o Upgrade to 1.2.23.

o Disable fax-related modules since they don't compile with newest spandsp.

o Enable FreeTDS support (could be disabled using WITHOUT_FREETDS knob).

Sponsored by: Sippy Software, Inc. ( http://www.sippysoft.com/ )

4 files changed, 22 insertions, 18 deletions

diff --git a/net/asterisk12/files/patch-Makefile b/net/asterisk12/files/patch-Makefile
index 432d517b4956..779f7d635207 100644
--- a/net/asterisk12/files/patch-Makefile
+++ b/net/asterisk12/files/patch-Makefile
@@ -343,7 +343,7 @@ $FreeBSD$
 +	$(BSD_INSTALL_SCRIPT) contrib/scripts/astgenkey $(DESTDIR)$(ASTSBINDIR)/
 +	$(BSD_INSTALL_SCRIPT) contrib/scripts/autosupport $(DESTDIR)$(ASTSBINDIR)/	
  	if [ ! -f $(DESTDIR)$(ASTSBINDIR)/safe_asterisk ]; then \
- 		cat contrib/scripts/safe_asterisk | sed 's|__ASTERISK_SBIN_DIR__|$(ASTSBINDIR)|;' > $(DESTDIR)$(ASTSBINDIR)/safe_asterisk ;\
+ 		cat contrib/scripts/safe_asterisk | sed 's|__ASTERISK_SBIN_DIR__|$(ASTSBINDIR)|;s|__ASTERISK_VARRUN_DIR__|$(ASTVARRUNDIR)|;' > $(DESTDIR)$(ASTSBINDIR)/safe_asterisk ;\
  		chmod 755 $(DESTDIR)$(ASTSBINDIR)/safe_asterisk;\
  	fi
 -	$(INSTALL) -d $(DESTDIR)$(ASTHEADERDIR)
@@ -470,7 +470,7 @@ $FreeBSD$
  	done
 -	mkdir -p $(DESTDIR)$(ASTVARLIBDIR)/mohmp3 ; \
 +	$(MKDIR) $(DESTDIR)$(ASTVARLIBDIR)/mohmp3 ; \
- 	for x in sounds/*.mp3; do \
+ 	for x in sounds/moh/*.mp3 sounds/moh/LICENSE; do \
 -		$(INSTALL) -m 644 $$x $(DESTDIR)$(ASTVARLIBDIR)/mohmp3 ; \
 +		$(BSD_INSTALL_DATA) $$x $(DESTDIR)$(ASTVARLIBDIR)/mohmp3 ; \
  	done
diff --git a/net/asterisk12/files/patch-cdr::Makefile b/net/asterisk12/files/patch-cdr::Makefile
index 8628383461d2..5b905c7778ef 100644
--- a/net/asterisk12/files/patch-cdr::Makefile
+++ b/net/asterisk12/files/patch-cdr::Makefile
@@ -30,7 +30,7 @@ $FreeBSD$
  #
 -ifneq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/include/tds.h)$(wildcard $(CROSS_COMPILE_TARGET)/usr/local/include/tds.h)$(wildcard $(CROSS_COMPILE_TARGET)/usr/include/freetds/tds.h),)
 -  ifeq ($(shell grep -s TDS_VERSION_NO $(CROSS_COMPILE_TARGET)/usr/include/tdsver.h $(CROSS_COMPILE_TARGET)/usr/local/include/tdsver.h $(CROSS_COMPILE_TARGET)/usr/include/freetds/tdsver.h | grep -c 0.64),1)
-+ifneq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/include/tds.h)$(wildcard $(CROSS_COMPILE_TARGET)$(LOCALBASE)/include/tds.h)$(wildcard $(CROSS_COMPILE_TARGET)/usr/include/freetds/tds.h),)
++ifdef WITH_FREETDS
 +  ifeq ($(shell grep -s TDS_VERSION_NO $(CROSS_COMPILE_TARGET)/usr/include/tdsver.h $(CROSS_COMPILE_TARGET)$(LOCALBASE)/include/tdsver.h $(CROSS_COMPILE_TARGET)/usr/include/freetds/tdsver.h | grep -c 0.64),1)
      CFLAGS += -DFREETDS_0_64
    else
diff --git a/net/asterisk12/files/patch-channels::Makefile b/net/asterisk12/files/patch-channels::Makefile
index b80c1bfdf134..56539d83d6b9 100644
--- a/net/asterisk12/files/patch-channels::Makefile
+++ b/net/asterisk12/files/patch-channels::Makefile
@@ -14,7 +14,7 @@ $FreeBSD$
    CHANH323LIB=-pthread
  endif
  
-@@ -75,11 +75,11 @@
+@@ -75,7 +75,7 @@
    SOLINK+=-lrt
  endif
  
@@ -23,12 +23,16 @@ $FreeBSD$
    CHANNEL_LIBS+=chan_phone.so
  endif
  
+@@ -83,7 +83,7 @@
+   CFLAGS+=-DHAVE_LINUX_COMPILER_H
+ endif
+ 
 -ifneq ($(wildcard h323/libchanh323.a),)
 +ifdef WITH_H323
    CHANNEL_LIBS+=chan_h323.so
  endif
  
-@@ -94,12 +94,12 @@
+@@ -102,12 +102,12 @@
    CHANNEL_LIBS+=chan_alsa.so
  endif
  
@@ -43,7 +47,7 @@ $FreeBSD$
    CFLAGS+=-DZAPATA_R2
    ZAPR2=-lmfcr2
  endif
-@@ -112,12 +112,12 @@
+@@ -120,12 +120,12 @@
  endif
  
  ifndef WITHOUT_ZAPTEL
@@ -58,7 +62,7 @@ $FreeBSD$
    endif
    CFLAGS+=-DIAX_TRUNKING
    CHANNEL_LIBS+=chan_zap.so
-@@ -233,8 +233,10 @@
+@@ -241,8 +241,10 @@
  chan_h323.so: chan_h323.o h323/libchanh323.a h323/Makefile.ast
  	$(CC) $(SOLINK) $(H323LDFLAGS) -o $@ $< h323/libchanh323.a $(H323LDLIBS) -lstdc++
  else
@@ -71,7 +75,7 @@ $FreeBSD$
  endif
  
  misdn/chan_misdn_lib.a:
-@@ -253,8 +255,7 @@
+@@ -261,8 +263,7 @@
  #	$(CC) -rdynamic -shared -Xlinker -x -o $@ $<
  
  install: all
diff --git a/net/asterisk12/files/patch-channels::chan_sip.c b/net/asterisk12/files/patch-channels::chan_sip.c
index 8c24eff3a3aa..c06789847723 100644
--- a/net/asterisk12/files/patch-channels::chan_sip.c
+++ b/net/asterisk12/files/patch-channels::chan_sip.c
@@ -3,7 +3,7 @@ $FreeBSD$
 
 --- channels/chan_sip.c.orig
 +++ channels/chan_sip.c
-@@ -340,7 +340,7 @@
+@@ -339,7 +339,7 @@
  
  static char default_language[MAX_LANGUAGE] = "";
  
@@ -20,7 +20,7 @@ $FreeBSD$
  	char hop[0];
  };
  
-@@ -2815,6 +2816,8 @@
+@@ -2835,6 +2836,8 @@
  	ast_codec_pref_remove2(&tmp->nativeformats, ~i->usercapability);
  	fmt = ast_codec_pref_index_audio(&tmp->nativeformats, 0);
  
@@ -29,7 +29,7 @@ $FreeBSD$
  	if (title)
  		snprintf(tmp->name, sizeof(tmp->name), "SIP/%s-%08x", title, (int)(long) i);
  	else if (strchr(i->fromdomain,':'))
-@@ -6222,6 +6225,7 @@
+@@ -6312,6 +6315,7 @@
  			/* Make a struct route */
  			thishop = malloc(sizeof(*thishop) + len);
  			if (thishop) {
@@ -37,7 +37,7 @@ $FreeBSD$
  				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 @@
+@@ -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) ) {
@@ -103,7 +103,7 @@ $FreeBSD$
  			}
  		}
  	}
-@@ -9248,6 +9262,13 @@
+@@ -9356,6 +9370,13 @@
   		secret =  p->peersecret;
   		md5secret = p->peermd5secret;
   	}
@@ -117,7 +117,7 @@ $FreeBSD$
  	if (ast_strlen_zero(username))	/* We have no authentication */
  		return -1;
   
-@@ -10621,7 +10642,11 @@
+@@ -10759,7 +10780,11 @@
  		gotdest = get_destination(p, NULL);
  
  		get_rdnis(p, NULL);
@@ -130,7 +130,7 @@ $FreeBSD$
  		build_contact(p);
  
  		if (gotdest) {
-@@ -10649,7 +10674,6 @@
+@@ -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 */
@@ -138,17 +138,17 @@ $FreeBSD$
  			if (c) {
  				/* Pre-lock the call */
  				ast_mutex_lock(&c->lock);
-@@ -10735,7 +10759,12 @@
+@@ -10867,7 +10891,12 @@
  			transmit_response(p, "180 Ringing", req);
  			break;
  		case AST_STATE_UP:
--                        /* Here we have reINVITE request - try to renegotiate codecs with */
+-			/* 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, 1);
+ 			transmit_response_with_sdp(p, "200 OK", req, (ignore || reinvite) ? 1 : 2);
  			break;
  		default: