[Asterisk-code-review] multicast RTP: Add dialing options (asterisk[master])

[Mark Michelson]

Mark Michelson has posted comments on this change.

Change subject: multicast RTP: Add dialing options
......................................................................


Patch Set 2:

(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 wi
It has to be done after the options are parsed since the options may have some bearing on what format is chosen. I can't move it back to where it used to be, but I should be able to move it to above the instance/channel allocations.


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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: Mark Michelson <mmichelson at digium.com>
Gerrit-Reviewer: Richard Mudgett <rmudgett at digium.com>
Gerrit-HasComments: Yes