[asterisk-commits] file: branch file/gulp_fax r394465 - /team/file/gulp_fax/res/res_sip_session.c

[SVN commits to the Asterisk project]

Author: file
Date: Tue Jul 16 13:04:28 2013
New Revision: 394465

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=394465
Log:
Allow media to be changed in the SDP so the negotiator does not add a dummy audio stream in the first slot.

Modified:
    team/file/gulp_fax/res/res_sip_session.c

Modified: team/file/gulp_fax/res/res_sip_session.c
URL: http://svnview.digium.com/svn/asterisk/team/file/gulp_fax/res/res_sip_session.c?view=diff&rev=394465&r1=394464&r2=394465
==============================================================================
--- team/file/gulp_fax/res/res_sip_session.c (original)
+++ team/file/gulp_fax/res/res_sip_session.c Tue Jul 16 13:04:28 2013
@@ -1230,6 +1230,7 @@
 		pjsip_dlg_terminate(dlg);
 		return NULL;
 	}
+	inv_session->sdp_neg_flags = PJMEDIA_SDP_NEG_ALLOW_MEDIA_CHANGE;
 
 	pjsip_timer_setting_default(&timer);
 	timer.min_se = endpoint->min_se;
@@ -1371,6 +1372,7 @@
 		pjsip_dlg_terminate(dlg);
 		return NULL;
 	}
+	inv_session->sdp_neg_flags = PJMEDIA_SDP_NEG_ALLOW_MEDIA_CHANGE;
 	if (pjsip_dlg_add_usage(dlg, &session_module, NULL) != PJ_SUCCESS) {
 		if (pjsip_inv_initial_answer(inv_session, rdata, 500, NULL, NULL, &tdata) != PJ_SUCCESS) {
 			pjsip_inv_terminate(inv_session, 500, PJ_FALSE);