[asterisk-users] DAHDI hangup detection

[Danny Nicholas]

This is how I'd do it with the "old" dialplan (extensions.conf) - 

-          exten => s,1,dial...

-          exten => s,n,Macro(voicemail-test,${EXTEN})

-          exten => s,n,hangup

-          [macro-voicemail-test]

-          Exten => s,1,AGI(linetest.agi,${ARG1})

-          Exten => s,n,gotoif($["${VAR}" = "AVAIL"]?hangup)

-          Exten => s,n,voicemail.

-          Exten => s,n(hangup),hangup

 

Linetest.agi would run an AMI session to see if DAHDI-X was in use and
return a variable as AVAIL or undef.  If the line is in use, record
voicemail, else hangup.

 

 

  _____  

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Stephen Brown
Jr
Sent: Tuesday, September 15, 2009 11:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DAHDI hangup detection

 

Ok on the workaround, how would I implement it? I'd like to give that a
shot. 

On Tue, Sep 15, 2009 at 12:23, Danny Nicholas <danny at debsinc.com> wrote:

The issue is that POTS as a technology does not have Answer/Hangup
Supervision control (This is per the good folks at Digium).  Your local
Telco may or may not be of assistance, but a simpler workaround might be to
hangup on silence or query the line through AMI before transferring to
voicemail.


-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Stephen Brown
Sent: Tuesday, September 15, 2009 11:18 AM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] DAHDI hangup detection

I am running Asterisk 1.6.1.6 and DAHDI/DAHDI tools 2.2.0.2 compiled
from source on a Debian Lenny box, also running FreePBX 2.5.2. I also
have an OpenVOX TDM400 card installed with an FXO port on port 1, and an
FXS port on port 2. I have a POTS line installed and working on the FXO
port.

However I've encountered a weird problem that I can't seem to figure
out. I have incoming POTS line calls set to ring a SIP extension (a
cisco IP phone). It works, but here is the issue:

- If an inbound call comes in on the POTS line and hangs up before the
call hits voicemail or during the voicemail greeting, Asterisk does not
appear to detect this condition.
- As a result of this, a voicemail is being left with a dialtone and
notifying me

As a workaround, I have set busy detection in chan_dahdi.conf and set 12
seconds for a minimum voicemail message in voicemail.conf, but I must be
missing something or something is potentially broke. While this
workaround works, I don't suspect this behavior I am experiencing is
normal and I am having a hard time putting a finger on it.

Thanks for any replies.

Stephen



_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090915/ac4c4235/attachment.htm