[asterisk-users] Caller hang up not detected

Danny Nicholas danny at debsinc.com
Thu Jan 21 08:14:36 CST 2010


Since there are no DAHDI lines involved, polarity probably won't help.
Call-limit might or might not help with this.  Does "core show channels"
show anything after the callee hangs up?

 

  _____  

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of hugolivude
Sent: Thursday, January 21, 2010 7:47 AM
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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