[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.
-- 
To view, visit https://gerrit.asterisk.org/2910
To unsubscribe, visit https://gerrit.asterisk.org/settings
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
    
    
More information about the asterisk-code-review
mailing list