Sponsor: VoiceMeUp - Corporate & Wholesale VoIP Services

VoIP Mailing List Archives
Mailing list archives for the VoIP community
 SearchSearch 

[asterisk-users] Multiple Servers: Multiple Peers: call-limit


 
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> Asterisk Users
View previous topic :: View next topic  
Author Message
b.lavallee at globalta...
Guest





PostPosted: Wed Jun 25, 2014 3:40 am    Post subject: [asterisk-users] Multiple Servers: Multiple Peers: call-limi Reply with quote

I would like to enforce call-limit across multiple servers. Is there
any way to pass a call-limit variable between servers 01 & 02, as shown
below? Use a global call-limit between multiple servers and peer
connections.

A -- 01 -- Z
A -- 02 -- Z

A is using round-robin to reach Z, but in the event that 01 or 02 fail,
I want the full call-limit available to A. The call-limit is only
applied between A and the middle servers. For the sake of discussion,
let's say call-limit=10 for both, and the total limit should also be 10.

Since my round-robin configuration will fall-back to the other server,
calls can reach a maximum of 20. Not a state I want to allow.

#server_a_extensions.conf
[SERVER01]
exten => _X.,1,NoOp(Use: First Server)
same => n,Dial(SIP/A-to-01-to-Z/${EXTEN})
same => n,GotoIf($["${DIALSTATUS}"="CHANUNAVAIL"]?SERVER02,${EXTEN},1)
same => n,NoOp(yes, it's incomplete)

[SERVER02]
exten => _X.,1,NoOp(Use: Second Server)
same => n,Dial(SIP/A-to-02-to-Z/${EXTEN})
same => n,GotoIf($["${DIALSTATUS}"="CHANUNAVAIL"]?SERVER01,${EXTEN},1)
same => n,NoOp(yes, it's incomplete)

I've though about passing the variable between the middle servers in a
SIP message, side communication channel. But, hoping there might be a
simpler solution.


Sincerely,
Brian LaVallee



--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Back to top
gmalsack at coastalacq...
Guest





PostPosted: Wed Jun 25, 2014 4:33 am    Post subject: [asterisk-users] Multiple Servers: Multiple Peers: call-limi Reply with quote

Store the call count in a shared SQL db.

Sent from my Verizon Wireless 4G LTE DROID



Brian LaVallee <b.lavallee@globaltank.jp> wrote:

I would like to enforce call-limit across multiple servers. Is there
any way to pass a call-limit variable between servers 01 & 02, as shown
below? Use a global call-limit between multiple servers and peer
connections.

A -- 01 -- Z
A -- 02 -- Z

A is using round-robin to reach Z, but in the event that 01 or 02 fail,
I want the full call-limit available to A. The call-limit is only
applied between A and the middle servers. For the sake of discussion,
let's say call-limit=10 for both, and the total limit should also be 10.

Since my round-robin configuration will fall-back to the other server,
calls can reach a maximum of 20. Not a state I want to allow.

#server_a_extensions.conf
[SERVER01]
exten => _X.,1,NoOp(Use: First Server)
same => n,Dial(SIP/A-to-01-to-Z/${EXTEN})
same => n,GotoIf($["${DIALSTATUS}"="CHANUNAVAIL"]?SERVER02,${EXTEN},1)
same => n,NoOp(yes, it's incomplete)

[SERVER02]
exten => _X.,1,NoOp(Use: Second Server)
same => n,Dial(SIP/A-to-02-to-Z/${EXTEN})
same => n,GotoIf($["${DIALSTATUS}"="CHANUNAVAIL"]?SERVER01,${EXTEN},1)
same => n,NoOp(yes, it's incomplete)

I've though about passing the variable between the middle servers in a
SIP message, side communication channel. But, hoping there might be a
simpler solution.


Sincerely,
Brian LaVallee



--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Back to top
EWieling at nyigc.com
Guest





PostPosted: Wed Jun 25, 2014 8:29 am    Post subject: [asterisk-users] Multiple Servers: Multiple Peers: call-limi Reply with quote

Something like memcachedb is also an option.  

From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Gregory Malsack
Sent: Wednesday, June 25, 2014 5:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Multiple Servers: Multiple Peers: call-limit


Store the call count in a shared SQL db.



Sent from my Verizon Wireless 4G LTE DROID




Brian LaVallee <b.lavallee@globaltank.jp (b.lavallee@globaltank.jp)> wrote:

I would like to enforce call-limit across multiple servers. Is there
any way to pass a call-limit variable between servers 01 & 02, as shown
below? Use a global call-limit between multiple servers and peer
connections.

A -- 01 -- Z
A -- 02 -- Z

A is using round-robin to reach Z, but in the event that 01 or 02 fail,
I want the full call-limit available to A. The call-limit is only
applied between A and the middle servers. For the sake of discussion,
let's say call-limit=10 for both, and the total limit should also be 10.

Since my round-robin configuration will fall-back to the other server,
calls can reach a maximum of 20. Not a state I want to allow.

#server_a_extensions.conf
[SERVER01]
exten => _X.,1,NoOp(Use: First Server)
same => n,Dial(SIP/A-to-01-to-Z/${EXTEN})
same => n,GotoIf($["${DIALSTATUS}"="CHANUNAVAIL"]?SERVER02,${EXTEN},1)
same => n,NoOp(yes, it's incomplete)

[SERVER02]
exten => _X.,1,NoOp(Use: Second Server)
same => n,Dial(SIP/A-to-02-to-Z/${EXTEN})
same => n,GotoIf($["${DIALSTATUS}"="CHANUNAVAIL"]?SERVER01,${EXTEN},1)
same => n,NoOp(yes, it's incomplete)

I've though about passing the variable between the middle servers in a
SIP message, side communication channel. But, hoping there might be a
simpler solution.


Sincerely,
Brian LaVallee



--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Back to top
Display posts from previous:   
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> Asterisk Users All times are GMT - 5 Hours
Page 1 of 1

 
Jump to:  
You cannot post new topics in this forum
You cannot reply to topics in this forum
You cannot edit your posts in this forum
You cannot delete your posts in this forum
You cannot vote in polls in this forum


Powered by phpBB © 2001, 2005 phpBB Group

VoiceMeUp - Corporate & Wholesale VoIP Services