[asterisk-users] Asterisk Capacity

[Eric Wieling]

If you set the ctime (or maybe mtime) of your spool file to a date in the future, then asterisk won't process the .call file until that future time.

I recommend creating your call files with a random ctime/mtime for 0 - 240 seconds in the future and make sure you have a random retry time in your .call file.

Also NEVER create your .call files in the asterisk outgoing directory -- asterisk might try processing the file before your script finishes writing the data.  Create the file in a different directory on the same filesystem and move it to the outgoing directory.

This is not a perfect solution, but it is "good enough" and worked well at the time.

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, May 03, 2012 11:56 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Asterisk Capacity

The "best" approach to this would be to have a "sender" that uses AMI to monitor channels and release .call files as channels become available.  About 100 lines in PERL.

 

From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Ashish Agarwal
Sent: Thursday, May 03, 2012 10:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk Capacity

 

So what is a better approach to achieve this

On May 3, 2012 9:20 PM, "Mitul Limbani" <mitul at enterux.in> wrote:

The other 70 will result into failure with .call file approach.


Regards,
Mitul Limbani,
Chief Architech & Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel, 
Vidyavihar (W), Mumbai - 400 086. India
http://www.enterux.com/
http://www.entvoice.com/
email: mitul at enterux.in
DID: +91-22-61447605
Cell: +91-9820332422





On Thu, May 3, 2012 at 9:11 PM, Ashish Agarwal <ashishag67 at gmail.com> wrote:

Hello,

We are currently working on a project where using .call file on asterisk spool, outbound calls will be made from a pri line and a voice clip will be played.

We know that pri has a capacity of handling only 30 channels at a time. Therefore, my worry is what happens if we write 100 files at a time on the spool. Will asterisk manage the queue or how exactly will it behave.

Regards,

Ashish


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