[asterisk-commits] oej: trunk r276951 - in /trunk: ./ channels/ channels/sip/include/ configs/ d...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Jul 16 05:01:02 CDT 2010
Author: oej
Date: Fri Jul 16 05:00:58 2010
New Revision: 276951
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=276951
Log:
Add ability to configure the Max-Forwards header in the dialplan, as well as in
sip.conf configuration for the channel and for devices.
The Max-Forwards header is used to prevent loops in a SIP network. Each intermediary,
like SIP proxys and SBCs, decrement this counter and detects when it reaches zero,
at which point the SIP request is nicely killed in a SIP-friendly way.
Review: https://reviewboard.asterisk.org/r/778/
Thanks to dvossel for the review and good advice.
Modified:
trunk/CHANGES
trunk/channels/chan_sip.c
trunk/channels/sip/include/sip.h
trunk/configs/sip.conf.sample
trunk/doc/tex/channelvariables.tex
Modified: trunk/CHANGES
URL: http://svnview.digium.com/svn/asterisk/trunk/CHANGES?view=diff&rev=276951&r1=276950&r2=276951
==============================================================================
--- trunk/CHANGES (original)
+++ trunk/CHANGES Fri Jul 16 05:00:58 2010
@@ -71,6 +71,8 @@
* Added support for 16khz signed linear media streams.
* SIP is now able to bind to and communicate with IPv6 addresses. In addition,
RTP has been outfitted with the same abilities.
+ * Added support for setting the Max-Forwards: header in SIP requests. Setting is
+ available in device configurations as well as in the dial plan.
IAX2 Changes
-----------
Modified: trunk/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_sip.c?view=diff&rev=276951&r1=276950&r2=276951
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Fri Jul 16 05:00:58 2010
@@ -417,6 +417,9 @@
<enum name="useragent">
<para>Current user agent id for peer.</para>
</enum>
+ <enum name="maxforwards">
+ <para>The value used for SIP loop prevention in outbound requests</para>
+ </enum>
<enum name="chanvar[name]">
<para>A channel variable configured with setvar for this peer.</para>
</enum>
@@ -1482,6 +1485,7 @@
static void build_callid_registry(struct sip_registry *reg, const struct ast_sockaddr *ourip, const char *fromdomain);
static void make_our_tag(char *tagbuf, size_t len);
static int add_header(struct sip_request *req, const char *var, const char *value);
+static int add_header_max_forwards(struct sip_pvt *dialog, struct sip_request *req);
static int add_content(struct sip_request *req, const char *line);
static int finalize_content(struct sip_request *req);
static int add_text(struct sip_request *req, const char *text);
@@ -6996,6 +7000,7 @@
p->session_modify = TRUE;
p->stimer = NULL;
p->prefs = default_prefs; /* Set default codecs for this call */
+ p->maxforwards = sip_cfg.default_max_forwards;
if (intended_method != SIP_OPTIONS) { /* Peerpoke has it's own system */
p->timer_t1 = global_t1; /* Default SIP retransmission timer T1 (RFC 3261) */
@@ -8794,6 +8799,23 @@
return 0;
}
+/*! \brief Add 'Max-Forwards' header to SIP message */
+static int add_header_max_forwards(struct sip_pvt *dialog, struct sip_request *req)
+{
+ char clen[10];
+ const char *max = NULL;
+
+ if (dialog->owner) {
+ max = pbx_builtin_getvar_helper(dialog->owner, "SIP_MAX_FORWARDS");
+ }
+
+ /* The channel variable overrides the peer/channel value */
+ if (max == NULL) {
+ snprintf(clen, sizeof(clen), "%d", dialog->maxforwards);
+ }
+ return add_header(req, "Max-Forwards", max != NULL ? max : clen);
+}
+
/*! \brief Add 'Content-Length' header and content to SIP message */
static int finalize_content(struct sip_request *req)
{
@@ -9292,7 +9314,7 @@
set_destination(p, p->route->hop);
add_route(req, is_strict ? p->route->next : p->route);
}
- add_header(req, "Max-Forwards", DEFAULT_MAX_FORWARDS);
+ add_header_max_forwards(p, req);
ot = get_header(orig, "To");
of = get_header(orig, "From");
@@ -10853,7 +10875,7 @@
snprintf(tmp_n, sizeof(tmp_n), "%d %s", ++p->ocseq, sip_methods[sipmethod].text);
add_header(req, "Via", p->via);
- add_header(req, "Max-Forwards", DEFAULT_MAX_FORWARDS);
+ add_header_max_forwards(p, req);
/* This will be a no-op most of the time. However, under certain circumstances,
* NOTIFY messages will use this function for preparing the request and should
* have Route headers present.
@@ -12005,7 +12027,7 @@
build_via(p);
add_header(&req, "Via", p->via);
- add_header(&req, "Max-Forwards", DEFAULT_MAX_FORWARDS);
+ add_header_max_forwards(p, &req);
add_header(&req, "From", from);
add_header(&req, "To", to);
add_header(&req, "Call-ID", p->callid);
@@ -14507,6 +14529,9 @@
p->capability = peer->capability;
p->prefs = peer->prefs;
p->jointcapability = peer->capability;
+ if (peer->maxforwards > 0) {
+ p->maxforwards = peer->maxforwards;
+ }
if (p->peercapability)
p->jointcapability &= p->peercapability;
p->maxcallbitrate = peer->maxcallbitrate;
@@ -15834,6 +15859,7 @@
ast_cli(fd, " VM Extension : %s\n", peer->vmexten);
ast_cli(fd, " LastMsgsSent : %d/%d\n", (peer->lastmsgssent & 0x7fff0000) >> 16, peer->lastmsgssent & 0xffff);
ast_cli(fd, " Call limit : %d\n", peer->call_limit);
+ ast_cli(fd, " Max forwards : %d\n", peer->maxforwards);
if (peer->busy_level)
ast_cli(fd, " Busy level : %d\n", peer->busy_level);
ast_cli(fd, " Dynamic : %s\n", AST_CLI_YESNO(peer->host_dynamic));
@@ -15944,7 +15970,9 @@
peer_mailboxes_to_str(&mailbox_str, peer);
astman_append(s, "VoiceMailbox: %s\r\n", mailbox_str->str);
astman_append(s, "TransferMode: %s\r\n", transfermode2str(peer->allowtransfer));
+ astman_append(s, "Maxforwards: %d\r\n", peer->maxforwards);
astman_append(s, "LastMsgsSent: %d\r\n", peer->lastmsgssent);
+ astman_append(s, "Maxforwards: %d\r\n", peer->maxforwards);
astman_append(s, "Call-limit: %d\r\n", peer->call_limit);
astman_append(s, "Busy-level: %d\r\n", peer->busy_level);
astman_append(s, "MaxCallBR: %d kbps\r\n", peer->maxcallbitrate);
@@ -16520,6 +16548,7 @@
ast_cli(a->fd, " Timer T1 minimum: %d\n", global_t1min);
ast_cli(a->fd, " Timer B: %d\n", global_timer_b);
ast_cli(a->fd, " No premature media: %s\n", AST_CLI_YESNO(global_prematuremediafilter));
+ ast_cli(a->fd, " Max forwards: %d\n", sip_cfg.default_max_forwards);
ast_cli(a->fd, "\nDefault Settings:\n");
ast_cli(a->fd, "-----------------\n");
@@ -17740,6 +17769,8 @@
snprintf(buf, len, "%d", peer->busy_level);
} else if (!strcasecmp(colname, "curcalls")) {
snprintf(buf, len, "%d", peer->inUse);
+ } else if (!strcasecmp(colname, "maxforwards")) {
+ snprintf(buf, len, "%d", peer->maxforwards);
} else if (!strcasecmp(colname, "accountcode")) {
ast_copy_string(buf, peer->accountcode, len);
} else if (!strcasecmp(colname, "callgroup")) {
@@ -25632,6 +25663,11 @@
ast_log(LOG_WARNING, "Invalid AMA Flags for peer: %s at line %d\n", v->value, v->lineno);
} else {
peer->amaflags = format;
+ }
+ } else if (!strcasecmp(v->name, "maxforwards")) {
+ if ((sscanf(v->value, "%30d", &peer->maxforwards) != 1) || (peer->maxforwards < 1)) {
+ ast_log(LOG_WARNING, "'%s' is not a valid maxforwards value at line %d. Using default.\n", v->value, v->lineno);
+ peer->maxforwards = sip_cfg.default_max_forwards;
}
} else if (!strcasecmp(v->name, "accountcode")) {
ast_string_field_set(peer, accountcode, v->value);
@@ -26200,6 +26236,7 @@
/* Initialize some reasonable defaults at SIP reload (used both for channel and as default for devices */
ast_copy_string(sip_cfg.default_context, DEFAULT_CONTEXT, sizeof(sip_cfg.default_context));
sip_cfg.default_subscribecontext[0] = '\0';
+ sip_cfg.default_max_forwards = DEFAULT_MAX_FORWARDS;
default_language[0] = '\0';
default_fromdomain[0] = '\0';
default_fromdomainport = 0;
@@ -26680,6 +26717,11 @@
}
} else if (!strcasecmp(v->name, "use_q850_reason")) {
ast_set2_flag(&global_flags[1], ast_true(v->value), SIP_PAGE2_Q850_REASON);
+ } else if (!strcasecmp(v->name, "maxforwards")) {
+ if ((sscanf(v->value, "%30d", &sip_cfg.default_max_forwards) != 1) || (sip_cfg.default_max_forwards < 1)) {
+ ast_log(LOG_WARNING, "'%s' is not a valid maxforwards value at line %d. Using default.\n", v->value, v->lineno);
+ sip_cfg.default_max_forwards = DEFAULT_MAX_FORWARDS;
+ }
} else if (!strcasecmp(v->name, "snom_aoc_enabled")) {
ast_set2_flag(&global_flags[2], ast_true(v->value), SIP_PAGE3_SNOM_AOC);
}
Modified: trunk/channels/sip/include/sip.h
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/sip/include/sip.h?view=diff&rev=276951&r1=276950&r2=276951
==============================================================================
--- trunk/channels/sip/include/sip.h (original)
+++ trunk/channels/sip/include/sip.h Fri Jul 16 05:00:58 2010
@@ -57,7 +57,7 @@
#define DEFAULT_MAX_EXPIRY 3600
#define DEFAULT_MWI_EXPIRY 3600
#define DEFAULT_REGISTRATION_TIMEOUT 20
-#define DEFAULT_MAX_FORWARDS "70"
+#define DEFAULT_MAX_FORWARDS 70
/* guard limit must be larger than guard secs */
/* guard min must be < 1000, and should be >= 250 */
@@ -703,6 +703,7 @@
struct ast_ha *contact_ha; /*! \brief Global list of addresses dynamic peers are not allowed to use */
format_t capability; /*!< Supported codecs */
int tcp_enabled;
+ int default_max_forwards; /*!< Default max forwards (SIP Anti-loop) */
};
/*! \brief The SIP socket definition */
@@ -953,6 +954,7 @@
AST_STRING_FIELD(dialstring); /*!< The dialstring used to call this SIP endpoint */
);
char via[128]; /*!< Via: header */
+ int maxforwards; /*!< SIP Loop prevention */
struct sip_socket socket; /*!< The socket used for this dialog */
unsigned int ocseq; /*!< Current outgoing seqno */
unsigned int icseq; /*!< Current incoming seqno */
@@ -1194,6 +1196,7 @@
int call_limit; /*!< Limit of concurrent calls */
int t38_maxdatagram; /*!< T.38 FaxMaxDatagram override */
int busy_level; /*!< Level of active channels where we signal busy */
+ int maxforwards; /*!< SIP Loop prevention */
enum transfermodes allowtransfer; /*! SIP Refer restriction scheme */
struct ast_codec_pref prefs; /*!< codec prefs */
int lastmsgssent;
Modified: trunk/configs/sip.conf.sample
URL: http://svnview.digium.com/svn/asterisk/trunk/configs/sip.conf.sample?view=diff&rev=276951&r1=276950&r2=276951
==============================================================================
--- trunk/configs/sip.conf.sample (original)
+++ trunk/configs/sip.conf.sample Fri Jul 16 05:00:58 2010
@@ -202,6 +202,8 @@
;minexpiry=60 ; Minimum length of registrations/subscriptions (default 60)
;defaultexpiry=120 ; Default length of incoming/outgoing registration
;mwiexpiry=3600 ; Expiry time for outgoing MWI subscriptions
+;maxforwards=70 ; Setting for the SIP Max-Forwards: header (loop prevention)
+ ; Default value is 70
;qualifyfreq=60 ; Qualification: How often to check for the host to be up in seconds
; Set to low value if you use low timeout for NAT of UDP sessions
; Default: 60
@@ -1035,6 +1037,7 @@
; directmediadeny
; unsolicited_mailbox
; use_q850_reason
+; maxforwards
;[sip_proxy]
; For incoming calls only. Example: FWD (Free World Dialup)
Modified: trunk/doc/tex/channelvariables.tex
URL: http://svnview.digium.com/svn/asterisk/trunk/doc/tex/channelvariables.tex?view=diff&rev=276951&r1=276950&r2=276951
==============================================================================
--- trunk/doc/tex/channelvariables.tex (original)
+++ trunk/doc/tex/channelvariables.tex Fri Jul 16 05:00:58 2010
@@ -933,6 +933,7 @@
${SIPFROMDOMAIN} Set SIP domain on outbound calls
${SIPUSERAGENT} * SIP user agent (deprecated)
${SIPURI} * SIP uri
+${SIP_MAX_FORWARDS} Set the value of the Max-Forwards header for outbound call
${SIP_CODEC} Set the SIP codec for an inbound call
${SIP_CODEC_INBOUND} Set the SIP codec for an inbound call
${SIP_CODEC_OUTBOUND} Set the SIP codec for an outbound call
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