asterisk users - Jun 2008 - Delaying SIP disconnect after incoming call hangs up?

I'm looking for a way to delay the disconnection of a call to
a SIP extension (or pad it with silence) for a few seconds, after
an incoming call to that extension hangs up.

Rationale:  I have an Asterisk PBX (current 1.4.20 codebase), with
a Leadtek BVP8051S ATA hooked to an analog phone which has a
built-in answering machine.  Incoming SIP connections to the
appropriate extension are dialed to this SIP ATA, the phone rings,
and the answering machine picks up... all as it should be.

However, when the caller hangs up, the ATA immediately starts
generating a fast-busy disconnect/congestion beeping.  The
answering machine doesn't recognize this as a hang-up situation
(it expects to hear the line go silent) and it keeps recording beeps
until its message-length timer expires and it hangs up the line
to the ATA.

Unfortunately, I can't change the answering machine's behavior,
and I don't think it's possible to change the Leadtek BAP8051S to
just go silent.

So, what I'm hoping, is that there is some way within Asterisk to
change the PBX behavior when the incoming call disconnects, so that
it can defer sending the disconnect event to the SIP extension
for 10 or 15 seconds... enough "quiet time" for the answering machine
to recognize end-of-call and hang up.  I think that either sending
nothing (no RTP stream) to the SIP extension, or sending "silence"
or "comfort noise" frames, would work fine.

I've looked through the documentation and through a fair bit of
the source code, and haven't found anything which actually works.
I tried adding an "h" hangup rule to the dialplan for this
extension, with a Wait(10) action, but this seemed to have no effect.
Either the "h" rule isn't working, or the disconnect frame has
already
been processed and a SIP BYE has been sent.

I've only been able to figure out one approach which *may* work...
use an "h" hangup rule for the extension, which runs a DeadAGI()
script, which contacts the SIP ATA via its http administrative
interface and reboots the ATA (which immediately drops the line).
This may very well work, but is about as elegant as a bag-full
of wet tree sloths, and I'd like to do a better job than this.

Is there any provision in Asterisk for being able to "catch" the
hangup/disconnect of the far end of a connection, and either wait
(with no activity) for a fixed period of time, or do the equivalent
of a Play() to send the contents of an audio file to the remaining
extension (the target of the Dial() in the extension dialplan)?

Currently, the SIP extension in question is behind a NAT, and
I've set "canreinvite=no", so I believe that all of the SIP and
RTP traffic is going through Asterisk.  It seems to me that it
*ought* to be possible for Asterisk to catch the end-of-
connection situation and react in some way other than immediately
disconnecting the receiving SIP peer, but I'm not sure that any
such capability has been implemented.

I realize that the outside-the-box answer to this would be "Why
use an answering machine?  Use the PBX voicemail!" but that's
not entirely desirable in this situation.  Since the phone /
answering machine is analog, it has no "message waiting" light
available to let us know that a call has come in, and we'd also
lose the ability to "jump onto" a call which is in the process
of being recorded.  My wife is comfortable with how the existing
answering machine system works, and I'd rather present her with an
IP-based solution which doesn't change the behavior she's used to...
she's not the most technophilic person around.

Thanks in advance for any ideas you can throw my way!

On Tue, Jun 10, 2008 at 5:28 PM, Dave Platt <dplatt at radagast.org>
wrote:
> I'm looking for a way to delay the disconnection of a call to
> a SIP extension (or pad it with silence) for a few seconds, after
> an incoming call to that extension hangs up.
>
> Rationale:  I have an Asterisk PBX (current 1.4.20 codebase), with
> a Leadtek BVP8051S ATA hooked to an analog phone which has a
> built-in answering machine.  Incoming SIP connections to the
> appropriate extension are dialed to this SIP ATA, the phone rings,
> and the answering machine picks up... all as it should be.
>
> However, when the caller hangs up, the ATA immediately starts
> generating a fast-busy disconnect/congestion beeping.  The
> answering machine doesn't recognize this as a hang-up situation
> (it expects to hear the line go silent) and it keeps recording beeps
> until its message-length timer expires and it hangs up the line
> to the ATA.
>
> Unfortunately, I can't change the answering machine's behavior,
> and I don't think it's possible to change the Leadtek BAP8051S to
> just go silent.
>
> So, what I'm hoping, is that there is some way within Asterisk to
> change the PBX behavior when the incoming call disconnects, so that
> it can defer sending the disconnect event to the SIP extension
> for 10 or 15 seconds... enough "quiet time" for the answering
machine
> to recognize end-of-call and hang up.  I think that either sending
> nothing (no RTP stream) to the SIP extension, or sending
"silence"
> or "comfort noise" frames, would work fine.
>
> I've looked through the documentation and through a fair bit of
> the source code, and haven't found anything which actually works.
> I tried adding an "h" hangup rule to the dialplan for this
> extension, with a Wait(10) action, but this seemed to have no effect.
> Either the "h" rule isn't working, or the disconnect frame
has already
> been processed and a SIP BYE has been sent.
>
> I've only been able to figure out one approach which *may* work...
> use an "h" hangup rule for the extension, which runs a DeadAGI()
> script, which contacts the SIP ATA via its http administrative
> interface and reboots the ATA (which immediately drops the line).
> This may very well work, but is about as elegant as a bag-full
> of wet tree sloths, and I'd like to do a better job than this.
>
> Is there any provision in Asterisk for being able to "catch" the
> hangup/disconnect of the far end of a connection, and either wait
> (with no activity) for a fixed period of time, or do the equivalent
> of a Play() to send the contents of an audio file to the remaining
> extension (the target of the Dial() in the extension dialplan)?
>
> Currently, the SIP extension in question is behind a NAT, and
> I've set "canreinvite=no", so I believe that all of the SIP
and
> RTP traffic is going through Asterisk.  It seems to me that it
> *ought* to be possible for Asterisk to catch the end-of-
> connection situation and react in some way other than immediately
> disconnecting the receiving SIP peer, but I'm not sure that any
> such capability has been implemented.
>
> I realize that the outside-the-box answer to this would be "Why
> use an answering machine?  Use the PBX voicemail!" but that's
> not entirely desirable in this situation.  Since the phone /
> answering machine is analog, it has no "message waiting" light
> available to let us know that a call has come in, and we'd also
> lose the ability to "jump onto" a call which is in the process
> of being recorded.  My wife is comfortable with how the existing
> answering machine system works, and I'd rather present her with an
> IP-based solution which doesn't change the behavior she's used
to...
> she's not the most technophilic person around.
>
> Thanks in advance for any ideas you can throw my way!
>
Your ATA is to blame.  It is generating the noise you describe.  Maybe
there is a setting on the ATA to address this.

Idea, try an Grandstream 286 ATA.  I know people bash Granstream and I
have in the past as to their phones, but their ATAs are pretty good.

Thanks,
Steve T