[asterisk-dev] [Code Review] 2455: Add support for DTMF via SIP INFO
Joshua Colp
reviewboard at asterisk.org
Wed Apr 17 12:31:04 CDT 2013
-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/2455/#review8300
-----------------------------------------------------------
/team/group/pimp_my_sip/channels/chan_gulp.c
<https://reviewboard.asterisk.org/r/2455/#comment15970>
Pfft, handle the case where this fails :P
- Joshua Colp
On April 17, 2013, 4:36 p.m., Jason Parker wrote:
>
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/2455/
> -----------------------------------------------------------
>
> (Updated April 17, 2013, 4:36 p.m.)
>
>
> Review request for Asterisk Developers.
>
>
> Bugs: ASTERISK-21261
> https://issues.asterisk.org/jira/browse/ASTERISK-21261
>
>
> Repository: Asterisk
>
>
> Description
> -------
>
> Sending is done inside of chan_gulp, if the dtmf= setting in res_sip.conf is set to info. Receiving is done as a separate module, since that made the most sense, to me.
>
>
> Diffs
> -----
>
> /team/group/pimp_my_sip/channels/chan_gulp.c 385911
> /team/group/pimp_my_sip/res/res_sip_dtmf_info.c PRE-CREATION
>
> Diff: https://reviewboard.asterisk.org/r/2455/diff/
>
>
> Testing
> -------
>
> DTMF gets parsed properly. Outgoing packet looks okay.
>
>
> Thanks,
>
> Jason Parker
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20130417/7a92382a/attachment.htm>
More information about the asterisk-dev
mailing list