[Asterisk-code-review] multicast RTP: Add dialing options (asterisk[master])
Richard Mudgett
asteriskteam at digium.com
Fri May 27 10:15:12 CDT 2016
Richard Mudgett has posted comments on this change.
Change subject: multicast RTP: Add dialing options
......................................................................
Patch Set 2: Code-Review-1
(1 comment)
https://gerrit.asterisk.org/#/c/2910/2/channels/chan_rtp.c
File channels/chan_rtp.c:
PS2, Line 198: fmt = ast_multicast_rtp_options_get_format(mcast_options);
: if (!fmt) {
: fmt = ast_format_cap_get_format(cap, 0);
: }
:
: if (!fmt) {
: ast_log(LOG_ERROR, "No format available for sending RTP to '%s'\n",
: args.destination);
: goto failure;
: }
:
Put the fmt code up where the check was previously. The error goto here will leak instance and chan.
--
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Gerrit-MessageType: comment
Gerrit-Change-Id: I033b706b533f0aa635c342eb738e0bcefa07e219
Gerrit-PatchSet: 2
Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-Owner: Mark Michelson <mmichelson at digium.com>
Gerrit-Reviewer: Anonymous Coward #1000019
Gerrit-Reviewer: Richard Mudgett <rmudgett at digium.com>
Gerrit-HasComments: Yes
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