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

[dkerr]

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https://reviewboard.asterisk.org/r/2275/
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Review request for Asterisk Developers, mattjordan and dkerr.


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.


This addresses bugs 20440 and 20462.
    https://issues.asterisk.org/jira/browse/20440
    https://issues.asterisk.org/jira/browse/20462


Diffs
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  /trunk/apps/app_meetme.c 379070 

Diff: https://reviewboard.asterisk.org/r/2275/diff


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.


Thanks,

dkerr

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