[asterisk-dev] Has the meaning of the "s" extension changed recently?

[Olle E Johansson]

2 feb 2006 kl. 17.20 skrev Enzo Michelangeli:

> Wasn't the "s" extension supposed to mean "the null extension"?
> http://www.digium.com/asterisk_handbook/extensions.conf.html says:
>
> s: Defines how to route a call when no other routing information  
> has been
> received. On a PRI or local FXS line, we will receive a number  
> string to
> route the call by. When receiving a call from an analog (FXO) line, we
> won't get any routing information, just a ring. In this case the 's'
> extension will be used.
>
> Also http://www.voip-info.org/wiki/index.php?page=Asterisk+s+extension
> says similar things, and that was consistent with the behaviour of
> Asterisk 1.0.x .
>
> However, I recently noticed that SIP INVITEs sent to a null user (e.g.
> "INVITE sip:jaxhk.hn.org SIP/2.0", generated by a
> "Dial(SIP/@jaxhk.hn.org)" in the dialplan) do not trigger the
> "exten => s,..." lines in the receiving peer (running * 1.2.4).  
> Instead,
> the latter appears to look for a _user_ equal to the domain, in an  
> empty
> domain:
>
>    Looking for jaxhk.hn.org in default (domain )

Looks and smells like a bug, since the source actually have code for  
routing to "s" in this
specific situation.

Please open a bug report and we'll look at it.
/O