Routing calls through two FXO cards without answering?

[fredtheman]

Hi

I'm having a hard time finding information to know whether Asterisk is able to route incoming calls from one FXO card to another one without answering the call and asking the user to type an extension. It looks like I'm the only person on this planet who ever had the need to hide from customers the fact that I'm off to a remote office, ie. not ask them to dial another POTS number to find me :-)

A picture being a thousand words, you'll find a diagram, the configuration files and two log files here:

http://codecomplete.free.fr/asterisk/goal.html

Can any experienced Asterisk users out there tell me whether what I'd like to do is possible, and if yes, show me some example configuration files that I could use?

Thank you!

[mberlant]

Asterisk is able to do whatever you can imagine. You just have to tell it what you need. It appears that you have not defined in extensions.conf what you want to happen when the one Zap channel receives an incoming call. As you read through the previous postings here and the documentation at www.asterisk.org and www.voip-info.org you will pick up the education you will need to understand how Asterisk works. Once you have that you will be able to set up your Asterisk for Call Forwarding.

By the way, is there some reason you don't just carry a softphone or hardware adapter with you as a VoIP extension of Asterisk? That's probably the most popular way of receiving calls when away from home. X-Lite works great with Asterisk.

[fredtheman]

Originally Posted by mberlant

It appears that you have not defined in extensions.conf what you want to happen when the one Zap channel receives an incoming call.

Yes, I did. The context [cherbourg] is applied to the Zap/1 interface in zapata.conf, and handled in extensions.conf. It just dials out a number through Zap/2. Do I need something else?

Originally Posted by mberlant

Once you have that you will be able to set up your Asterisk for Call Forwarding.

If Call Forwarding means that 1) phones in the main office (where Asterisk is) won't ring anymore, or 2) that Asterisk will need to go off-hook on Zap/1 to first answer the call before ringing the remote phone, it's not what I need: I want Asterisk to simply ring all phones at the local office and one phone at the remote office, all without going off-hook, so that customers aren't charged for unanswered calls.

Originally Posted by mberlant

By the way, is there some reason you don't just carry a softphone or hardware adapter with you as a VoIP extension of Asterisk?

Because the ADSL modem to which the second FXO card is connected allows for free calls over the POTS network (so-called "triple play ADSL"), so that I can use a regular phone at the remote office. If I can't get Asterisk to handle things this way, I'll buy an IP phone or ATA adapter for the remote office and use SIP/IAX2 instead.

Thank you!

[mberlant]

Using a remote client is cleaner, but your description indicates that all of your processing pieces are in order.

I recommend that you go into extensions.conf and put a few NoOp("I got this far") lines in the section of code that receives the inbound call and sends it out the second line. Then, place a test call while watching your CLI to see just what happens. If you don't see any activity, your problem is in zapata.conf not handing off the call properly to extensions.conf. If you do see activity, you can see by the NoOp comments just where the call is failing.

Let us know.

p.s. Call Forwarding is taking the call and sending it somewhere else without manual intervention. Call Transfer is manually picking up the call and then sending it somewhere else. Cheers.

[fredtheman]

Originally Posted by mberlant

I recommend that you go into extensions.conf and put a few NoOp("I got this far") lines in the section of code that receives the inbound call and sends it out the second line. Then, place a test call while watching your CLI to see just what happens. If you don't see any activity, your problem is in zapata.conf not handing off the call properly to extensions.conf. If you do see activity, you can see by the NoOp comments just where the call is failing.

Great. So here's the extensions.conf I'll use:

[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no

[globals]
TRUNK=Zap/2

[cherbourg]
exten => s,1,NoOp(Before Dialing out through ${TRUNK})
exten => s,n,Dial(${TRUNK}/01XXXXXX)
exten => s,n,NoOp(After Dialing out through ${TRUNK})

Originally Posted by mberlant

p.s. Call Forwarding is taking the call and sending it somewhere else without manual intervention. Call Transfer is manually picking up the call and then sending it somewhere else.

Thanks for the clarification. I'll actually get rid of the rPath PoundKey Linux distribution, install Fedora 5 instead, compile the latest Zaptel + Asterisk, and report back what I found.

Thx everyone for your help

[fredtheman]

For those trying this... with much help from the experts on the IRC channel, especially the mighty Strom_C...

1. we did get some routing happening between FXO2 -> FXO1 (not the other way around, go figure), but...
- all I hear when the I pick up the phone on FXO1 is noise,
- we had to have the ingress FXO card go off-hook to handle the call (and charging customers a call even if no one ends up answering...), and
- the FXO detects neither caller ID nor call termination.

So it doesn't look like routing calls between txo FXO cards is doable, at least with X101P clones (possibly OK with Digium cards). I'm told a much better solution is to go SIP (a.k.a. "media gateway".)

2. In zapata.conf, make sure you use "channel=>1-2" instead of "channel=>1,2"

3. Here's what you can try in extensions.conf

[general] 
static=yes 
writeprotect=no 

[globals]
TRUNK=Zap/1
;TRUNK=Zap/2
NUMBER=5551234

[incoming]
exten => s,1,Answer
exten => s,n,Wait(1)
exten => s,n,Dial(${TRUNK}/${NUMBER})

;Didn't work
;exten => s,1,NoOp(Before Dialing out through ${TRUNK} - Caller ID is ${CALLERID}) 
;exten => s,n,Dial(${TRUNK}/${NUMBER}) 
;exten => s,n,NoOp(After Dialing out through ${TRUNK})

HTH
Fred.