[Asterisk-Users] How to make Asterisk to generate and terminate calls

[Ravi Shankar]

After some search in wiki I was able to do what I wanted. Here is how it is,

The .call file should appear something like this and it has to be placed
in /var/spool/asterisk/outgoing of asterisk-1,

Channel: local/3001 at sip ; Any extension can be called using
local/<extension>@<context>
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context: sip
Extension: 2001
Priority: 1

In asterisk-1 we should have the following entries in extensions.conf file,

[sip]
exten => 3001,1,MyOriginateScript()
exten => 3001,2,Hangup

In asterisk-2 we should have the following entries in extensions.conf file,

[sip]
exten => 2001,1,MyTerminateScript()
exten => 2001,2,Hangup

We can do whatever we want in our MyoriginateScript/MyTerminateScript.
The features provided by asterisk is simply amazing !!! Long live asterisk

cheers,
Ravi

Ravi Shankar wrote:

> Shawn,
>   Thanks for info that would solve the problem of manually making 
> calls and connecting the phones at the either ends. But my requirement 
> is slightly different. I've the following .call file in the 
> /var/spool/asterisk/outgoing directory of asterisk-1
>
> asterisk-1 ----- SIP ----- asterisk-2
>
> Channel: SIP/3001
> MaxRetries: 2
> RetryTime: 60
> WaitTime: 30
> Context: sip
> Extension: 2222
> Priority: 1
>
> So Asterisk-1 bridges 3001 and 2222 (which is in asterisk-2). Since 
> 2222 is the terminating side I can have an AGI script handle the call 
> and do whatever I wanted and I don't need a real IP Phone. On the 
> other hand on the originating side 3001 has to be a real SIP Phone.
>
> My question is on the originating side, can a AGI script answer the 
> call instead of real IP Phone. This way I can simulate multiple IP 
> Phones without having them physically available. I know this is not 
> the intended usage of asterisk but it would serve to test bulk 
> deployments and find out the capacity of the asterisk without having 
> so many real phones.
>
> thanks,
> Ravi
>
> Shawn Porter wrote:
>
>> Ravi,
>>
>> Take a look here. http://www.voip-info.org/wiki-Asterisk+auto-dial+out
>> I would think that for what you are doing use a cron job and a shell 
>> script.
>>
>>
>> Shawn
>>
>> -----Original Message-----
>> From: asterisk-users-bounces at lists.digium.com
>> [mailto:asterisk-users-bounces at lists.digium.com]On Behalf Of Ravi
>> Shankar
>> Sent: Friday, December 23, 2005 8:41 AM
>> To: Asterisk Users
>> Subject: [Asterisk-Users] How to make Asterisk to generate and
>> terminatecalls
>>
>>
>> Hi,
>> I would like to connect two linux machines running asterisk and then
>> originate SIP calls from one asterisk and terminate it on the other
>> asterisk. Terminating the call is not a problem because I can give the
>> call handle to say AGI application on the terminating asterisk. How do i
>> originate a call from the asterisk ? Is this possible using AGI ? Any
>> pointers in this regard would be of great help.
>>
>> This type of application can be used two simulate bulk calls and find
>> out what is the maximum limit for the asterisk in terms of CPU
>> utilization, memory, etc. before it can be deployed in production
>> environment.
>>
>> thanks,
>> Ravi
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