diff options
Diffstat (limited to 'net/asterisk10/files/patch-channels::chan_sip.c')
| -rw-r--r-- | net/asterisk10/files/patch-channels::chan_sip.c | 153 |
1 files changed, 17 insertions, 136 deletions
diff --git a/net/asterisk10/files/patch-channels::chan_sip.c b/net/asterisk10/files/patch-channels::chan_sip.c index 8c24eff3a3aa..06bb9c822545 100644 --- a/net/asterisk10/files/patch-channels::chan_sip.c +++ b/net/asterisk10/files/patch-channels::chan_sip.c @@ -1,109 +1,24 @@ - -$FreeBSD$ - ---- 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 @@ +--- channels/chan_sip.c.orig Wed Mar 28 11:23:42 2007 ++++ channels/chan_sip.c Wed Mar 28 11:36:27 2007 +@@ -484,7 +484,7 @@ static const struct cfsip_options { + #define DEFAULT_MOHINTERPRET "default" + #define DEFAULT_MOHSUGGEST "" + #define DEFAULT_VMEXTEN "asterisk" +-#define DEFAULT_CALLERID "asterisk" ++#define DEFAULT_CALLERID "Unknown" + #define DEFAULT_NOTIFYMIME "application/simple-message-summary" + #define DEFAULT_MWITIME 10 + #define DEFAULT_ALLOWGUEST TRUE +@@ -3822,6 +3823,8 @@ static struct ast_channel *sip_new(struc 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 @@ + /* If we have a prefcodec setting, we have an inbound channel that set a + preferred format for this call. Otherwise, we check the jointcapability + We also check for vrtp. If it's not there, we are not allowed do any video anyway. +@@ -11203,6 +11215,13 @@ static int build_reply_digest(struct sip secret = p->peersecret; md5secret = p->peermd5secret; } @@ -116,39 +31,5 @@ $FreeBSD$ + } 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: + |