[asterisk-bugs] [Asterisk 0006335]: [patch] Realtime call control
noreply at bugs.digium.com
noreply at bugs.digium.com
Mon Sep 17 08:15:12 CDT 2007
A NOTE has been added to this issue.
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http://bugs.digium.com/view.php?id=6335
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Reported By: KNK
Assigned To: file
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Project: Asterisk
Issue ID: 6335
Category: Applications/app_dial
Reproducibility: N/A
Severity: feature
Priority: normal
Status: assigned
Asterisk Version: SVN
SVN Branch (only for SVN checkouts, not tarball releases): trunk
SVN Revision (number only!): 56340
Disclaimer on File?: No
Request Review:
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Date Submitted: 01-24-2006 02:45 CST
Last Modified: 09-17-2007 08:15 CDT
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Summary: [patch] Realtime call control
Description:
Adds new variables to the L option of Dial application to recheck the call
limit during the call.
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Relationships ID Summary
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related to 0007531 [patch] call not terminated after timel...
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greyvoip - 09-17-07 08:15
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I've played around with this patch and it works well. The only problem I
have with it is related to load issues. The approach used creates an extra
channel to do the real-time call control, so for example if two SIP
channels are bridged on a call the patch would create a third channel to
execute the AGI application and update the original SIP channel timeouts.
The load implications of this have me a bit worried as channels are not
light weight.
I have written a new patch based on KNK's approach that replaces the extra
channel and AGI approach with a Curl and scheduler approach. This lets a
single thread do the real-time call control for all in progress calls and
should mean the real-time call control mechanism has a negligible increase
on system load. In case it's useful I'll upload the patch once I've got
diff working properly.
Issue History
Date Modified Username Field Change
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09-17-07 08:15 greyvoip Note Added: 0070664
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