> On Dec. 19, 2014, 3:59 p.m., Scott Griepentrog wrote:
> > http://svn.asterisk.org/svn/asterisk/branches/11/channels/chan_sip.c, lines
> > 11834-11842
> > <https://reviewboard.asterisk.org/r/4279/diff/1/?file=69816#file69816line11834>
> >
> > I would recommend appending to the comment block something like this:
> >
> > ASTERISK-24628: Send UPDATE to the same destination as CANCEL
OK. I think in the comment is a little typo ("unless this is a CANCEL _on_ an
ACK.."). I think, it should have been "unless this is a CANCEL _or_ an ACK..".
At least this is, what the code is doing. Should I fix this, when adding the
new comment?
> On Dec. 19, 2014, 3:59 p.m., Scott Griepentrog wrote:
> > http://svn.asterisk.org/svn/asterisk/branches/11/channels/chan_sip.c, lines
> > 11840-11842
> > <https://reviewboard.asterisk.org/r/4279/diff/1/?file=69816#file69816line11840>
> >
> > Purely for readability, the sipmethod == SIP_UPDATE condition should be
> > indented to the same level as SIP_ACK, since it is similarly nested under
> > the same negation started on the line containing SIP_CANCEL.
OK. Shall I update my patch?
- kwemheuer
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On Dec. 17, 2014, 6:11 p.m., kwemheuer wrote:
>
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/4279/
> -----------------------------------------------------------
>
> (Updated Dec. 17, 2014, 6:11 p.m.)
>
>
> Review request for Asterisk Developers.
>
>
> Bugs: ASTERISK-24628
> https://issues.asterisk.org/jira/browse/ASTERISK-24628
>
>
> Repository: Asterisk
>
>
> Description
> -------
>
> Given three SIP phones (A, B, C), all communicating via a proxy (in this case
> OpenSIPs) with asterisk. A call is established between A and B. B sets the
> call on hold (A is hearing MOH) and dials the extension of C. While phone C
> is ringing, B transfers the call (A is hearing ringing of phone C, B is
> idle). On SIP there is a REFER to transfer the call. In case "sendrpid=yes"
> asterisk sends an UPDATE to phone C. This update is sent directly (not
> through the proxy). If someone (e.g. B) is trying to get the call back
> (through a directed pickup), asterisk sends a CANCEL to C. But this CANCEL is
> sent directly to C not through the proxy. The phone ignores this CANCEL
> according to the RFC3261 (Section 9.1).
>
> In other situations asterisk ensures, that a CANCEL is sent the same way the
> INVITE has been sent. In this case the previously sent UPDATE overwrites the
> address used later for the CANCEL. On the other hand the UPDATE has to be
> sent via the proxy too (IMHO). The call is in the ringing state and other
> endpoints maybe have to been updated too.
>
> The patch solves the issue. In function reqprep the UPDATE is handled the
> same way the CANCEL was (in case the state is in ringing state).
>
>
> Diffs
> -----
>
> http://svn.asterisk.org/svn/asterisk/branches/11/channels/chan_sip.c 429698
>
> Diff: https://reviewboard.asterisk.org/r/4279/diff/
>
>
> Testing
> -------
>
> Patch is tested in the described scenario and solves the issue. Patched
> against 11.15.0.
>
>
> Thanks,
>
> kwemheuer
>
>
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