--- /dev/null
+From c2279540bade208dad35f7760ebd4a7cc94731fe Mon Sep 17 00:00:00 2001
+From: Ben Ford <bford@digium.com>
+Date: Mon, 21 Oct 2019 14:55:06 -0500
+Subject: [PATCH] chan_sip.c: Prevent address change on unauthenticated SIP request.
+
+If the name of a peer is known and a SIP request is sent using that
+peer's name, the address of the peer will change even if the request
+fails the authentication challenge. This means that an endpoint can
+be altered and even rendered unusuable, even if it was in a working
+state previously. This can only occur when the nat option is set to the
+default, or auto_force_rport.
+
+This change checks the result of authentication first to ensure it is
+successful before setting the address and the nat option.
+
+ASTERISK-28589 #close
+
+Change-Id: I581c5ed1da60ca89f590bd70872de2b660de02df
+---
+
+diff --git a/channels/chan_sip.c b/channels/chan_sip.c
+index ea78d23..4a8d344 100644
+--- a/channels/chan_sip.c
++++ b/channels/chan_sip.c
+@@ -19103,18 +19103,6 @@
+ bogus_peer = NULL;
+ }
+
+- /* build_peer, called through sip_find_peer, is not able to check the
+- * sip_pvt->natdetected flag in order to determine if the peer is behind
+- * NAT or not when SIP_PAGE3_NAT_AUTO_RPORT or SIP_PAGE3_NAT_AUTO_COMEDIA
+- * are set on the peer. So we check for that here and set the peer's
+- * address accordingly.
+- */
+- set_peer_nat(p, peer);
+-
+- if (p->natdetected && ast_test_flag(&peer->flags[2], SIP_PAGE3_NAT_AUTO_RPORT)) {
+- ast_sockaddr_copy(&peer->addr, &p->recv);
+- }
+-
+ if (!ast_apply_acl(peer->acl, addr, "SIP Peer ACL: ")) {
+ ast_debug(2, "Found peer '%s' for '%s', but fails host access\n", peer->name, of);
+ sip_unref_peer(peer, "sip_unref_peer: check_peer_ok: from sip_find_peer call, early return of AUTH_ACL_FAILED");
+@@ -19183,6 +19171,21 @@
+ ast_string_field_set(p, peermd5secret, NULL);
+ }
+ if (!(res = check_auth(p, req, peer->name, p->peersecret, p->peermd5secret, sipmethod, uri2, reliable))) {
++
++ /* build_peer, called through sip_find_peer, is not able to check the
++ * sip_pvt->natdetected flag in order to determine if the peer is behind
++ * NAT or not when SIP_PAGE3_NAT_AUTO_RPORT or SIP_PAGE3_NAT_AUTO_COMEDIA
++ * are set on the peer. So we check for that here and set the peer's
++ * address accordingly. The address should ONLY be set once we are sure
++ * authentication was a success. If, for example, an INVITE was sent that
++ * matched the peer name but failed the authentication check, the address
++ * would be updated, which is bad.
++ */
++ set_peer_nat(p, peer);
++ if (p->natdetected && ast_test_flag(&peer->flags[2], SIP_PAGE3_NAT_AUTO_RPORT)) {
++ ast_sockaddr_copy(&peer->addr, &p->recv);
++ }
++
+ /* If we have a call limit, set flag */
+ if (peer->call_limit)
+ ast_set_flag(&p->flags[0], SIP_CALL_LIMIT);
+@@ -19282,6 +19285,7 @@
+ }
+ }
+ sip_unref_peer(peer, "check_peer_ok: sip_unref_peer: tossing temp ptr to peer from sip_find_peer");
++
+ return res;
+ }
+
--- /dev/null
+From 1b9281a5ded62e5d30af2959e5aa33bc5a0fc285 Mon Sep 17 00:00:00 2001
+From: George Joseph <gjoseph@digium.com>
+Date: Thu, 24 Oct 2019 11:41:23 -0600
+Subject: [PATCH] manager.c: Prevent the Originate action from running the Originate app
+
+If an AMI user without the "system" authorization calls the
+Originate AMI command with the Originate application,
+the second Originate could run the "System" command.
+
+Action: Originate
+Channel: Local/1111
+Application: Originate
+Data: Local/2222,app,System,touch /tmp/owned
+
+If the "system" authorization isn't set, we now block the
+Originate app as well as the System, Exec, etc. apps.
+
+ASTERISK-28580
+Reported by: Eliel Sardañons
+
+Change-Id: Ic4c9dedc34c426f03c8c14fce334a71386d8a5fa
+---
+
+diff --git a/doc/UPGRADE-staging/AMI-Originate.txt b/doc/UPGRADE-staging/AMI-Originate.txt
+new file mode 100644
+index 0000000..f2d3133
+--- /dev/null
++++ b/doc/UPGRADE-staging/AMI-Originate.txt
+@@ -0,0 +1,5 @@
++Subject: AMI
++
++The AMI Originate action, which optionally takes a dialplan application as
++an argument, no longer accepts "Originate" as the application due to
++security concerns.
+diff --git a/main/manager.c b/main/manager.c
+index fc602bc..44e25b8 100644
+--- a/main/manager.c
++++ b/main/manager.c
+@@ -5708,6 +5708,7 @@
+ EAGI(/bin/rm,-rf /) */
+ strcasestr(app, "mixmonitor") || /* MixMonitor(blah,,rm -rf) */
+ strcasestr(app, "externalivr") || /* ExternalIVR(rm -rf) */
++ strcasestr(app, "originate") || /* Originate(Local/1234,app,System,rm -rf) */
+ (strstr(appdata, "SHELL") && (bad_appdata = 1)) || /* NoOp(${SHELL(rm -rf /)}) */
+ (strstr(appdata, "EVAL") && (bad_appdata = 1)) /* NoOp(${EVAL(${some_var_containing_SHELL})}) */
+ )) {
--- /dev/null
+From c257794330db49f4079a7108d51da60696269b36 Mon Sep 17 00:00:00 2001
+From: Ben Ford <bford@digium.com>
+Date: Fri, 08 Nov 2019 13:21:15 -0600
+Subject: [PATCH] res_pjsip_session.c: Check for port of zero on incoming SDP.
+
+If a re-invite comes in initiating T.38, but there is no c line in the
+SDP and the port is also 0, a crash can occur. A check is now done on
+the port to see if the steam is already declined, preventing the crash.
+The logic was moved to res_pjsip_session.c because it is handled in a
+similar manner in later versions of Asterisk.
+
+ASTERISK-28612
+Reported by: Salah Ahmed
+
+Change-Id: Ifc4a0d05b32c7f2156e77fc8435a6ecaa6abada0
+---
+
+diff --git a/res/res_pjsip_session.c b/res/res_pjsip_session.c
+index 81f36a7..12cf41d 100644
+--- a/res/res_pjsip_session.c
++++ b/res/res_pjsip_session.c
+@@ -235,6 +235,13 @@
+ continue;
+ }
+
++ /* If we have a port of 0, ignore this stream */
++ if (!sdp->media[i]->desc.port) {
++ ast_debug(1, "Declining incoming SDP media stream '%s' at position '%d'\n",
++ session_media->stream_type, i);
++ continue;
++ }
++
+ if (session_media->handler) {
+ handler = session_media->handler;
+ ast_debug(1, "Negotiating incoming SDP media stream '%s' using %s SDP handler\n",
projects / feed / telephony.git / commitdiff