[asterisk-dev] [Code Review] 4316: ARI/PJSIP: Add the ability to redirect (transfer) a channel in a Stasis dialplan application to another system; improve and fix PJSIP's transfer ability
Joshua Colp
reviewboard at asterisk.org
Mon Jan 19 10:25:43 CST 2015
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/branches/13/channels/chan_pjsip.c
<https://reviewboard.asterisk.org/r/4316/#comment24682>
I think it may be useful to have AOR support here as well.
/branches/13/include/asterisk/stasis_app.h
<https://reviewboard.asterisk.org/r/4316/#comment24681>
Everything else has doxygen. Why not this?
/branches/13/res/res_pjsip_multihomed.c
<https://reviewboard.asterisk.org/r/4316/#comment24683>
There's no guarantee that the host will be NULL terminated.
/branches/13/res/res_pjsip_nat.c
<https://reviewboard.asterisk.org/r/4316/#comment24684>
There's no guarantee that host will be NULL terminated.
/branches/13/res/res_pjsip_transport_websocket.c
<https://reviewboard.asterisk.org/r/4316/#comment24685>
Etc.
- Joshua Colp
On Jan. 19, 2015, 3:16 a.m., Matt Jordan wrote:
>
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> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/4316/
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>
> (Updated Jan. 19, 2015, 3:16 a.m.)
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>
> Review request for Asterisk Developers and Joshua Colp.
>
>
> Bugs: ASTERISK-24015 and ASTERISK-24703
> https://issues.asterisk.org/jira/browse/ASTERISK-24015
> https://issues.asterisk.org/jira/browse/ASTERISK-24703
>
>
> Repository: Asterisk
>
>
> Description
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>
> This patch adds a new feature to ARI to redirect a channel to another server, and fixes a few bugs in PJSIP's handling of the Transfer dialplan application/ARI redirect capability.
>
> *New Feature*
> A new operation has been added to the ARI channels resource, redirect. With this, a channel in a Stasis application can be redirected to another endpoint of the same underlying channel technology.
>
> - Preemptive question: why 'redirect', and not 'transfer'? Mostly because 'transfer' was always kind of a bad name. If the channel isn't answered, we aren't transferring, we're forwarding. If it is answered, the type of transfer being performed is somewhat vague - is it blind? Is it attended? 'redirect' - while also a slightly loaded term - is a bit more generic and yet descriptive of what is happening: we're redirecting the channel to somewhere else. Answered, not answered, it doesn't matter: your channel is no good here!
>
> *Bug fixes*
> In the process of writing this new feature, two bugs were fixed in the PJSIP stack:
> (1) The existing .transfer channel callback had the limitation that it could only transfer channels to a SIP URI, i.e., you had to pass 'PJSIP/sip:foo at my_provider.com' to the dialplan application. While this is still supported, it is somewhat unintuitive - particularly in a world full of endpoints. As such, we now also support specifying the PJSIP endpoint to transfer to.
> (2) res_pjsip_multihomed was, unfortunately, trying to 'help' a 302 redirect by updating its Contact header. Alas, that resulted in the forwarding destination set by the dialplan application/ARI resource/whatever being rewritten with very incorrect information. Hence, we now don't bother updating an outgoing response if it is a 302. Since this took a looong time to find, some additional debug statements have been added to those modules that update the Contact headers.
>
>
> Diffs
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>
> /branches/13/rest-api/api-docs/channels.json 430751
> /branches/13/res/stasis/control.c 430751
> /branches/13/res/res_pjsip_transport_websocket.c 430751
> /branches/13/res/res_pjsip_nat.c 430751
> /branches/13/res/res_pjsip_multihomed.c 430751
> /branches/13/res/res_ari_channels.c 430751
> /branches/13/res/ari/resource_channels.c 430751
> /branches/13/res/ari/resource_channels.h 430751
> /branches/13/include/asterisk/stasis_app.h 430751
> /branches/13/channels/chan_pjsip.c 430751
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> Diff: https://reviewboard.asterisk.org/r/4316/diff/
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>
> Testing
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> Tests were written both for the PJSIP stack as well as the new ARI operation. See https://reviewboard.asterisk.org/r/4352.
>
>
> Thanks,
>
> Matt Jordan
>
>
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