On 04/25/2012 04:45 PM, brya...@zktech.com 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"<kpflem...@digium.com> 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: kflem...@digium.com | SIP: kpflem...@digium.com | 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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Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | 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 --
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asterisk-users mailing list
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