[Asterisk-Users] Dialing dead SIP peers give misleading (BUSY) voicemail result ...

[Darren Nickerson]

> >>Is this the intended result when trying to dial a disconnected SIP
> >>extension, or have I misconfigured something? Does the dialplan above,
which
> >>was built using analog handsets, need to be more intelligent to deal
with
> >>SIP connectivity/registration status?
> >
> >
> > I've noticed the same thing. I don't believe its the desired behaviour
> > but I don't read C-code well enough to fix it either.
>
> Add this check to your stdexten macro:
> exten => s,3,DBget(sipcheck=SIP/Registry/${sipdevicenameinsip_conf})
>
> Check DBget on the wiki. If it fails, there's nothing registred and you
can forward
> directly to unavailable. If it succeeds, there's a current registration
and you can dial as usual.


Olle,

Having thought about this more, ... would you say your solution is

a) a workaround to an Asterisk bug, or
b) a normal part of SIP peer dialing best practice?

Does a SIP peer actually need to be registered with Asterisk to be able to
receive calls? I thought registration was only necessary when there's funky
network stuff in the way, such as NAT. If registration is REQUIRED, it seems
to me that Asterisk should already be doing the check you proposed I add
before it dials out as part of the 'Dial' application, and should simply
move on to the next step in the dialplan if fails this check. If one dials a
SIP peer that is unreachable for any reason (independent of registration
status), shouldn't the failure mode be 'unavailable' instead of 'busy'? Busy
just seems so misleading.

I tried an analogous experiment with a Zap extension ... unplugged it from
the Asterix box and dialed that extension. Asterix rang that extension until
timeout and then rolled over to the unavailable message. That's exactly the
behaviour I would have expected on a SIP dialout to an unreachable peer ...
Dial() would just think it's ringing the remote, but the remote never sees
the traffic cuz it's not in a connected state.

In the case of a network-disconnected SIP peer, the Dial application fails,
routing you to n+101. That's what happens when you dial a busy Zap extension
(I think the same thing also happens when someone's talking on that SIP
peer) making disconnected/unreachable SIP  Dial look like a BUSY. This seems
wrong to my (admittedly still quite naieve) understanding.

In the case of a physically-disconnected ZAP extension, the Dial application
succeeds, moving on to the next step in the dialplan. That is much more in
line with my expectation.

Thoughts?

-Darren

--
Darren Nickerson
Senior Sales & Support Engineer
iFax Solutions, Inc. www.ifax.com
darren.nickerson at ifax.com
+1.215.438.4638 office
+1.215.243.8335 fax