[asterisk-users] Caller hang up not detected

[hugolivude]

Thanks responding guys.  It appears that it's the canreinvite that's causing
the problem.  Interesting results tho:

With canreinvite=yes, leaving out the transfer options leads to a Dial
command that _never_ exits:
exten => 1,n,Dial(SIP/14168724765 at 6135551212-sw1|120|g)

I have 2 channels seemingly forever - kinda scary!

With canreinvite=yes, but with transfer options, I get the behaviour I
reported - it hangs up but only after the callee hangs up.
exten => 1,n,Dial(SIP/14168724765 at 6135551212-sw1|120|gtT)

adding this had no effect:
exten => 1,n,Hangup

except of course that it hangs up after the callee hangs up (but not when
the caller hangs up)

With canreinvite=no, Dial exits when the caller (me) hangs up in both cases,
which is of course the desired behaviour.

Will I still be able to transfer calls w/ canreinvite=no?  I'd like to test
that but my second problem is that the feature codes don't seem to be
working for me!  I posted that problem in a separate thread, but it didn't
get posted on the list; that seems to happen to me frequently.

Thanks again guys.

H



On Thu, Jan 21, 2010 at 10:39 AM, Steven Davison
<steven.davison at ntsols.com>wrote:

>  Hi,
>
>
>
> Couple of questions...
>
>
>
> Are you allowing reinvites, and what happens if you change the dialplan to
> this?
>
>
>
> exten => 1,n,Dial(SIP/14168724765 at 6135551212-sw1|120|gtT)
> exten => 1,n,Playback(vm-goodbye)
> exten => 1,n,Hangup()
>
>
>
> help this helps J
>
>
>
> Steven Davison
>
> Net Technial Solutions
>
>
>
> *From:* asterisk-users-bounces at lists.digium.com [mailto:
> asterisk-users-bounces at lists.digium.com] *On Behalf Of *hugolivude
> *Sent:* 21 January 2010 13:47
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] Caller hang up not detected
>
>
>
> Hi,
>
> I'm having trouble getting Dial to exit when the caller hangs up in
> Asterisk 1.4.21.2.
>
> I use a POTS line to call into the DiD given to me by VOIP service
> provider.  When the call comes in, I have the VOIP provider send it to
> another POTS line.  All this works fine however when the caller (me) hangs
> up, the Dial command does not exit.  The callee stays connected (and my
> billing continues!). Dial doesn't exit until the callee hangs up. Here's a
> snip from extensions.conf:
>
> exten => 1,n,Dial(SIP/14168724765 at 6135551212-sw1|120|gtT)
> exten => 1,n,Playback(vm-goodbye)
>
> Here's the CLI output (verbosity = 4):
>
> -- Executing [1 at Trunk-0001:1] NoOp("SIP/77.57.127.163-09023590", "") in
> new stack
> -- Executing [1 at Trunk-0001:2] Dial("SIP/77.57.127.163-09023590",
> "SIP/14168724765 at 6135551212-sw1|120|gtT") in new stack
> -- Called 14168724765 at 6135551212-sw1
> -- SIP/6135551212-sw1-090275d0 is making progress passing it to
> SIP/77.57.127.163-09023590
> -- SIP/6135551212-sw1-090275d0 answered SIP/77.57.127.163-09023590
> *** I hang up here, but the call continues.  A while later the callee hangs
> up:
> -- Executing [1 at Trunk-0001:3] Playback("SIP/77.57.127.163-09023590",
> "vm-goodbye") in new stack
> *** obviously I don't here this, just see it in the CLI
>
> I'd be grateful for any troubleshooting tips that will help me get asterisk
> to quit the Dial command when the originator hangs up.
>
> Thanks,
> H
>
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