
> > > Sent: Wednesday, September 17, 2003 11:59 AM
> > > To: asterisk-users@lists.digium.com
> > > Subject: Re: [Asterisk-Users] Hangups after voicemail
> > >
> > > Hi Martin,
> > >
> > > Yes, silence detection in voicemail is working. I am using
Voicemail2
> >
> > with
> >
> > > the
> > > silencethreshold set to 256. However, the line doesn't hang
up
after
> >
> > the
> >
> > > silence is detected; instead, Voicemail2 exits after recording
the
> > > voicemail
> > > correctly, and Asterisk loops back into the main menu as if the #
key
> >
> > was
> >
> > > pressed because the channel is still alive. Then it times out
after 15
> > > seconds, as you can see below.
> > >
> > > From extensions.conf:
> > >
> > > [incoming]
> > > exten => s,1,Answer
> > > exten => s,2,DigitTimeout,5
> > > exten => s,3,ResponseTimeout,10
> > > exten => s,4,BackGround(corp_greeting)
> > > include => locals
> > > include => errors
> > >
> > > The locals context consists of macros which look like this:
> > >
> > > exten => s,1,Playback(transfer,skip)
> > > exten => s,2,Dial(${ARG2},20)
> > > exten => s,3,Voicemail2(u${ARG1})
> > > exten => s,4,Goto(incoming,s,1)
> > > exten => s,103,Voicemail2(b${ARG1})
> > > exten => s,104,Goto(incoming,s,1)
> > >
> > > So after a voicemail is left, there is a Goto back into the
incoming
> > > context.
> > > It all works great, except for when the line gets tied up by the
> > > DigitTimeout
> > > and ResponseTimeout bits when hangups aren't detected.
> > >
> > > I've tried using BUSYDETECT_MARTIN with busydetect=yes and it
didn't
> >
> > work.
> >
> > > The
> > > channel stays up after the outside caller hangs up.
> > >
> > > Since all of our inside phones are SIP lines, there is no problem
> > > detecting
> > > hangups when a voice conversation is taking place, since Asterisk
> > > obviously
> > > detects SIP hangups correctly whether it's SIP to SIP or SIP
to
> >
> > outside
> >
> > > line.
> > > The problem is really only when outside callers leave voicemail.
> > >
> > > Thanks,
> > >
> > > Chris
> > >
> > > On Wednesday 17 September 2003 08:09, Martin Pycko wrote:
> > > > Do you have silence in the channel when the remote user
hangs up
or
> >
> > busy
> >
> > > > tone ?
> > > >
> > > > If you have silence you can use maxsilence=x_seconds in
> >
> > voicemail.conf
> >
> > > > with
> > > > Voicemail2 application and that will make sure the calls are
hanged
> >
> > up
> >
> > > > after x_seconds of silence in the channel.
> > > >
> > > > If you have busy tone then use the busydetect=yes in
zapata.conf.
> > > > You can also limit the length of the voicemail message with
> > > > maxmessage=x_seconds in the voicemail.conf
> > > >
> > > > regards
> > > > Martin
> > > >
> > > > On Tue, 16 Sep 2003, Christian Hecimovic wrote:
> > > > > Hi,
> > > > >
> > > > > Try as I might, I can't get hangups detected on a
Zap channel
with
> > >
> > > loop
> > >
> > > > > start lines. So, after someone leaves a voicemail and
then
hangs
> >
> > up,
> >
> > > > > Asterisk doesn't know it, exits VoicemailMain2, and
loops back
to
> >
> > the
> >
> > > > > corporate greeting, tying up the line even though the
outside
> >
> > caller
> >
> > > has
> > >
> > > > > hung up.
> > > > >
> > > > > Therefore, I've added the following hideous hack -
er, code -
to
> > > > > voicemail2.c. It starts right after the call to
play_and_record()
> >
> > in
> >
> > > > > leave_voicemail().
> > > > >
> > > > > if (res != '#' && chan != NULL
&& !strncmp(chan->name, "Zap",
3))
> >
> > {
> >
> > > > > 	/* Hang up the Zap channel only */
> > > > > 	ast_softhangup(chan, AST_SOFTHANGUP_EXPLICIT);
> > > > > }
> > > > >
> > > > > Obviously, it hangs up the channel after the voicemail
has
been
> > >
> > > recorded,
> > >
> > > > > if the # key wasn't pressed, if the channel still
exists, and
if
> >
> > it's
> >
> > > a
> > >
> > > > > Zap channel. I couldn't see a way to do this with
AGI.
> > > > >
> > > > > Question: is this safe? I used a soft hangup because
the
channel
> >
> > is
> >
> > > > > controlled by another thread. I also modified channel.c
so
that
> > > > > ast_channel_free() sets chan to NULL after it's
freed, just in
> >
> > case.
> >
> > > Is
> > >
> > > > > there anything else I should be aware of? The code
seems to
work
> >
> > in my
> >
> > > > > testing, resulting in a proper hangup right after the
voicemail
> >
> > has
> >
> > > been
> > >
> > > > > recorded. I'm not up on my Asterisk internals, so
I'm not
totally
> > > > > confident about this.
> > > > >
> > > > > Thanks,
> > > > >
> > > > > Chris
> > > > >
> > > > >
> > > > > _______________________________________________
> > > > > Asterisk-Users mailing list
> > > > > Asterisk-Users@lists.digium.com
> > > > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > >
> > > > _______________________________________________
> > > > Asterisk-Users mailing list
> > > > Asterisk-Users@lists.digium.com
> > > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> > > _______________________________________________
> > > Asterisk-Users mailing list
> > > Asterisk-Users@lists.digium.com
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users

At 11:55 AM 9/17/2003, Doug Heckaman III <tad@heckaman.com>
wrote:
>I have a Cisco 2600 with a T1 CSU/DSU. Would that work the same way? I 
>would really like to see if I can use the cisco csu/dsu rather than buying 
>a 500 dollar T1 card...
The cards serve a completely different function.

The FXO card only handles POTS, and the WIC-1DSU-T1 is not a PRI interface...

I have no access to either of the VWIC Multiflex trunk cards, or the 
NM-xCT1-CSU cards so therefore I have no clue if they would work. (Not sure 
which one you are talking about)

Sorry