[Asterisk-code-review] res pjsip: Add ignore uri user options option. (asterisk[certified/13.8])
Richard Mudgett
asteriskteam at digium.com
Thu Sep 8 19:25:42 CDT 2016
Richard Mudgett has uploaded a new change for review.
https://gerrit.asterisk.org/3860
Change subject: res_pjsip: Add ignore_uri_user_options option.
......................................................................
res_pjsip: Add ignore_uri_user_options option.
This implements the chan_sip legacy_useroption_parsing option but with a
better name.
ASTERISK-26316 #close
Reported by: Kevin Harwell
Change-Id: Ib42b0e940dd34d84c7b14bc2e90d1ba392624f62
---
M CHANGES
M configs/samples/pjsip.conf.sample
A contrib/ast-db-manage/config/versions/a6ef36f1309_ps_globals_add_ignore_uri_user_options.py
M include/asterisk/res_pjsip.h
M res/res_pjsip.c
M res/res_pjsip/config_global.c
M res/res_pjsip/pjsip_options.c
M res/res_pjsip_endpoint_identifier_user.c
M res/res_pjsip_messaging.c
M res/res_pjsip_path.c
M res/res_pjsip_pubsub.c
M res/res_pjsip_refer.c
M res/res_pjsip_registrar.c
M res/res_pjsip_session.c
14 files changed, 235 insertions(+), 16 deletions(-)
git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/60/3860/1
diff --git a/CHANGES b/CHANGES
index fed0d40..756e224 100644
--- a/CHANGES
+++ b/CHANGES
@@ -9,6 +9,20 @@
==============================================================================
------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 13.8-cert3 to Asterisk 13.8-cert4 ----
+------------------------------------------------------------------------------
+
+res_pjsip
+------------------
+ * Added "ignore_uri_user_options" global configuration option for
+ compatibility with an ITSP that sends URI user field options. When enabled
+ the user field is truncated at the first semicolon.
+ Example:
+ URI: "sip:1235557890;phone-context=national at x.x.x.x;user=phone"
+ The user field is "1235557890;phone-context=national"
+ Which is truncated to this: "1235557890"
+
+------------------------------------------------------------------------------
--- Functionality changes from Asterisk 13.8-cert2 to Asterisk 13.8-cert3 ----
------------------------------------------------------------------------------
diff --git a/configs/samples/pjsip.conf.sample b/configs/samples/pjsip.conf.sample
index 0fa1a9d..6f83f86 100644
--- a/configs/samples/pjsip.conf.sample
+++ b/configs/samples/pjsip.conf.sample
@@ -910,6 +910,18 @@
; with us. The extension added is the name of the endpoint.
;regcontext=sipregistrations
+;ignore_uri_user_options=no ; Enable/Disable ignoring SIP URI user field options.
+ ; If you have this option enabled and there are semicolons
+ ; in the user field of a SIP URI then the field is truncated
+ ; at the first semicolon. This effectively makes the semicolon
+ ; a non-usable character for PJSIP endpoint names, extensions,
+ ; and AORs. This can be useful for improving compatability with
+ ; an ITSP that likes to use user options for whatever reason.
+ ; Example:
+ ; URI: "sip:1235557890;phone-context=national at x.x.x.x;user=phone"
+ ; The user field is "1235557890;phone-context=national"
+ ; Which becomes this: "1235557890"
+
; MODULE PROVIDING BELOW SECTION(S): res_pjsip_acl
;==========================ACL SECTION OPTIONS=========================
;[acl]
diff --git a/contrib/ast-db-manage/config/versions/a6ef36f1309_ps_globals_add_ignore_uri_user_options.py b/contrib/ast-db-manage/config/versions/a6ef36f1309_ps_globals_add_ignore_uri_user_options.py
new file mode 100644
index 0000000..2ce40a0
--- /dev/null
+++ b/contrib/ast-db-manage/config/versions/a6ef36f1309_ps_globals_add_ignore_uri_user_options.py
@@ -0,0 +1,32 @@
+"""ps_globals add ignore_uri_user_options
+
+Revision ID: a6ef36f1309
+Revises: 4e2493ef32e6
+Create Date: 2016-08-31 12:24:22.368956
+
+"""
+
+# revision identifiers, used by Alembic.
+revision = 'a6ef36f1309'
+down_revision = '4e2493ef32e6'
+
+from alembic import op
+import sqlalchemy as sa
+from sqlalchemy.dialects.postgresql import ENUM
+
+YESNO_NAME = 'yesno_values'
+YESNO_VALUES = ['yes', 'no']
+
+def upgrade():
+ ############################# Enums ##############################
+
+ # yesno_values have already been created, so use postgres enum object
+ # type to get around "already created" issue - works okay with mysql
+ yesno_values = ENUM(*YESNO_VALUES, name=YESNO_NAME, create_type=False)
+
+ op.add_column('ps_globals', sa.Column('ignore_uri_user_options', yesno_values))
+
+
+def downgrade():
+ op.drop_column('ps_globals', 'ignore_uri_user_options')
+
diff --git a/include/asterisk/res_pjsip.h b/include/asterisk/res_pjsip.h
index 4475f8d..68f9544 100644
--- a/include/asterisk/res_pjsip.h
+++ b/include/asterisk/res_pjsip.h
@@ -2193,6 +2193,38 @@
void ast_sip_unregister_supplement(struct ast_sip_supplement *supplement);
/*!
+ * \brief Retrieve the global setting 'ignore_uri_user_options'.
+ * \since 13.12.0
+ *
+ * \retval non zero if ignore the user field options.
+ */
+unsigned int ast_sip_get_ignore_uri_user_options(void);
+
+/*!
+ * \brief Truncate the URI user field options string if enabled.
+ * \since 13.12.0
+ *
+ * \param str URI user field string to truncate if enabled
+ *
+ * \details
+ * We need to be able to handle URI's looking like
+ * "sip:1235557890;phone-context=national at x.x.x.x;user=phone"
+ *
+ * Where the URI user field is:
+ * "1235557890;phone-context=national"
+ *
+ * When truncated the string will become:
+ * "1235557890"
+ */
+#define AST_SIP_USER_OPTIONS_TRUNCATE_CHECK(str) \
+ do { \
+ char *__semi = strchr((str), ';'); \
+ if (__semi && ast_sip_get_ignore_uri_user_options()) { \
+ *__semi = '\0'; \
+ } \
+ } while (0)
+
+/*!
* \brief Retrieve the system debug setting (yes|no|host).
*
* \note returned string needs to be de-allocated by caller.
diff --git a/res/res_pjsip.c b/res/res_pjsip.c
index 5fda931..ac403a6 100644
--- a/res/res_pjsip.c
+++ b/res/res_pjsip.c
@@ -1330,6 +1330,27 @@
set to this value if there is no better option (such as CallerID) to be
used.</synopsis>
</configOption>
+ <configOption name="ignore_uri_user_options">
+ <synopsis>Enable/Disable ignoring SIP URI user field options.</synopsis>
+ <description>
+ <para>If you have this option enabled and there are semicolons
+ in the user field of a SIP URI then the field is truncated
+ at the first semicolon. This effectively makes the semicolon
+ a non-usable character for PJSIP endpoint names, extensions,
+ and AORs. This can be useful for improving compatability with
+ an ITSP that likes to use user options for whatever reason.
+ </para>
+ <example title="Sample SIP URI">
+ sip:1235557890;phone-context=national at x.x.x.x;user=phone
+ </example>
+ <example title="Sample SIP URI user field">
+ 1235557890;phone-context=national
+ </example>
+ <example title="Sample SIP URI user field truncated">
+ 1235557890
+ </example>
+ </description>
+ </configOption>
</configObject>
</configFile>
</configInfo>
diff --git a/res/res_pjsip/config_global.c b/res/res_pjsip/config_global.c
index ca608dc..7a34876 100644
--- a/res/res_pjsip/config_global.c
+++ b/res/res_pjsip/config_global.c
@@ -37,6 +37,7 @@
#define DEFAULT_FROM_USER "asterisk"
#define DEFAULT_REGCONTEXT ""
#define DEFAULT_DISABLE_MULTI_DOMAIN 0
+#define DEFAULT_IGNORE_URI_USER_OPTIONS 0
static char default_useragent[256];
@@ -61,6 +62,8 @@
unsigned int max_initial_qualify_time;
/*! Nonzero to disable multi domain support */
unsigned int disable_multi_domain;
+ /*! Nonzero if URI user field options are ignored. */
+ unsigned int ignore_uri_user_options;
};
static void global_destructor(void *obj)
@@ -232,6 +235,21 @@
}
}
+unsigned int ast_sip_get_ignore_uri_user_options(void)
+{
+ unsigned int ignore_uri_user_options;
+ struct global_config *cfg;
+
+ cfg = get_global_cfg();
+ if (!cfg) {
+ return DEFAULT_IGNORE_URI_USER_OPTIONS;
+ }
+
+ ignore_uri_user_options = cfg->ignore_uri_user_options;
+ ao2_ref(cfg, -1);
+ return ignore_uri_user_options;
+}
+
/*!
* \internal
* \brief Observer to set default global object if none exist.
@@ -351,6 +369,9 @@
OPT_STRINGFIELD_T, 0, STRFLDSET(struct global_config, regcontext));
ast_sorcery_object_field_register(sorcery, "global", "disable_multi_domain", "no",
OPT_BOOL_T, 1, FLDSET(struct global_config, disable_multi_domain));
+ ast_sorcery_object_field_register(sorcery, "global", "ignore_uri_user_options",
+ DEFAULT_IGNORE_URI_USER_OPTIONS ? "yes" : "no",
+ OPT_BOOL_T, 1, FLDSET(struct global_config, ignore_uri_user_options));
if (ast_sorcery_instance_observer_add(sorcery, &observer_callbacks_global)) {
return -1;
diff --git a/res/res_pjsip/pjsip_options.c b/res/res_pjsip/pjsip_options.c
index 80fa1c2..15158bc 100644
--- a/res/res_pjsip/pjsip_options.c
+++ b/res/res_pjsip/pjsip_options.c
@@ -701,8 +701,7 @@
pjsip_sip_uri *sip_ruri;
char exten[AST_MAX_EXTENSION];
- if (pjsip_method_cmp(&rdata->msg_info.msg->line.req.method,
- &pjsip_options_method)) {
+ if (pjsip_method_cmp(&rdata->msg_info.msg->line.req.method, &pjsip_options_method)) {
return PJ_FALSE;
}
@@ -719,13 +718,20 @@
sip_ruri = pjsip_uri_get_uri(ruri);
ast_copy_pj_str(exten, &sip_ruri->user, sizeof(exten));
+ /*
+ * We may want to match in the dialplan without any user
+ * options getting in the way.
+ */
+ AST_SIP_USER_OPTIONS_TRUNCATE_CHECK(exten);
+
if (ast_shutting_down()) {
/*
* Not taking any new calls at this time.
* Likely a server availability OPTIONS poll.
*/
send_options_response(rdata, 503);
- } else if (!ast_strlen_zero(exten) && !ast_exists_extension(NULL, endpoint->context, exten, 1, NULL)) {
+ } else if (!ast_strlen_zero(exten)
+ && !ast_exists_extension(NULL, endpoint->context, exten, 1, NULL)) {
send_options_response(rdata, 404);
} else {
send_options_response(rdata, 200);
diff --git a/res/res_pjsip_endpoint_identifier_user.c b/res/res_pjsip_endpoint_identifier_user.c
index 10b08af..4ad10bd 100644
--- a/res/res_pjsip_endpoint_identifier_user.c
+++ b/res/res_pjsip_endpoint_identifier_user.c
@@ -33,6 +33,7 @@
{
pjsip_uri *from = rdata->msg_info.from->uri;
pjsip_sip_uri *sip_from;
+
if (!PJSIP_URI_SCHEME_IS_SIP(from) && !PJSIP_URI_SCHEME_IS_SIPS(from)) {
return -1;
}
@@ -69,6 +70,12 @@
return NULL;
}
+ /*
+ * We may want to be matched without any user options getting
+ * in the way.
+ */
+ AST_SIP_USER_OPTIONS_TRUNCATE_CHECK(endpoint_name);
+
if (!ast_sip_get_disable_multi_domain()) {
/* Attempt to find the endpoint given the name and domain provided */
snprintf(id, sizeof(id), "%s@%s", endpoint_name, domain_name);
diff --git a/res/res_pjsip_messaging.c b/res/res_pjsip_messaging.c
index 594c0fd..edc66bf 100644
--- a/res/res_pjsip_messaging.c
+++ b/res/res_pjsip_messaging.c
@@ -133,6 +133,12 @@
} else if ((aor_uri = strchr(name, '@'))) {
/* format was 'endpoint@' - don't use the rest */
*aor_uri = '\0';
+
+ /*
+ * We may want to match without any user options getting
+ * in the way.
+ */
+ AST_SIP_USER_OPTIONS_TRUNCATE_CHECK(name);
}
/* at this point, if name is not empty then it
@@ -448,6 +454,12 @@
sip_ruri = pjsip_uri_get_uri(ruri);
ast_copy_pj_str(exten, &sip_ruri->user, AST_MAX_EXTENSION);
+ /*
+ * We may want to match in the dialplan without any user
+ * options getting in the way.
+ */
+ AST_SIP_USER_OPTIONS_TRUNCATE_CHECK(exten);
+
endpt = ast_pjsip_rdata_get_endpoint(rdata);
ast_assert(endpt != NULL);
@@ -528,7 +540,7 @@
static struct msg_data* msg_data_create(const struct ast_msg *msg, const char *to, const char *from)
{
- char *tag;
+ char *uri_params;
struct msg_data *mdata = ao2_alloc(sizeof(*mdata), msg_data_destroy);
if (!mdata) {
@@ -553,9 +565,14 @@
return NULL;
}
- /* sometimes from can still contain the tag at this point, so remove it */
- if ((tag = strchr(mdata->from, ';'))) {
- *tag = '\0';
+ /*
+ * Sometimes from URI can contain URI parameters, so remove them.
+ *
+ * sip:user;user-options at domain;uri-parameters
+ */
+ uri_params = strchr(mdata->from, '@');
+ if (uri_params && (uri_params = strchr(mdata->from, ';'))) {
+ *uri_params = '\0';
}
return mdata;
}
diff --git a/res/res_pjsip_path.c b/res/res_pjsip_path.c
index 47d6a79..20c9d43 100644
--- a/res/res_pjsip_path.c
+++ b/res/res_pjsip_path.c
@@ -40,7 +40,8 @@
char *configured_aors, *aor_name;
pjsip_sip_uri *sip_uri;
char *domain_name;
- RAII_VAR(struct ast_str *, id, NULL, ast_free);
+ char *username;
+ struct ast_str *id = NULL;
if (ast_strlen_zero(endpoint->aors)) {
return NULL;
@@ -49,6 +50,14 @@
sip_uri = pjsip_uri_get_uri(uri);
domain_name = ast_alloca(sip_uri->host.slen + 1);
ast_copy_pj_str(domain_name, &sip_uri->host, sip_uri->host.slen + 1);
+ username = ast_alloca(sip_uri->user.slen + 1);
+ ast_copy_pj_str(username, &sip_uri->user, sip_uri->user.slen + 1);
+
+ /*
+ * We may want to match without any user options getting
+ * in the way.
+ */
+ AST_SIP_USER_OPTIONS_TRUNCATE_CHECK(username);
configured_aors = ast_strdupa(endpoint->aors);
@@ -60,15 +69,16 @@
continue;
}
- if (!pj_strcmp2(&sip_uri->user, aor_name)) {
+ if (!strcmp(username, aor_name)) {
break;
}
- if (!id && !(id = ast_str_create(sip_uri->user.slen + sip_uri->host.slen + 2))) {
- return NULL;
+ if (!id && !(id = ast_str_create(strlen(username) + sip_uri->host.slen + 2))) {
+ aor_name = NULL;
+ break;
}
- ast_str_set(&id, 0, "%.*s@", (int)sip_uri->user.slen, sip_uri->user.ptr);
+ ast_str_set(&id, 0, "%s@", username);
if ((alias = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "domain_alias", domain_name))) {
ast_str_append(&id, 0, "%s", alias->domain);
ao2_cleanup(alias);
@@ -77,10 +87,10 @@
}
if (!strcmp(aor_name, ast_str_buffer(id))) {
- ast_free(id);
break;
}
}
+ ast_free(id);
if (ast_strlen_zero(aor_name)) {
return NULL;
diff --git a/res/res_pjsip_pubsub.c b/res/res_pjsip_pubsub.c
index ae75ce7..c24ac01 100644
--- a/res/res_pjsip_pubsub.c
+++ b/res/res_pjsip_pubsub.c
@@ -1400,6 +1400,12 @@
resource = ast_alloca(resource_size);
ast_copy_pj_str(resource, &request_uri->user, resource_size);
+ /*
+ * We may want to match without any user options getting
+ * in the way.
+ */
+ AST_SIP_USER_OPTIONS_TRUNCATE_CHECK(resource);
+
handler = subscription_get_handler_from_rdata(rdata);
if (!handler || !handler->notifier) {
ast_log(LOG_WARNING, "Failed recreating '%s' subscription: Could not get subscription handler.\n",
@@ -2796,6 +2802,12 @@
resource = ast_alloca(resource_size);
ast_copy_pj_str(resource, &request_uri_sip->user, resource_size);
+ /*
+ * We may want to match without any user options getting
+ * in the way.
+ */
+ AST_SIP_USER_OPTIONS_TRUNCATE_CHECK(resource);
+
expires_header = pjsip_msg_find_hdr(rdata->msg_info.msg, PJSIP_H_EXPIRES, rdata->msg_info.msg->hdr.next);
if (expires_header) {
@@ -3009,6 +3021,12 @@
resource_name = ast_alloca(resource_size);
ast_copy_pj_str(resource_name, &request_uri_sip->user, resource_size);
+ /*
+ * We may want to match without any user options getting
+ * in the way.
+ */
+ AST_SIP_USER_OPTIONS_TRUNCATE_CHECK(resource_name);
+
resource = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "inbound-publication", resource_name);
if (!resource) {
pjsip_endpt_respond_stateless(ast_sip_get_pjsip_endpoint(), rdata, 404, NULL, NULL, NULL);
diff --git a/res/res_pjsip_refer.c b/res/res_pjsip_refer.c
index f3af65c..f586718 100644
--- a/res/res_pjsip_refer.c
+++ b/res/res_pjsip_refer.c
@@ -809,6 +809,13 @@
/* Using the user portion of the target URI see if it exists as a valid extension in their context */
ast_copy_pj_str(exten, &target->user, sizeof(exten));
+
+ /*
+ * We may want to match in the dialplan without any user
+ * options getting in the way.
+ */
+ AST_SIP_USER_OPTIONS_TRUNCATE_CHECK(exten);
+
if (!ast_exists_extension(NULL, context, exten, 1, NULL)) {
ast_log(LOG_ERROR, "Channel '%s' from endpoint '%s' attempted blind transfer to '%s@%s' but target does not exist\n",
ast_channel_name(session->channel), ast_sorcery_object_get_id(session->endpoint), exten, context);
diff --git a/res/res_pjsip_registrar.c b/res/res_pjsip_registrar.c
index 29ee077..099ccc0 100644
--- a/res/res_pjsip_registrar.c
+++ b/res/res_pjsip_registrar.c
@@ -519,6 +519,7 @@
struct ast_sip_aor *aor = NULL;
pjsip_sip_uri *uri;
char *domain_name;
+ char *username;
char *configured_aors;
char *aor_name;
struct ast_str *id = NULL;
@@ -526,6 +527,14 @@
uri = pjsip_uri_get_uri(rdata->msg_info.to->uri);
domain_name = ast_alloca(uri->host.slen + 1);
ast_copy_pj_str(domain_name, &uri->host, uri->host.slen + 1);
+ username = ast_alloca(uri->user.slen + 1);
+ ast_copy_pj_str(username, &uri->user, uri->user.slen + 1);
+
+ /*
+ * We may want to match without any user options getting
+ * in the way.
+ */
+ AST_SIP_USER_OPTIONS_TRUNCATE_CHECK(username);
configured_aors = ast_strdupa(endpoint->aors);
@@ -537,16 +546,16 @@
continue;
}
- if (!pj_strcmp2(&uri->user, aor_name)) {
+ if (!strcmp(username, aor_name)) {
break;
}
- if (!id && !(id = ast_str_create(uri->user.slen + uri->host.slen + 2))) {
+ if (!id && !(id = ast_str_create(strlen(username) + uri->host.slen + 2))) {
aor_name = NULL;
break;
}
- ast_str_set(&id, 0, "%.*s@", (int)uri->user.slen, uri->user.ptr);
+ ast_str_set(&id, 0, "%s@", username);
if ((alias = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "domain_alias", domain_name))) {
ast_str_append(&id, 0, "%s", alias->domain);
ao2_cleanup(alias);
diff --git a/res/res_pjsip_session.c b/res/res_pjsip_session.c
index a4108d5..c04d019 100644
--- a/res/res_pjsip_session.c
+++ b/res/res_pjsip_session.c
@@ -1952,6 +1952,12 @@
sip_ruri = pjsip_uri_get_uri(ruri);
ast_copy_pj_str(session->exten, &sip_ruri->user, sizeof(session->exten));
+ /*
+ * We may want to match in the dialplan without any user
+ * options getting in the way.
+ */
+ AST_SIP_USER_OPTIONS_TRUNCATE_CHECK(session->exten);
+
pickup_cfg = ast_get_chan_features_pickup_config(session->channel);
if (!pickup_cfg) {
ast_log(LOG_ERROR, "Unable to retrieve pickup configuration options. Unable to detect call pickup extension\n");
@@ -2903,6 +2909,13 @@
char exten[AST_MAX_EXTENSION];
ast_copy_pj_str(exten, &uri->user, sizeof(exten));
+
+ /*
+ * We may want to match in the dialplan without any user
+ * options getting in the way.
+ */
+ AST_SIP_USER_OPTIONS_TRUNCATE_CHECK(exten);
+
ast_channel_call_forward_set(session->channel, exten);
} else if (session->endpoint->redirect_method == AST_SIP_REDIRECT_URI_CORE) {
char target_uri[PJSIP_MAX_URL_SIZE];
--
To view, visit https://gerrit.asterisk.org/3860
To unsubscribe, visit https://gerrit.asterisk.org/settings
Gerrit-MessageType: newchange
Gerrit-Change-Id: Ib42b0e940dd34d84c7b14bc2e90d1ba392624f62
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: certified/13.8
Gerrit-Owner: Richard Mudgett <rmudgett at digium.com>
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