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Add another local patch that will be submitted to the vendor.

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This commit is contained in:
Maxim Sobolev 2006-12-05 11:17:04 +00:00
parent aab36845be
commit 1ff992b93c
Notes: svn2git 2021-03-31 03:12:20 +00:00 
svn path=/head/; revision=178929
5 changed files with 110 additions and 50 deletions
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32
net/asterisk/files/patch-channels::chan_sip.c
View file

@ -1,6 +1,9 @@
--- channels/chan_sip.c.orig Fri Nov 17 15:51:56 2006
+++ channels/chan_sip.c Fri Nov 17 15:53:30 2006
@@ -340,7 +340,7 @@ static char global_vmexten[AST_MAX_EXTEN
$FreeBSD$
--- channels/chan_sip.c.orig
+++ channels/chan_sip.c
@@ -340,7 +340,7 @@
static char default_language[MAX_LANGUAGE] = "";
@ -9,7 +12,7 @@
static char default_callerid[AST_MAX_EXTENSION] = DEFAULT_CALLERID;
static char default_fromdomain[AST_MAX_EXTENSION] = "";
@@ -483,6 +483,7 @@ struct sip_invite_param {
@@ -483,6 +483,7 @@
struct sip_route {
struct sip_route *next;
@ -17,7 +20,16 @@
char hop[0];
};
@@ -6222,6 +6223,7 @@ static void build_route(struct sip_pvt *
@@ -2815,6 +2816,8 @@
ast_codec_pref_remove2(&tmp->nativeformats, ~i->usercapability);
fmt = ast_codec_pref_index_audio(&tmp->nativeformats, 0);
+ pbx_builtin_setvar_helper(tmp, "SIP_CODEC_USED", ast_getformatname(fmt));
+
if (title)
snprintf(tmp->name, sizeof(tmp->name), "SIP/%s-%08x", title, (int)(long) i);
else if (strchr(i->fromdomain,':'))
@@ -6222,6 +6225,7 @@
/* Make a struct route */
thishop = malloc(sizeof(*thishop) + len);
if (thishop) {
@ -25,7 +37,7 @@
ast_copy_string(thishop->hop, rr, len);
ast_log(LOG_DEBUG, "build_route: Record-Route hop: <%s>\n", thishop->hop);
/* Link in */
@@ -6247,31 +6249,41 @@ static void build_route(struct sip_pvt *
@@ -6247,31 +6251,41 @@
/* Only append the contact if we are dealing with a strict router */
if (!head || (!ast_strlen_zero(head->hop) && strstr(head->hop,";lr") == NULL) ) {
@ -91,7 +103,7 @@
}
}
}
@@ -9248,6 +9260,13 @@ static int build_reply_digest(struct sip
@@ -9248,6 +9262,13 @@
secret = p->peersecret;
md5secret = p->peermd5secret;
}
@ -105,7 +117,7 @@
if (ast_strlen_zero(username)) /* We have no authentication */
return -1;
@@ -10621,7 +10640,11 @@ static int handle_request_invite(struct
@@ -10621,7 +10642,11 @@
gotdest = get_destination(p, NULL);
get_rdnis(p, NULL);
@ -118,7 +130,7 @@
build_contact(p);
if (gotdest) {
@@ -10649,7 +10672,6 @@ static int handle_request_invite(struct
@@ -10649,7 +10674,6 @@
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 */
@ -126,7 +138,7 @@
if (c) {
/* Pre-lock the call */
ast_mutex_lock(&c->lock);
@@ -10735,7 +10757,12 @@ static int handle_request_invite(struct
@@ -10735,7 +10759,12 @@
transmit_response(p, "180 Ringing", req);
break;
case AST_STATE_UP:

32
net/asterisk10/files/patch-channels::chan_sip.c
View file

@ -1,6 +1,9 @@
--- channels/chan_sip.c.orig Fri Nov 17 15:51:56 2006
+++ channels/chan_sip.c Fri Nov 17 15:53:30 2006
@@ -340,7 +340,7 @@ static char global_vmexten[AST_MAX_EXTEN
$FreeBSD$
--- channels/chan_sip.c.orig
+++ channels/chan_sip.c
@@ -340,7 +340,7 @@
static char default_language[MAX_LANGUAGE] = "";
@ -9,7 +12,7 @@
static char default_callerid[AST_MAX_EXTENSION] = DEFAULT_CALLERID;
static char default_fromdomain[AST_MAX_EXTENSION] = "";
@@ -483,6 +483,7 @@ struct sip_invite_param {
@@ -483,6 +483,7 @@
struct sip_route {
struct sip_route *next;
@ -17,7 +20,16 @@
char hop[0];
};
@@ -6222,6 +6223,7 @@ static void build_route(struct sip_pvt *
@@ -2815,6 +2816,8 @@
ast_codec_pref_remove2(&tmp->nativeformats, ~i->usercapability);
fmt = ast_codec_pref_index_audio(&tmp->nativeformats, 0);
+ pbx_builtin_setvar_helper(tmp, "SIP_CODEC_USED", ast_getformatname(fmt));
+
if (title)
snprintf(tmp->name, sizeof(tmp->name), "SIP/%s-%08x", title, (int)(long) i);
else if (strchr(i->fromdomain,':'))
@@ -6222,6 +6225,7 @@
/* Make a struct route */
thishop = malloc(sizeof(*thishop) + len);
if (thishop) {
@ -25,7 +37,7 @@
ast_copy_string(thishop->hop, rr, len);
ast_log(LOG_DEBUG, "build_route: Record-Route hop: <%s>\n", thishop->hop);
/* Link in */
@@ -6247,31 +6249,41 @@ static void build_route(struct sip_pvt *
@@ -6247,31 +6251,41 @@
/* Only append the contact if we are dealing with a strict router */
if (!head || (!ast_strlen_zero(head->hop) && strstr(head->hop,";lr") == NULL) ) {
@ -91,7 +103,7 @@
}
}
}
@@ -9248,6 +9260,13 @@ static int build_reply_digest(struct sip
@@ -9248,6 +9262,13 @@
secret = p->peersecret;
md5secret = p->peermd5secret;
}
@ -105,7 +117,7 @@
if (ast_strlen_zero(username)) /* We have no authentication */
return -1;
@@ -10621,7 +10640,11 @@ static int handle_request_invite(struct
@@ -10621,7 +10642,11 @@
gotdest = get_destination(p, NULL);
get_rdnis(p, NULL);
@ -118,7 +130,7 @@
build_contact(p);
if (gotdest) {
@@ -10649,7 +10672,6 @@ static int handle_request_invite(struct
@@ -10649,7 +10674,6 @@
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 */
@ -126,7 +138,7 @@
if (c) {
/* Pre-lock the call */
ast_mutex_lock(&c->lock);
@@ -10735,7 +10757,12 @@ static int handle_request_invite(struct
@@ -10735,7 +10759,12 @@
transmit_response(p, "180 Ringing", req);
break;
case AST_STATE_UP:

32
net/asterisk12/files/patch-channels::chan_sip.c
View file

@ -1,6 +1,9 @@
--- channels/chan_sip.c.orig Fri Nov 17 15:51:56 2006
+++ channels/chan_sip.c Fri Nov 17 15:53:30 2006
@@ -340,7 +340,7 @@ static char global_vmexten[AST_MAX_EXTEN
$FreeBSD$
--- channels/chan_sip.c.orig
+++ channels/chan_sip.c
@@ -340,7 +340,7 @@
static char default_language[MAX_LANGUAGE] = "";
@ -9,7 +12,7 @@
static char default_callerid[AST_MAX_EXTENSION] = DEFAULT_CALLERID;
static char default_fromdomain[AST_MAX_EXTENSION] = "";
@@ -483,6 +483,7 @@ struct sip_invite_param {
@@ -483,6 +483,7 @@
struct sip_route {
struct sip_route *next;
@ -17,7 +20,16 @@
char hop[0];
};
@@ -6222,6 +6223,7 @@ static void build_route(struct sip_pvt *
@@ -2815,6 +2816,8 @@
ast_codec_pref_remove2(&tmp->nativeformats, ~i->usercapability);
fmt = ast_codec_pref_index_audio(&tmp->nativeformats, 0);
+ pbx_builtin_setvar_helper(tmp, "SIP_CODEC_USED", ast_getformatname(fmt));
+
if (title)
snprintf(tmp->name, sizeof(tmp->name), "SIP/%s-%08x", title, (int)(long) i);
else if (strchr(i->fromdomain,':'))
@@ -6222,6 +6225,7 @@
/* Make a struct route */
thishop = malloc(sizeof(*thishop) + len);
if (thishop) {
@ -25,7 +37,7 @@
ast_copy_string(thishop->hop, rr, len);
ast_log(LOG_DEBUG, "build_route: Record-Route hop: <%s>\n", thishop->hop);
/* Link in */
@@ -6247,31 +6249,41 @@ static void build_route(struct sip_pvt *
@@ -6247,31 +6251,41 @@
/* Only append the contact if we are dealing with a strict router */
if (!head || (!ast_strlen_zero(head->hop) && strstr(head->hop,";lr") == NULL) ) {
@ -91,7 +103,7 @@
}
}
}
@@ -9248,6 +9260,13 @@ static int build_reply_digest(struct sip
@@ -9248,6 +9262,13 @@
secret = p->peersecret;
md5secret = p->peermd5secret;
}
@ -105,7 +117,7 @@
if (ast_strlen_zero(username)) /* We have no authentication */
return -1;
@@ -10621,7 +10640,11 @@ static int handle_request_invite(struct
@@ -10621,7 +10642,11 @@
gotdest = get_destination(p, NULL);
get_rdnis(p, NULL);
@ -118,7 +130,7 @@
build_contact(p);
if (gotdest) {
@@ -10649,7 +10672,6 @@ static int handle_request_invite(struct
@@ -10649,7 +10674,6 @@
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 */
@ -126,7 +138,7 @@
if (c) {
/* Pre-lock the call */
ast_mutex_lock(&c->lock);
@@ -10735,7 +10757,12 @@ static int handle_request_invite(struct
@@ -10735,7 +10759,12 @@
transmit_response(p, "180 Ringing", req);
break;
case AST_STATE_UP:

32
net/asterisk14/files/patch-channels::chan_sip.c
View file

@ -1,6 +1,9 @@
--- channels/chan_sip.c.orig Fri Nov 17 15:51:56 2006
+++ channels/chan_sip.c Fri Nov 17 15:53:30 2006
@@ -340,7 +340,7 @@ static char global_vmexten[AST_MAX_EXTEN
$FreeBSD$
--- channels/chan_sip.c.orig
+++ channels/chan_sip.c
@@ -340,7 +340,7 @@
static char default_language[MAX_LANGUAGE] = "";
@ -9,7 +12,7 @@
static char default_callerid[AST_MAX_EXTENSION] = DEFAULT_CALLERID;
static char default_fromdomain[AST_MAX_EXTENSION] = "";
@@ -483,6 +483,7 @@ struct sip_invite_param {
@@ -483,6 +483,7 @@
struct sip_route {
struct sip_route *next;
@ -17,7 +20,16 @@
char hop[0];
};
@@ -6222,6 +6223,7 @@ static void build_route(struct sip_pvt *
@@ -2815,6 +2816,8 @@
ast_codec_pref_remove2(&tmp->nativeformats, ~i->usercapability);
fmt = ast_codec_pref_index_audio(&tmp->nativeformats, 0);
+ pbx_builtin_setvar_helper(tmp, "SIP_CODEC_USED", ast_getformatname(fmt));
+
if (title)
snprintf(tmp->name, sizeof(tmp->name), "SIP/%s-%08x", title, (int)(long) i);
else if (strchr(i->fromdomain,':'))
@@ -6222,6 +6225,7 @@
/* Make a struct route */
thishop = malloc(sizeof(*thishop) + len);
if (thishop) {
@ -25,7 +37,7 @@
ast_copy_string(thishop->hop, rr, len);
ast_log(LOG_DEBUG, "build_route: Record-Route hop: <%s>\n", thishop->hop);
/* Link in */
@@ -6247,31 +6249,41 @@ static void build_route(struct sip_pvt *
@@ -6247,31 +6251,41 @@
/* Only append the contact if we are dealing with a strict router */
if (!head || (!ast_strlen_zero(head->hop) && strstr(head->hop,";lr") == NULL) ) {
@ -91,7 +103,7 @@
}
}
}
@@ -9248,6 +9260,13 @@ static int build_reply_digest(struct sip
@@ -9248,6 +9262,13 @@
secret = p->peersecret;
md5secret = p->peermd5secret;
}
@ -105,7 +117,7 @@
if (ast_strlen_zero(username)) /* We have no authentication */
return -1;
@@ -10621,7 +10640,11 @@ static int handle_request_invite(struct
@@ -10621,7 +10642,11 @@
gotdest = get_destination(p, NULL);
get_rdnis(p, NULL);
@ -118,7 +130,7 @@
build_contact(p);
if (gotdest) {
@@ -10649,7 +10672,6 @@ static int handle_request_invite(struct
@@ -10649,7 +10674,6 @@
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 */
@ -126,7 +138,7 @@
if (c) {
/* Pre-lock the call */
ast_mutex_lock(&c->lock);
@@ -10735,7 +10757,12 @@ static int handle_request_invite(struct
@@ -10735,7 +10759,12 @@
transmit_response(p, "180 Ringing", req);
break;
case AST_STATE_UP:

32
net/asterisk16/files/patch-channels::chan_sip.c
View file

@ -1,6 +1,9 @@
--- channels/chan_sip.c.orig Fri Nov 17 15:51:56 2006
+++ channels/chan_sip.c Fri Nov 17 15:53:30 2006
@@ -340,7 +340,7 @@ static char global_vmexten[AST_MAX_EXTEN
$FreeBSD$
--- channels/chan_sip.c.orig
+++ channels/chan_sip.c
@@ -340,7 +340,7 @@
static char default_language[MAX_LANGUAGE] = "";
@ -9,7 +12,7 @@
static char default_callerid[AST_MAX_EXTENSION] = DEFAULT_CALLERID;
static char default_fromdomain[AST_MAX_EXTENSION] = "";
@@ -483,6 +483,7 @@ struct sip_invite_param {
@@ -483,6 +483,7 @@
struct sip_route {
struct sip_route *next;
@ -17,7 +20,16 @@
char hop[0];
};
@@ -6222,6 +6223,7 @@ static void build_route(struct sip_pvt *
@@ -2815,6 +2816,8 @@
ast_codec_pref_remove2(&tmp->nativeformats, ~i->usercapability);
fmt = ast_codec_pref_index_audio(&tmp->nativeformats, 0);
+ pbx_builtin_setvar_helper(tmp, "SIP_CODEC_USED", ast_getformatname(fmt));
+
if (title)
snprintf(tmp->name, sizeof(tmp->name), "SIP/%s-%08x", title, (int)(long) i);
else if (strchr(i->fromdomain,':'))
@@ -6222,6 +6225,7 @@
/* Make a struct route */
thishop = malloc(sizeof(*thishop) + len);
if (thishop) {
@ -25,7 +37,7 @@
ast_copy_string(thishop->hop, rr, len);
ast_log(LOG_DEBUG, "build_route: Record-Route hop: <%s>\n", thishop->hop);
/* Link in */
@@ -6247,31 +6249,41 @@ static void build_route(struct sip_pvt *
@@ -6247,31 +6251,41 @@
/* Only append the contact if we are dealing with a strict router */
if (!head || (!ast_strlen_zero(head->hop) && strstr(head->hop,";lr") == NULL) ) {
@ -91,7 +103,7 @@
}
}
}
@@ -9248,6 +9260,13 @@ static int build_reply_digest(struct sip
@@ -9248,6 +9262,13 @@
secret = p->peersecret;
md5secret = p->peermd5secret;
}
@ -105,7 +117,7 @@
if (ast_strlen_zero(username)) /* We have no authentication */
return -1;
@@ -10621,7 +10640,11 @@ static int handle_request_invite(struct
@@ -10621,7 +10642,11 @@
gotdest = get_destination(p, NULL);
get_rdnis(p, NULL);
@ -118,7 +130,7 @@
build_contact(p);
if (gotdest) {
@@ -10649,7 +10672,6 @@ static int handle_request_invite(struct
@@ -10649,7 +10674,6 @@
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 */
@ -126,7 +138,7 @@
if (c) {
/* Pre-lock the call */
ast_mutex_lock(&c->lock);
@@ -10735,7 +10757,12 @@ static int handle_request_invite(struct
@@ -10735,7 +10759,12 @@
transmit_response(p, "180 Ringing", req);
break;
case AST_STATE_UP: