[asterisk-commits] file: branch group/pimp_my_sip r379428 - /team/group/pimp_my_sip/res/

[SVN commits to the Asterisk project]

Author: file
Date: Fri Jan 18 08:04:05 2013
New Revision: 379428

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=379428
Log:
Properly initialize the codecs structure for use.

Modified:
    team/group/pimp_my_sip/res/res_sip_sdp_audio.c

Modified: team/group/pimp_my_sip/res/res_sip_sdp_audio.c
URL: http://svnview.digium.com/svn/asterisk/team/group/pimp_my_sip/res/res_sip_sdp_audio.c?view=diff&rev=379428&r1=379427&r2=379428
==============================================================================
--- team/group/pimp_my_sip/res/res_sip_sdp_audio.c (original)
+++ team/group/pimp_my_sip/res/res_sip_sdp_audio.c Fri Jan 18 08:04:05 2013
@@ -135,6 +135,8 @@
 	ast_sockaddr_set_port(addrs, stream->desc.port);
 	ast_rtp_instance_set_remote_address(session->media.audio, addrs);
 	ast_free(addrs);
+
+	ast_rtp_codecs_payloads_initialize(&codecs);
 
 	/* Iterate through provided formats */
 	for (format = 0; format < stream->desc.fmt_count; format++) {