[asterisk-dev] [Code Review] 3540: chan_local+app_dial: Propagagate call answered elsewhere over local channels.

Jared Smith reviewboard at asterisk.org
Wed May 14 10:46:52 CDT 2014


-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/3540/#review11888
-----------------------------------------------------------

Ship it!


Looks simple and straightforward to me.

- Jared Smith


On May 14, 2014, 3:19 p.m., wdoekes wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/3540/
> -----------------------------------------------------------
> 
> (Updated May 14, 2014, 3:19 p.m.)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Repository: Asterisk
> 
> 
> Description
> -------
> 
> When dialing SIP/account_a + SIP/account_b, and account_b picks up, chan_sip sends
> out a Reason header with SIP;cause=200;text="Call completed elsewhere", signifying
> that the call was picked up.
> 
> The SIP phone then does not show "1 missed call".
> 
> However, then dialing Local/account_a + Local/account_b, this does not work.
> 
> This review addresses that.
> 
> 
> When hanging up obsolete channels in chan_local, the answered_elsewhere flag is
> propagated to cancelled (parent) channel using the hangupcause.
> 
> In app_dial, this hangupcause is checked and passed down to the other calls to be
> cancelled.
> 
> 
> Diffs
> -----
> 
>   /branches/1.8/channels/chan_local.c 413892 
>   /branches/1.8/apps/app_dial.c 413892 
> 
> Diff: https://reviewboard.asterisk.org/r/3540/diff/
> 
> 
> Testing
> -------
> 
> Dialplan:
> 
>     [default]
>     exten => 201,1,Dial(SIP/account_b&SIP/account_c,5)
>     exten => 202,1,Dial(Local/b at dial&Local/c at dial,5)
>     ;; also tested with /n for no-local-channel-optimization, behaves the same as without
>     
>     [dial]
>     exten => b,1,Dial(SIP/account_b)
>     exten => c,1,Dial(SIP/account_c)
> 
> sip.conf held 3 accounts: account_a, account_b and account_c.
> 
> 
> Before patch:
> 
>                           201             202 <-- account_a calls these
>               +---------------+---------------+
>     timeout   | 1 missed call | 1 missed call |
>               +---------------+---------------+         
>     account_b |               | 1 missed call | <-- account_c sees these
>     picks up  +---------------+---------------+         
> 
> 
> After patch:
> 
>                           201             202 <-- account_a calls these
>               +---------------+---------------+
>     timeout   | 1 missed call | 1 missed call |
>               +---------------+---------------+         
>     account_b |               |               | <-- account_c sees these
>     picks up  +---------------+---------------+         
> 
> 
> Thanks,
> 
> wdoekes
> 
>

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20140514/6e814fde/attachment-0001.html>


More information about the asterisk-dev mailing list