Review Board 1.7.16


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

Review Request #4316 - Created Jan. 18, 2015 and submitted

Matt Jordan
13
ASTERISK-24015, ASTERISK-24703
Reviewers
asterisk-dev
file
Asterisk
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@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.
Tests were written both for the PJSIP stack as well as the new ARI operation. See https://reviewboard.asterisk.org/r/4352.
Total:
6
Open:
0
Resolved:
5
Dropped:
1
Status:
From:
Review request changed
Updated (Feb. 12, 2015, 3:25 p.m.)
  • changed from pending to submitted
Committed in revision 431733

https://reviewboard.asterisk.org/ runs on a server provided by Digium, Inc. and uses bandwidth donated to the open source Asterisk community by API Digital Communications in Huntsville, AL USA.
Please report problems with this site to asteriskteam@digium.com.