Re: Hangup Cause and SIP Response Code

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On 04/25/2012 04:45 PM, BryantZ@xxxxxxxxxx wrote:
Kevin

I am using 1.8.x&  10.x

Then you have SIP_CAUSE available, although you'll have to enable it because it is off by default due to performance concerns.


Bryant Zimmerman (ZK Tech Inc./interNetGR)

(616) 855-1030 Ext. 2003

On Apr 25, 2012, at 5:00 PM, "Kevin P. Fleming"<kpfleming@xxxxxxxxxx>  wrote:

On 04/25/2012 07:08 AM, Bryant Zimmerman wrote:
I can log the ISDN cause code using ${HANGUPCAUSE} but I also need to
track the actual SIP response code as well. How do I get access to it
durring the hangup?

It's rather hard to answer that question without at least knowing what version of Asterisk you are using. In some versions there is a SIP_CAUSE feature that can be used to extract that information (although this has been reimplemented for Asterisk 11 in a way that doesn't affect performance as much as the old method did).

--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kfleming@xxxxxxxxxx | SIP: kpfleming@xxxxxxxxxx | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com&  www.asterisk.org

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_____________________________________________________________________
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New to Asterisk? Join us for a live introductory webinar every Thurs:
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--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kfleming@xxxxxxxxxx | SIP: kpfleming@xxxxxxxxxx | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
              http://www.asterisk.org/hello

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


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