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|
$FreeBSD$
--- channels/chan_sip.c.orig
+++ channels/chan_sip.c
@@ -339,7 +339,7 @@
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];
};
@@ -2835,6 +2836,8 @@
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,':'))
@@ -6312,6 +6315,7 @@
/* 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 */
@@ -6337,31 +6341,41 @@
/* 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;
+ }
}
}
}
@@ -9356,6 +9370,13 @@
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;
@@ -10759,7 +10780,11 @@
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) {
@@ -10780,7 +10805,6 @@
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);
@@ -10867,7 +10891,12 @@
transmit_response(p, "180 Ringing", req);
break;
case AST_STATE_UP:
- /* If this is not a re-invite or something to ignore - it's critical */
+ /* 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, (ignore || reinvite) ? 1 : 2);
break;
default:
|
