[Asterisk-Users] FW: This newbie gives up for now - sadly (2)

[Adam Goryachev]

asterisk-users-admin at lists.digium.com <> wrote:
> This newbie has been trying out Asterisk. It has been both a)
surprisingly
> painful and b) impressive in terms of helpful support from
> other users.
> 
> Having got two phones to communicate and then got voicemail MWI going
> (neither painlessly) I decided the next step was to implement
> call transfer
> as per nearly all commercial PBX systems i.e.
> 
> 	hold call
> 	consult another extension
> 	either exit and let the two speak
> 	or get back the original caller
> 
> - an utterly fundamental office procedure on a PBX.
[SNIP]
> I found comments like
> 
> "This is where it might come down to redesigning the way
> calls are dealt
> with in an organization. Sometimes new phone systems do this,
> and hopefully
> the company sees new efficiencies with dealing with the customer in
> general." 
> 
> unhelpful and out of touch with user's and managers needs:

Actually, this feature is extremely simple to use, and I don't
understand why you might have asked the question and got anything other
than the simple instructions on how to make it work. In fact, the
default sample conf files already includes the needed config details.

Of course, that implies that you are using hardware that supports that
function. AFAIK, only Zapata connected hardware and some IP phones
support that feature. So, if you are trying to use CAPI, or I4L, or IP
phones, then maybe you are having that problem, and maybe it isn't
possible. Other people on this list are better qualified to respond, but
I am pretty sure the cisco and snom phones are capable of this.

So, it might be that you have chosen non-optimal hardware to 'test'
asterisk with. It would be 'better' to choose the easiest hardware to
learn the software, and then after you know about the software, try with
more complex/less supported hardware.

It would seem to me that a lot of people (and hey, I did this too), try
to use non-digium hardware to 'test' asterisk with before going out and
buying the digium hardware for full production use. However, this makes
it much more difficult to 'test' asterisk because it is harder to
configure, and often causes problems you wouldn't normally have (ie,
echo, etc).

(Of course, there are STILL valid reasons for not using digium
hardware!! Ie, in Australia, it is still illegal to use any of their
hardware (for PSTN connectivity) because they do not have the relevant
approval. Yes, the E400P is supposedly approved, but where is the
paperwork/stickers/etc? Is that approval going to carry across to the
TE405P ?? In fact, where is the TE405P?)

[SNIP]
> help other newbies to get going but I think its time to give
> up and re-visit
> Asterisk in some months time. I am really disappointed not to
> be able to use
> asterisk now.

This can often work surprisingly well. Just going away and coming back
in a few months allows two things:
A) You have time to mature/learn new things about Linux/IP Telephony.
B) The project has time to mature, new/better documentation + more
features + more bug fixes.

This doesn't just apply to you, but hopefully the above makes you sit up
and consider that you are blaming your problems/difficulty on asterisk
when in fact you should blame to in-compatible hardware or even the
protocols you are forcing asterisk to use (ie, SIP/H323)

Regards,
Adam

 --
Adam Goryachev
Website Managers
Ph:  +61 2 9345 4395                        adam at websitemanagers.com.au
Fax: +61 2 9345 4396                        www.websitemanagers.com.au