[asterisk-users] Hangup Cause and SIP Response Code

[Kevin P. Fleming]

On 04/25/2012 04:45 PM, BryantZ at 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"<kpfleming at 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: kfleming at digium.com | SIP: kpfleming at 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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>
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-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kfleming at digium.com | SIP: kpfleming at digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org