[asterisk-users] Caller hang up not detected

[Steven Davison]

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 :)

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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