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-rw-r--r--net/asterisk10/files/patch-channels::chan_sip.c154
1 files changed, 0 insertions, 154 deletions
diff --git a/net/asterisk10/files/patch-channels::chan_sip.c b/net/asterisk10/files/patch-channels::chan_sip.c
deleted file mode 100644
index 8c24eff3a3aa..000000000000
--- a/net/asterisk10/files/patch-channels::chan_sip.c
+++ /dev/null
@@ -1,154 +0,0 @@
-
-$FreeBSD$
-
---- channels/chan_sip.c.orig
-+++ channels/chan_sip.c
-@@ -340,7 +340,7 @@
-
- static char default_language[MAX_LANGUAGE] = "";
-
--#define DEFAULT_CALLERID "asterisk"
-+#define DEFAULT_CALLERID "Unknown"
- static char default_callerid[AST_MAX_EXTENSION] = DEFAULT_CALLERID;
-
- static char default_fromdomain[AST_MAX_EXTENSION] = "";
-@@ -483,6 +483,7 @@
-
- struct sip_route {
- struct sip_route *next;
-+ int lr;
- char hop[0];
- };
-
-@@ -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) {
-+ thishop->lr = (strnstr(rr, ";lr", len) != NULL ? 1 : 0);
- ast_copy_string(thishop->hop, rr, len);
- ast_log(LOG_DEBUG, "build_route: Record-Route hop: <%s>\n", thishop->hop);
- /* Link in */
-@@ -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) ) {
-- /* 2nd append the Contact: if there is one */
-- /* Can be multiple Contact headers, comma separated values - we just take the first */
-- contact = get_header(req, "Contact");
-- if (!ast_strlen_zero(contact)) {
-- ast_log(LOG_DEBUG, "build_route: Contact hop: %s\n", contact);
-- /* Look for <: delimited address */
-- c = strchr(contact, '<');
-- if (c) {
-- /* Take to > */
-- ++c;
-- len = strcspn(c, ">") + 1;
-- } else {
-- /* No <> - just take the lot */
-- c = contact;
-- len = strlen(contact) + 1;
-- }
-- thishop = malloc(sizeof(*thishop) + len);
-+ /* Duplicate first route from the list */
-+ if (head && head->lr) {
-+ thishop = (struct sip_route *)malloc(sizeof(struct sip_route)+strlen(head->hop)+1);
- if (thishop) {
-- ast_copy_string(thishop->hop, c, len);
-- thishop->next = NULL;
-- /* Goes at the end */
-- if (tail)
-- tail->next = thishop;
-- else
-- head = thishop;
-+ memcpy(thishop, head, sizeof(struct sip_route)+strlen(head->hop)+1);
-+ thishop->next = head;
-+ head = thishop;
-+ }
-+ } else {
-+ /* Append the Contact: if there is one and first route is w/o `lr' param */
-+ /* Can be multiple Contact headers, comma separated values - we just take the first */
-+ contact = get_header(req, "Contact");
-+ if (!ast_strlen_zero(contact)) {
-+ ast_log(LOG_DEBUG, "build_route: Contact hop: %s\n", contact);
-+ /* Look for <: delimited address */
-+ c = strchr(contact, '<');
-+ if (c) {
-+ /* Take to > */
-+ ++c;
-+ len = strcspn(c, ">") + 1;
-+ } else {
-+ /* No <> - just take the lot */
-+ c = contact;
-+ len = strlen(contact) + 1;
-+ }
-+ thishop = malloc(sizeof(*thishop) + len);
-+ if (thishop) {
-+ ast_copy_string(thishop->hop, c, len);
-+ thishop->next = NULL;
-+ /* Goes at the end */
-+ if (tail)
-+ tail->next = thishop;
-+ else
-+ head = thishop;
-+ }
- }
- }
- }
-@@ -9248,6 +9262,13 @@
- secret = p->peersecret;
- md5secret = p->peermd5secret;
- }
-+ /* No authentication. Try to get auth info from channel vars */
-+ if (ast_strlen_zero(username))
-+ {
-+ username = pbx_builtin_getvar_helper(p->owner, "SIP_AUTH_NAME");
-+ secret = pbx_builtin_getvar_helper(p->owner, "SIP_AUTH_SECRET");
-+ md5secret = pbx_builtin_getvar_helper(p->owner, "SIP_AUTH_MD5SECRET");
-+ }
- if (ast_strlen_zero(username)) /* We have no authentication */
- return -1;
-
-@@ -10621,7 +10642,11 @@
- gotdest = get_destination(p, NULL);
-
- get_rdnis(p, NULL);
-- extract_uri(p, req);
-+ build_route(p, req, 0);
-+ if (!p->route->lr)
-+ strncpy(p->uri, p->route->hop, sizeof(p->uri) - 1);
-+ else
-+ extract_uri(p, req);
- build_contact(p);
-
- if (gotdest) {
-@@ -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 */
-- build_route(p, req, 0);
- if (c) {
- /* Pre-lock the call */
- ast_mutex_lock(&c->lock);
-@@ -10735,7 +10759,12 @@
- 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);
-+ } else {
-+ ast_log(LOG_DEBUG, "Hm.... No sdp for the moment\n");
-+ }
- transmit_response_with_sdp(p, "200 OK", req, 1);
- break;
- default: