[asterisk-dev] disable client side hangup after dialing 911on SIP Dial

[Kaloyan Kovachev]

On Fri, 9 Mar 2007 14:24:47 -0600, Tilghman Lesher wrote
> On Friday 09 March 2007 13:58, Kevin P. Fleming wrote:
> > Admin DeryTelecom wrote:
> > > I am talking about a SIP Dial and not a Zap Dial, it seems that
> > > this option is available on Zap Dial.
> >
> > Then this will be the responsibility of the SIP endpoint, not
> > Asterisk.
> 
> It cannot.  It is a violation of the SIP specification for a UAS to act
> in this way, and therefore, the UAC has no requirement to work that way,
> either.  See RFC 3261, section 15.1.2:
> 
>    A UAS core receiving a BYE request for an existing dialog MUST follow
>    the procedures of Section 12.2.2 to process the request.  Once done,
>    the UAS SHOULD terminate the session (and therefore stop sending and
>    listening for media).  The only case where it can elect not to are
>    multicast sessions, where participation is possible even if the other
>    participant in the dialog has terminated its involvement in the
>    session.  Whether or not it ends its participation on the session,
>    the UAS core MUST generate a 2xx response to the BYE, and MUST pass
>    that to the server transaction for transmission.
> 
> In other words, the server is not permitted to deny the BYE if it is
> properly formatted and within an existing dialog.

Just an option may be to use Page() and do this from the dialplan and small
change in Dial (http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Page).

As a next priority after the dial you may set a variable to specify, that the
caller's channel was up after hangup and if it was not to forward (instead of
hangup in app_dial) the called channel to a Page(SIP/xxx|dq) ... may be an
option in dial to transfer the called channel after the bridge ends (like it
is for both parties after answer) will be useful?

> 
> -- 
> Tilghman
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