[asterisk-commits] anthonyl: branch anthonyl/8350-codec-2 r48341 - /team/anthonyl/8350-codec-2/c...

[asterisk-commits at lists.digium.com]

Author: anthonyl
Date: Wed Dec  6 17:27:39 2006
New Revision: 48341

URL: http://svn.digium.com/view/asterisk?view=rev&rev=48341
Log:
one more line of debugging

Modified:
    team/anthonyl/8350-codec-2/channels/chan_sip.c

Modified: team/anthonyl/8350-codec-2/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/anthonyl/8350-codec-2/channels/chan_sip.c?view=diff&rev=48341&r1=48340&r2=48341
==============================================================================
--- team/anthonyl/8350-codec-2/channels/chan_sip.c (original)
+++ team/anthonyl/8350-codec-2/channels/chan_sip.c Wed Dec  6 17:27:39 2006
@@ -2922,6 +2922,11 @@
 		return res;
 
 	p->callingpres = ast->cid.cid_pres;
+	ast_log(LOG_DEBUG, "sip_call: cap: %s  pref:  %s  joint:%s",
+			ast_getformatname_multiple(codec_buf, BUFSIZ, p->capability),
+			ast_getformatname_multiple(codec_buf, BUFSIZ, p->prefcodec),
+			ast_getformatname_multiple(codec_buf, BUFSIZ, p->jointcapability));
+	
 	p->jointcapability = ast_translate_available_formats(p->capability, p->prefcodec);
 	ast_log(LOG_DEBUG,"jointcapability %s\n",  ast_getformatname_multiple(codec_buf, BUFSIZ, p->jointcapability)); 	
 	/* If there are no audio formats left to offer, punt */