[asterisk-dev] Re: 'IAX2 call variable passing between servers '

Benny Amorsen benny+usenet at amorsen.dk
Mon Aug 14 15:01:25 MST 2006


>>>>> "AK" == Andrew Kohlsmith <akohlsmith-asterisk at benshaw.com> writes:

AK> It's time for me to eat some crow here. I was under the impression
AK> that Asterisk wouldn't jump back into the dialplan on a 302, but
AK> it does indeed do this. So, if pbxA calls in to pbxB's [ooga]
AK> context, and pbxB calls SIP/foo but gets a 302 to SIP/bar, pbxB
AK> will do the equivalent of Goto(ooga,bar,1).

Which versions of Asterisk do this?

AK> The exception to this is if the Dial() 'i' flag is used. In that
AK> case, Dial() will quit with a BUSY ${DIALSTATUS}, and
AK> ${CALLERID(DNID)} will show the original extension. I have not,
AK> however, found a variable that tells me what the SIP phone has
AK> suggested for a redirection. ${CALLERID(RDNIS)} does *not* contain
AK> this information, and in my testing this is all SIP-to-SIP, no
AK> IAX2 at all.

Is the i flag documented? It doesn't seem to be mentioned on
voip-info.org.


/Benny





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