[Asterisk-Users] Asterisk Passthrough

[Jayson Vantuyl]

On Tue, Mar 02, 2004 at 02:19:16PM -0500, Steve Creel wrote:
> [incoming]
> exten => 12345,1,Dial(ZAP/g2) ; Send incoming call for 12345 to the PBX
FWIW, I've done something like this and it was absolutely wonderful.
We were actually running new phones and putting them in parallel with
the existing system over the same phone lines (they ran 4-pair UTP to
each phone jack, so we just stole the outer pair and bought some magic
adapters to pull out the "second line").

You can imagine the surprise when both phones worked simultaneously.
It was even more surprising when we were training people on the new
phones by having them dial an "outside line" and then use the new
dial codes for voicemail and such.  It went over VERY well.

> >How I can forward a call? It's simply an extension.conf rule?
> Yes.
Most people miss this.  Use the Dial application (as the example shows).
Dial is used for outside lines and such, but it is *'s fundamental way
to make one channel dial to another.  Virtually every situation where
you are "forwarding" something (say Zap to SIP, SIP to IAX, TDM to
TDMoe) you end up using a Dial.

> >When I make the forward in this way (with extension.conf rule) asterisk
> >make some work or is a simple passthrough from interfaces?
> Yes, it's some switching/callsetup work, but no codec translation, which
> is by far your biggest CPU consumer.
It acts as a simple passthru for the CHANNELS.  That is, what comes in
on an individual channel goes out on another.  Mapping the whole T1
would be another story (it can be done, I had to once).  DACS works well
for that but Asterisk can't get at the calls.  I recommend the above.
The only time it doesn't work well is when people want to do something
with "line 5".  I had a situation where certain lines couldn't dial long
distance.  Since the above would dynamically choose a line, it would
cause unexpected problems because the old PBX's line X was no longer
actually the same T1 channel on the outside.
> 
> >I need that calls "from PRI to PRI" don't load the computer.
> >I want to use all CPU to (future) SIP calls.
Once the call is linked, all the load is on the Zaptel board.  That is
REALLY handy.  I can't tell you how surprised some of my customers get
when I have three machines switching 300 lines with like 5% or so load a
piece.

Feel free to e-mail me or jabber me (same as my e-mail address) if you
have problems.  I love to help set things like this up--especially in an
more casual setting (you never get to have FUN with people's
businesses).

Jayson Vantuyl