Pass parameter from Asterisk to Asterisk via IAX/SIP

[nauliv]

Hello,


The config:
-------------------


PSTN <----pri-t1----> Asterisk_Box_1 <---Local LAN----> Asterisk_Box_2


When a call comes in on the PRI, it lands on Asterisk Box 1.

In some cases, the call will then be passed onto the Asterisk Box 2 using "Dial(IAX2/loginname)" with 'g' flag, so that the dialplan on the Box 1 continues after Box 2 hangs up the call.


The Question:
------------------

Depending on what choices the caller has made while going through the dialplan on Box 2, I'd like the dialplan on Box 1 to do different things after Box 2 hangs up.

How can the Box 2 pass a message to the Box 1, when hanging up the call, to ask Box 1 what to do next ?


Hope there is a way to do so...


Thanks to All !

 

[busster]

Can you further explain what you are trying to do.

Why multiple * boxes?

 

[nauliv]

Thanks for your e-mail.
I actually have 3 asterisk boxes. Each are doing a particular task:
* one is connected directly outside to the T1, and handle basic applications.
* one is more of a development box, and has the latest and greatest dialplans, a database server, some AGI scripts, a SQL server, etc.
* one has a LumenVox server installed, and is more dedicated to doing voice recognition.

The purpose of having all these in different boxes are (not in order of importance):
* stability: there's a lot of development going on in the dev box, which requires often times to restart the asterisk process, and being on a separate box, there is no risk to disconnect current calls or having to wait for everyone to disconnect
* load balancing: not all tasks are handled by one machine
* fault tolerance: the dev box or lumenvox can crash, the first one is still up

And the problem is that when the call is sent to the lumenvox box or the dev box, and the dialplan want to "send it back to the original T1 box", we can simply hang up the call and let the T1 box continue the dialplan, but how to pass a PARAMETER so that the T1 box knows how things happened on the second asterisk box.

Hope it makes sense ;-)

Thanks for all your help, Folks !

 

[busster]

We have two asterisk boxes connected via IAX trunking. Primary office has 1000 - 1099 extensions. Office two has 1100 thru 1199 extensions. Incoming calls and outgoing calls except 911 all go through the Primary office. DID calls for extensions in the 1100 range go directly to office two. The second office gets the incoming callerid just as if they were in the primary office. If they transfer the call back to the primary office, we still see the callerid. These are transferred calls, but assume that the basics are still the same. If the Lumenox is doing the voice-rec, it will being doing a transfer to either local extensions, or out iax to one of the other two boxes, correct? Callerid should be passed.

 

[nauliv]

Hi !

Well.. CallerID works and is not my issue here.
My question is:

* I receive a call on Asterisk-Server-A
* I then pass it on to Asterisk-Server B
(using IAX2, through a Dial).
* When Asterisk-Server-B hangs up the call, Asterisk-Server-A continue the dialplan.

I'd like Asterisk-Server-B to be able to pass a "return value" to Asterisk-Server-A to make a decision on how to further handle the call once it's back on Asterisk-Server-A.

Thanks Folks !

 

[engjohn]

Well, if you cannot find the proper way (if there is one)...

You could probably rewrite the callerid on box-B to append a "Return Value" to the callerID and then parse the callerID on Box-A to retrieve and remove the appended value...

Just a thought...

 

[nauliv]

I thought of that, but it's the pbx that has been CALLED that needs to pass parameters back to the pbx that was CALLING; the CallerID would be able to work the other way around...

 

[robboardman]

I think if you use the switch statement instead of dial to connect to the remote dialplan all variables are passed along, Try it and report back

Robb

 

[nauliv]

Hi Robb,

Thanks for your message. I'm not familiar with the switch command... Haven't seen it in the bible TFOT.

Can you paste an example or link to the syntax ?

Thanks!

http://www.google.com/search?q=asterisk+cmd+switch

 

[robboardman]

here you go

http://www.voip-info.org/wiki/view/Asterisk+-+dual+servers

the switch statement allows you to run remote contexts within the current context

Robb