Re: Re: axc ( Get VAR in Log) ?

[Gregg Liming <gregg@xxxxxxxxxxx>]

Quoting hometecky (2/16/07 2:10 PM):

> Well I wonder if its something in the axc.conf that I have set up
> incorrectly?

All of the DEBUG statements that you are seeing in the console or log
are specific to trixbox settings.  They're not impacted by anything that
is configurable in axc.  I'm hoping that some trixbox user may be seeing
this dialog and pipe up w/ the setting.  To help guide you, it would be
some setting that impacts verbose logging that is specific to the
asterisk manager interface.

> Let me ask some questions.

Sorry about the overly lengthy answers ;)

> 1) network_type = VOIP or PSTN or VPSTN
> are these just lables that I want to see in Switch Board or do they
> actualy do something. I understand what they are abreveating. but
> what is the difference between a voice over IP network and a V public
> switched telephone network.

A very good question.  Since the possible values don't seem to be
limited in the current schemas nor are their semantics defined, this is
a "gray area".  From SB's perspective (and, I'll assume that
James will
pipe up if I misspeak anything), they're just labels in terms of
choices.  However, SB does keep track of matching up line availability
and network type w/ the dialler type.  So, one strategy is to think of
how you want things to actually work (more on this below).

axc actually does understand the difference between PSTN and everything
else in that it assumes PSTN has a line capacity of a single call at a
time--whereas all of the other types (which I'm generically thinking of
as VOIP) have virtualization capability.  Eventually, I'll probably add
(yet one more) configuration option in axc to specify line capacity.
So, for now, axc will always report a line as being "busy"
(incoming or
outgoing) if a call comes in on something considered to be PSTN and will
always keep all other line types "free".  A reason for this is
that in
my own setup, I can have multiple incoming and outgoing calls on the
same VOIP line.

> 2) dialler = None, OFFICE, VPSTN, PSTN
> could you explain these options?

If you don't want the line showing up in SB's dial list, mark the
dialler as "None"; otherwise, make it the same value as
network_type.
This may seem redundant (and, I would agree), but I'm trying to live w/
the schema.

> 3)Dial context = intern Boes this need to match the dial context in
> Asterisks extensions, Like from-internal
> or this does not matter?

Yes--it must match up exactly and therefore does matter.  I actually
quite like this capability because it allows you to insert into contexts
that you might not otherwise connect into.  For example, my dialplan for
my main desk phone automatically dials out on a VOIP carrier unless I
prefix it w/ a "9"--in which case it goes to my PSTN circuit. 
Now, I
don't want to have axc or SB worry about whether to prefix or not
numbers w/ a 9.  So, if I want to use SB's dialler to dial the PSTN
line, then it matches on a context that intercepts post the "9"
prefix
(hopefully that makes some sense).

> 5) set hub = 1 for no xAP HUB running on the Asterisx box? is this
> correct?

nohub (not "hub) should be set to 1 if no xAP hub is running

> 0= yes there is a HUB running on the Asterisk box?

Correct--a "double negative" ;)

> I have a xAP HUB running on a different computer in the network so I
> have mine set to =1 hope I have that right.

Yes, if you don't have a hub running on the same system as axc is
running on (remember, axc doesn't have to run on the asterisk box--but,
most likely is), then set nohub=1

> here is part of my config may be you will see something
> these are the extensions:
>
> [kitchen]
> channel			= SIP/400
> incoming		= Packet8, vitel, stanaphone, fwd
> outgoing		= Packet8,vitel
> [loft_office]
> channel			= SIP/403
> incoming		= vitel
> outgoing		= vitel
>
>
> There are my providers and one Zap that I use for Packet8
>
> [vitel]
> incoming_channel	= IAX2/64.2.142.26
> outgoing_channel	= IAX2/vitel
> country_code		= 1
> network_type		= VPSTN
> dialler			= PSTN

I would change dialler to VPSTN (to be consistent w/ network type)

> dial_context		= internal
> number_formatter1       = s/1?(\d{10})/$1/

In case it's not obvious, the above regex (for number_formatter1) strips
off a leading "1" from a 10-digit US number.  This keeps the
lookups
that SB does "clean".  The same thing can be applied to other
numbers if
you use extension prefixing like I mentioned above.

> [stanaphone]
> incoming_channel	= SIP/08149345
> outgoing_channel	= SIP/stanaphone
> country_code		= 1
> network_type		= VOIP
> dialler			= none
> dial_context		= internal
> number_formatter1       = s/1?(\d{10})/$1/

Let's suppose that you wanted to have one "virtual" VOIP channel
that
will match outgoing numbers from either stanaphone or vitel (I'm
assuming for the sake of this example that they're both VOIP-PSTN
gateways).  You could instead do something like the following:

[voip_pstn]
incoming_channel	= IAX2/64.2.142.26,SIP/08149345
outgoing_channel	= IAX2/vitel,SIP/stanaphone

... then include the other stuff as before (you could also leave off
standaphone from outgoing if you really didn't want to dial using it)

Then, you'd alter the incoming line for your kitchen line to be like:
incoming		= Packet8, voip_pstn, fwd

I'm not suggesting that you need to do the above as the way that you are
doing it is identical to what I was originally doing before becoming
very dependent on SB (vice mh) to perform all of my telephony related
UI-duties.  The virtualization allows a reduction of the number of lines
that appear.  In addition, in my case, I have multiple outbound carriers
that exist only for redundancy and cost savings that my internal
dial-plan handles.

> [fwd]
> outgoing_channel	= IAX2/fwd
> incoming_channel	= IAX2/fwd
> country_code		= 1
> network_type		= VPSTN
> dialler			= none
> dial_context		= internal
> number_formatter1       = s/1?(\d{10})/$1/
>
> [Packet8]
> # you've no doubt noticed that incomgin_channel doesn't get used;
> well it's here in case
> # I change my mind
> incoming_channel	= Zap/1
> # the Zap/<x> syntax applies to x100p devices as well as other
> related Digium cards
> outgoing_channel	= Zap/1
> country_code		= 1
> network_type		= PSTN
> dialler			= PSTN
> dial_context		= internal
> number_formatter1       = s/9?1?(\d{10})/$1/
>
>
> FYI I do get xAP CTI messages and Meteor CID messages.
> Just want these messages below to Stop.
> every second there is another. Any Ideas would be nice.

Understood.  I will try to do some googling on trixbox and this DEBUG
stuff.  The reason for the rapid rate is that there is a "bug"
(or
undocumented "feature") of the asterisk manager interface that
will
cause some of the messages to buffer up and not be sent out (there is no
obvious pattern or reason for this occurring).  The "GetVar"
message is
a "light-weight" way of flushing any buffering.  This is
necessary to
ensure that axc maintains tight sync w/ asterisk.  The problem in your
case is that some trixbox setting is causing you to see way to much.

> Thanks
>
> Feb 16 13:33:58 DEBUG[3174] manager.c: Manager received
> command 'GetVar'
> Feb 16 13:33:59 DEBUG[3174] manager.c: Manager received
> command 'GetVar'
> Feb 16 13:34:00 DEBUG[3174] manager.c: Manager received