[asterisk-commits] oej: trunk r47893 - in /trunk: ./ channels/chan_sip.c

asterisk-commits at lists.digium.com asterisk-commits at lists.digium.com
Tue Nov 21 08:25:39 MST 2006


Author: oej
Date: Tue Nov 21 09:25:38 2006
New Revision: 47893

URL: http://svn.digium.com/view/asterisk?view=rev&rev=47893
Log:
Treat 101 as 100, not 183 session progress

Modified:
    trunk/   (props changed)
    trunk/channels/chan_sip.c

Propchange: trunk/
------------------------------------------------------------------------------
Binary property 'branch-1.4-blocked' - no diff available.

Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?view=diff&rev=47893&r1=47892&r2=47893
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Tue Nov 21 09:25:38 2006
@@ -558,7 +558,7 @@
 static int allow_external_domains;	/*!< Accept calls to external SIP domains? */
 static int global_callevents;		/*!< Whether we send manager events or not */
 static int global_t1min;		/*!< T1 roundtrip time minimum */
-static int global_autoframing;          /*!< ?????????? */
+static int global_autoframing;          /*!< Turn autoframing on or off. */
 static enum transfermodes global_allowtransfer;	/*!< SIP Refer restriction scheme */
 
 /*! \brief Codecs that we support by default: */
@@ -11724,7 +11724,7 @@
 	/* RFC3261 says we must treat every 1xx response (but not 100)
 	   that we don't recognize as if it was 183.
 	*/
-	if (resp > 100 && resp < 200 && resp != 180 && resp != 183)
+	if (resp > 100 && resp < 200 && resp!=101 && resp != 180 && resp != 183)
 		resp = 183;
 
  	/* Any response between 100 and 199 is PROCEEDING */
@@ -11738,6 +11738,7 @@
 
 	switch (resp) {
 	case 100:	/* Trying */
+	case 101:	/* Dialog establishment */
 		if (!ast_test_flag(req, SIP_PKT_IGNORE))
 			sip_cancel_destroy(p);
 		check_pendings(p);
@@ -12232,6 +12233,7 @@
 	} else if (ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
 		switch(resp) {
 		case 100:	/* 100 Trying */
+		case 101:	/* 101 Dialog establishment */
 			if (sipmethod == SIP_INVITE) 
 				handle_response_invite(p, resp, rest, req, seqno);
 			break;
@@ -12940,6 +12942,7 @@
 		respcode = atoi(code);
 		switch (respcode) {
 		case 100:	/* Trying: */
+		case 101:	/* dialog establishment */
 			/* Don't do anything yet */
 			break;
 		case 183:	/* Ringing: */



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