[asterisk-bugs] [Asterisk 0014592]: [patch] export the SIP peer username of the transferer
Asterisk Bug Tracker
noreply at bugs.digium.com
Mon Sep 21 12:53:42 CDT 2009
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=14592
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Reported By: klaus3000
Assigned To:
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Project: Asterisk
Issue ID: 14592
Category: Channels/chan_sip/NewFeature
Reproducibility: have not tried
Severity: feature
Priority: normal
Status: ready for testing
Asterisk Version: SVN
JIRA:
Regression: No
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
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Date Submitted: 2009-03-03 09:42 CST
Last Modified: 2009-09-21 12:53 CDT
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Summary: [patch] export the SIP peer username of the
transferer
Description:
Hi!
When a blind transfer is initiated from a SIP, often the name of the
transferer is needed (e.g. billing, applying restrictions ...). Currently
there is a variable called SIPTRANSFER_REFERER which contains the value of
the Refered-By header - but this variable is not trustworthy, as the SIP
client can put anything into this variable.
Attached patch adds the variables SIPTRANSFERER_PEERNAME and
SIPTRANSFERER_USERNAME which contain the respective SIP username (peer or
user)
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(0111092) kaii (reporter) - 2009-09-21 12:53
https://issues.asterisk.org/view.php?id=14592#c111092
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In the discussion on IRC with oej and others, we made clear that there is a
difference between "call transfer" and "call forward".
TRANSFER requests are (at least in channel SIP) handled inside the sip
channel module. a variable BLINDTRANSFER is already being set when a SIP
REPLACE occurs.
this is not the case for attended transfers, but i think scenarios are
rare where the dialplan needs information about who did an attended
transfer _after_ a call is completed. (thus rendering an "ATTENDEDTRANSFER"
variable unneccessary)
FORWARD requests are raised from chan_sip into the core and the forward is
actually processed in app_dial. in this case, no variable is set to
indicate who actually transfered the call -- a valuable information in some
scenarios.
i started the discussion on IRC asking if it was a bug that
"BLINDTRANSFER" was not set when a call forward occured. it is set on
REPLACE, but not on REDIR events..
i was then told the difference between TRANSFER and FORWARD and we
discussed about other solutions to retrieve this information from the
dialplan.
oej came up with a few ideas to solve this in dialplan, but none worked
for me..
i totally agree on the need of such a variable. i patched this in 1.2 and
1.4 for our own needs.
i just want to add a few points to think about in the discussion:
* a call can be forwarded multiple times: SIP/10 -> SIP/20 -> SIP/30 ...
i suspect that in your solution, as in mine too, the dialplan only
recognizes a forward from SIP/20 to SIP/30, but not that SIP/10 was
initially forwarding.
* as forward requests are raised through the core and are handled inside
the app itself (dial), i see this feature independent of channels. i
implemented it inside app_dial and it works like a charm. (and i suspect
it will work with other channels than chan_sip, too)
Issue History
Date Modified Username Field Change
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2009-09-21 12:53 kaii Note Added: 0111092
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