[Asterisk-code-review] chan_pjsip: Transfer failure sip result passed through channel: Trans... (asterisk[16])

Dan Cropp asteriskteam at digium.com
Fri Jan 22 09:40:04 CST 2021


Dan Cropp has posted comments on this change. ( https://gerrit.asterisk.org/c/asterisk/+/15368 )

Change subject: chan_pjsip: Transfer failure sip result passed through channel: Transfer parameter returns the sip protocol error code control: Redirect support additional transfer protocol parameter app_transfer: TRANSFERSTATUSPROTOCOL channel variable addition
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Patch Set 1:

> Patch Set 1:
> 
> Just a heads up (I haven't reviewed this fully) but this is not suitable for inclusion into 16 or 18 as-is. This changes the ast_transfer API in a breaking fashion.

Thank you Joshua.

I believe I modified everywhere ast_transfer was called to use the additional parameter.  Is it unacceptable to do this?

I believe I could make it work without the additional parameter.  It would require setting the return value to the error. Then, in code that calls ast_transfer, it would need to process the result (checking for anything 300+).  Basically, setting it's process error value and change the original value to AST_SUCCESS/AST_FAILURE.  Would this be a more acceptable approach (provided it's well documented)?


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Gerrit-Project: asterisk
Gerrit-Branch: 16
Gerrit-Change-Id: Ia6a94784b4925628af122409cdd733c9f29abfc4
Gerrit-Change-Number: 15368
Gerrit-PatchSet: 1
Gerrit-Owner: Dan Cropp <dan at amtelco.com>
Gerrit-Reviewer: Friendly Automation
Gerrit-CC: Joshua Colp <jcolp at sangoma.com>
Gerrit-Comment-Date: Fri, 22 Jan 2021 15:40:04 +0000
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