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Diffstat (limited to 'net/asterisk-bristuff/files/patch-channels::chan_sip.c')
-rw-r--r--net/asterisk-bristuff/files/patch-channels::chan_sip.c22
1 files changed, 9 insertions, 13 deletions
diff --git a/net/asterisk-bristuff/files/patch-channels::chan_sip.c b/net/asterisk-bristuff/files/patch-channels::chan_sip.c
index 2b2fae7a311f..d1f186aba8d7 100644
--- a/net/asterisk-bristuff/files/patch-channels::chan_sip.c
+++ b/net/asterisk-bristuff/files/patch-channels::chan_sip.c
@@ -1,9 +1,6 @@
-
-$FreeBSD$
-
---- channels/chan_sip.c.orig Tue Jan 24 16:25:53 2006
-+++ channels/chan_sip.c Tue Jan 24 16:30:44 2006
-@@ -337,7 +337,7 @@ static char global_vmexten[AST_MAX_EXTEN
+--- channels/chan_sip.c.orig Sat Jun 3 21:41:59 2006
++++ channels/chan_sip.c Sat Jun 3 21:43:40 2006
+@@ -337,7 +337,7 @@
static char default_language[MAX_LANGUAGE] = "";
@@ -12,7 +9,7 @@ $FreeBSD$
static char default_callerid[AST_MAX_EXTENSION] = DEFAULT_CALLERID;
static char default_fromdomain[AST_MAX_EXTENSION] = "";
-@@ -475,6 +475,7 @@ struct sip_invite_param {
+@@ -475,6 +475,7 @@
struct sip_route {
struct sip_route *next;
@@ -20,7 +17,7 @@ $FreeBSD$
char hop[0];
};
-@@ -6015,6 +6016,7 @@ static void build_route(struct sip_pvt *
+@@ -6048,6 +6049,7 @@
/* Make a struct route */
thishop = malloc(sizeof(*thishop) + len);
if (thishop) {
@@ -28,7 +25,7 @@ $FreeBSD$
ast_copy_string(thishop->hop, rr, len);
ast_log(LOG_DEBUG, "build_route: Record-Route hop: <%s>\n", thishop->hop);
/* Link in */
-@@ -6040,31 +6042,41 @@ static void build_route(struct sip_pvt *
+@@ -6073,31 +6075,41 @@
/* Only append the contact if we are dealing with a strict router */
if (!head || (!ast_strlen_zero(head->hop) && strstr(head->hop,";lr") == NULL) ) {
@@ -94,7 +91,7 @@ $FreeBSD$
}
}
}
-@@ -10352,7 +10364,11 @@ static int handle_request_invite(struct
+@@ -10440,7 +10452,11 @@
gotdest = get_destination(p, NULL);
get_rdnis(p, NULL);
@@ -107,7 +104,7 @@ $FreeBSD$
build_contact(p);
if (gotdest) {
-@@ -10380,7 +10396,6 @@ static int handle_request_invite(struct
+@@ -10468,7 +10484,6 @@
c = sip_new(p, AST_STATE_DOWN, ast_strlen_zero(p->username) ? NULL : p->username );
*recount = 1;
/* Save Record-Route for any later requests we make on this dialogue */
@@ -115,11 +112,10 @@ $FreeBSD$
if (c) {
/* Pre-lock the call */
ast_mutex_lock(&c->lock);
-@@ -10466,7 +10481,12 @@ static int handle_request_invite(struct
+@@ -10554,6 +10569,12 @@
transmit_response(p, "180 Ringing", req);
break;
case AST_STATE_UP:
-- /* Here we have reINVITE request - try to renegotiate codecs with */
+ /* Assuming this to be reinvite, process new SDP portion */
+ if (!ast_strlen_zero(get_header(req, "Content-Type"))) {
+ process_sdp(p, req);