diff options
Diffstat (limited to 'net/asterisk/files/patch-channels::chan_sip.c')
| -rw-r--r-- | net/asterisk/files/patch-channels::chan_sip.c | 32 |
1 files changed, 22 insertions, 10 deletions
diff --git a/net/asterisk/files/patch-channels::chan_sip.c b/net/asterisk/files/patch-channels::chan_sip.c index 30eca068f9b1..8c24eff3a3aa 100644 --- a/net/asterisk/files/patch-channels::chan_sip.c +++ b/net/asterisk/files/patch-channels::chan_sip.c @@ -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: |