[asterisk-dev] [Code Review] Fix SLA bugs with SIP Channels (bugs 20440 and 20462)

[rmudgett]

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Please always refer to the issues by their correct id tags: ASTERISK-20440 and ASTERISK-20462.  We have not used Mantis for well over a year.  Using the correct id tags allows automatic hotlinks to work as well as commit scripts to put attributes to the correct issue.

Thanks


- rmudgett


On Jan. 14, 2013, 4:47 p.m., dkerr wrote:
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> This is an automatically generated e-mail. To reply, visit:
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> (Updated Jan. 14, 2013, 4:47 p.m.)
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> 
> Review request for Asterisk Developers, mattjordan and dkerr.
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> Summary
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> SLA has two major problems when using the feature with SIP channels (and I suspect any channel type other than analog POTS).
> 1) No ringback is presented to the calling channel.
> 2) If the calling channel hangs up before called party answers, then the called channel is not hungup.
> See bugids 20440 and 20462 for more detail.  Both are included in a single patch as they hit the exact same area of code.
> 
> Note also the addition of a 1/10th second sleep. The SLA code has a tight loop during which it is polling for status change on the outgoing channel.  The loop ends when the channel answers.  This seems like an unnecessary CPU hog and so I added the 1/10th second sleep to moderate things -- seems unnecessary to be polling any more frequently than this.  However, the sleep is not integral to fixing either of these bugs.
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> This addresses bugs 20440 and 20462.
>     https://issues.asterisk.org/jira/browse/20440
>     https://issues.asterisk.org/jira/browse/20462
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> 
> Diffs
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>   /trunk/apps/app_meetme.c 379070 
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> Diff: https://reviewboard.asterisk.org/r/2275/diff
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> Testing
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> I have had this patch running on my system for several weeks and made many calls over two different SIP trunk providers and with a GTalk channel.  No problems encountered.
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> Thanks,
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> dkerr
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>

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