[Asterisk-Users] Asterisk Manager Interface notes

Scott Stingel scott at evtmedia.com
Thu Jan 29 14:30:27 MST 2004


Hi Matt-

Thanks for posting all of that!  I was just starting to look into using this
interface, and now maybe have some second thoughts after reading your post.
You're right, it would be great for someone to fix the buffer/deadlock
problems.

Cheers
Scott Stingel

Scott M. Stingel 
Emerging Voice Technology Inc.
Palo Alto, California and London, England

Email:          scott at evtmedia.com  
URL:            www.evtmedia.com  



-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of mattf
Sent: Thursday, January 29, 2004 7:11 AM
To: 'asterisk-users at lists.digium.com'
Subject: [Asterisk-Users] Asterisk Manager Interface notes


Hello,

After battling with the Asterisk Manager interface(and getting it to pretty
much do everything I want to do with it) I thought I'd share my experiences
with those who are developing or are thinking of developing applications
using it.

First here's a list of some of the things the manager interface will let you
do:
- Dial a call from any extension/resource to any other extension/resource,
some examples:
	- initiate a call from a sip phone to external zap line
	- initiate a call from an extension number to an external iax
extension
	- initiate a call from within a meetme room to a zap channel
- Redirect ANY live call to ANY destination, some examples:
	- take a live SIP call from whereever it is connectedand sed it to a
meetme room
	- take a zap call from a meetme room and send it to an extension
	- take a ANY call and redirect it to an outside Zap channel
	- As an alternative to parking, just dump a call to a long
background sound file or MoH and then retrieve it later by redirecting their
channel somewhere else
- Hangup any live channel whether it be SIP, Zap, IAX, H323, etc...
	- Hangup individual channels within a meetme conference
- Initiate a recording(and stop it too) of any live Zap channel at any
time(with custom filename)
- Get the status of voicemail in any mailbox
- Get the status of an extension
- Get data from commands such as "show channels"
- Get data on Queues
- Get data on IAX peers

	
Second, here are some things you should know if you are going to program for
the manager interface:

- The manager API is not well documented(Yes I know I need to add my notes
to the wiki)
- All connected terminals will receive all "Events" that happen on the
Asterisk box
- Not all Asterisk commands will be accepted with the "Action: Command"
action
- Disconnecting the connection between a remote connected terminal and the
Asterisk box will often cause a deadlock
(http://bugs.digium.com/bug_view_page.php?bug_id=0000861)
- Any form of freeze on the connected terminal will cause a buffer overflow
and will also deadlock Asterisk
- "Status" Actions can yield upto a hundred lines of output depending on how
busy your Asterisk machine is.
- "Ping" and "Show uptime" may not return results in some applications(like
perl Net::Telnet)
- generally for applications where you will be sending under a hundred
commands daily, the manager interface will work well and shouldn't cause any
crashes/deadlocks.
- for applications where you could be sending over a thousand manager
commands to an Asterisk server from different client machines daily you will
almost definitely have at least one crash/deadlock happen per day.
- you can connect to the asterisk manager interface through any programming
language that has a socket IO implementation(C, Python, Perl, PHP, etc...)
- you can program an AGI to use the manager even(it you really wanted to)


How I solved my crash problem:

I had a problem, I was using the manager interface through over 30 desktop
machines to do a high number of redirect, originate, recording, hangup and
command actions to the tune of over five thousand commands per day. This
lead to upto 4 crashes/deadlocks per day on one of my Asterisk servers.
That's when I started to look hard at a centralized manager queue. After
some initial testing I determined that the best way for me to process all of
these actions was to have a database driven system by which two constantly
running scripts would separately send and parse manager actions and events.
Here's how the steps proceed on an Originate command:

1. client inserts an Originate record into the queue table (with a unique
Callerid value)
2. action_sender application grabs the new record, submits the action and
marks the record as SENT
3. manager_listen application parses every line of manager output and sends
blind UPDATE commands to the database based on the action CallerID field
and/or the uniqueid field for a key
	- in the manager interface the callerid field can be unique to the
call as sent into the manager
4. client can grab the uniqueid and channel values out of the database now
that the record has been processed

Under this process there is no risk of losing a manager connection on the
client machine, all manager connections exist on the localhost Asterisk
machine. Also, there is very little lag in processing actions through this
model even on a busy machine.


My suggestions for improving the manager interface:

- make it more fault tolerant, I can live with the querky API and data
formatting, but buffer overflows and not killing inactive connections
causing crashes/deadlocks is VERY bad.
- make a simple manager action file parser(sample.action), something like
the sample.call interface except for manager actions, for output you can
have the manager generate a sampleaction.out file that would have the output
for that specific command on it. Many people that currently use the
sample.call format would love to have a simple way to add manager functions
to their apps that already generate sample.call files
- make a transaction-based send-receive protocol, something like the HTTP
protocol. This would be a lot more involved than the sample.action idea, but
it is probably a step in the right direction for the future of Asterisk.
(this one isn't my idea, "jayson" is working on
this[http://bugs.digium.com/bug_view_page.php?bug_id=0000123])


Well, that's the end of my rambling for now, hope this helps.

MATT---
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