[asterisk-bugs] [Asterisk 0012544]: Congestion feature request

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=12544 
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Reported By:                kactus
Assigned To:                
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Project:                    Asterisk
Issue ID:                   12544
Category:                   Channels/NewFeature
Reproducibility:            N/A
Severity:                   feature
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.6.0-beta7.1 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             04-28-2008 21:45 CDT
Last Modified:              04-30-2008 10:24 CDT
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Summary:                    Congestion feature request
Description: 
Hi

I've been looking at a few different ways to provide failover for clients
and was hoping that there was a way to set auto_congest timeout values for
non registered peers.

This would allow for dial plans such as

1,Answer()
2,Dial(Exten at peer1)
3,Dial(Exten at peer2)
etc

Autocongestion currently works if the peer returns an error but if the
remote host is down it waits the full 32 seconds before progressing to the
next priority. I'd like to set this so that if there is no response after 5
seconds that it goes to the next priority. (that way the cheaper but less
reliable trunks are used first)

It appears asterisk 1.6 no longer uses the qualify column to determine
maximum time before failover, and the timerb in sip.conf for example only
works on registered peers. Passing time in the dial only seems to work once
the other side receives it and setting timeout absolute = 5 redials the
same priority every 5 seconds.

If this configuration has been moved somewhere else already can you please
let me know as I was unable to find it in the documentation, online or
after greping the source code.

Thanks
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---------------------------------------------------------------------- 
 oej - 04-30-08 10:24  
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Why don't you use the SIPPEER dialplan function BEFORE you place the call
to check the status of the peer? That way, you don't even have to place the
call. 

Issue History 
Date Modified   Username       Field                    Change               
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04-30-08 10:24  oej            Note Added: 0086202                          
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