[asterisk-dev] [Code Review] Trunk implementation of fix for issue 12434
David Vossel
dvossel at digium.com
Fri Jul 17 15:57:47 CDT 2009
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https://reviewboard.asterisk.org/r/313/#review988
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/trunk/channels/chan_sip.c
<https://reviewboard.asterisk.org/r/313/#comment2341>
update this to reflect the additions in Trunk
/trunk/channels/chan_sip.c
<https://reviewboard.asterisk.org/r/313/#comment2342>
"Hi?" is confusing. I don't know how to respond. Maybe change this to "Hi, how are you?" to clarify the question mark.
- David
On 2009-07-17 14:25:54, Mark Michelson wrote:
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> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/313/
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> (Updated 2009-07-17 14:25:54)
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> Review request for Asterisk Developers.
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> Summary
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> As the summary suggests, this is the trunk implementation of the fix for issue 12434. There is not much difference between this review request and https://reviewboard.asterisk.org/r/311/ . The two main differences are that trunk supports text streams, whereas 1.4 does not, and trunk will respond with a 0 port T.38 answer even with udptl_pt enabled if an accompanying audio stream is offered.
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> This addresses bug 12434.
> https://issues.asterisk.org/view.php?id=12434
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> Diffs
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> /trunk/channels/chan_sip.c 206453
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> Diff: https://reviewboard.asterisk.org/r/313/diff
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> Testing
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> I used the same two sipp scenarios I used in review 311. I also set up a third scenario where audio and text were offered to be sure that the SDP response is appropriate.
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> Thanks,
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> Mark
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>
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