[asterisk-commits] res pjsip: Add ignore uri user options option. (asterisk[certified/13.8])
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Wed Sep 14 12:08:17 CDT 2016
Anonymous Coward #1000019 has submitted this change and it was merged.
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.
* Made the caller-id number and redirecting number strings obtained from
incoming SIP URI user fields always truncated at the first semicolon.
People don't care about anything after the semicolon showing up on their
displays even though the RFC allows the semicolon.
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_caller_id.c
M res/res_pjsip_diversion.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
16 files changed, 286 insertions(+), 18 deletions(-)
Approvals:
Mark Michelson: Looks good to me, approved
George Joseph: Looks good to me, but someone else must approve
Anonymous Coward #1000019: Verified
diff --git a/CHANGES b/CHANGES
index fed0d40..0fac84e 100644
--- a/CHANGES
+++ b/CHANGES
@@ -9,6 +9,23 @@
==============================================================================
------------------------------------------------------------------------------
+--- 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"
+
+ Note: The caller-id and redirecting number strings obtained from incoming
+ SIP URI user fields are now always truncated at the first semicolon.
+
+------------------------------------------------------------------------------
--- 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..4b8f385 100644
--- a/configs/samples/pjsip.conf.sample
+++ b/configs/samples/pjsip.conf.sample
@@ -910,6 +910,22 @@
; 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"
+ ;
+ ; Note: The caller-id and redirecting number strings obtained
+ ; from incoming SIP URI user fields are always truncated at the
+ ; first semicolon.
+
; 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..269bfe3 100644
--- a/res/res_pjsip.c
+++ b/res/res_pjsip.c
@@ -1330,6 +1330,30 @@
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>
+ <note><para>The caller-id and redirecting number strings
+ obtained from incoming SIP URI user fields are always truncated
+ at the first semicolon.</para></note>
+ </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_caller_id.c b/res/res_pjsip_caller_id.c
index efa1b89..4a5fc8d 100644
--- a/res/res_pjsip_caller_id.c
+++ b/res/res_pjsip_caller_id.c
@@ -46,11 +46,29 @@
char cid_num[AST_CHANNEL_NAME];
pjsip_sip_uri *uri;
pjsip_name_addr *id_name_addr = (pjsip_name_addr *) hdr->uri;
+ char *semi;
uri = pjsip_uri_get_uri(id_name_addr);
ast_copy_pj_str(cid_name, &id_name_addr->display, sizeof(cid_name));
ast_copy_pj_str(cid_num, &uri->user, sizeof(cid_num));
+ /* Always truncate caller-id number at a semicolon. */
+ semi = strchr(cid_num, ';');
+ if (semi) {
+ /*
+ * 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 will result in:
+ * "1235557890;phone-context=national"
+ *
+ * People don't care about anything after the semicolon
+ * showing up on their displays even though the RFC
+ * allows the semicolon.
+ */
+ *semi = '\0';
+ }
+
ast_free(id->name.str);
id->name.str = ast_strdup(cid_name);
if (!ast_strlen_zero(cid_name)) {
diff --git a/res/res_pjsip_diversion.c b/res/res_pjsip_diversion.c
index 41e6c82..c328842 100644
--- a/res/res_pjsip_diversion.c
+++ b/res/res_pjsip_diversion.c
@@ -148,11 +148,32 @@
struct ast_set_party_id *update)
{
pjsip_sip_uri *uri = pjsip_uri_get_uri(name_addr->uri);
+ char *semi;
+ pj_str_t uri_user;
- if (pj_strlen(&uri->user)) {
+ uri_user = uri->user;
+
+ /* Always truncate redirecting number at a semicolon. */
+ semi = pj_strchr(&uri_user, ';');
+ if (semi) {
+ /*
+ * 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 will result in:
+ * "1235557890;phone-context=national"
+ *
+ * People don't care about anything after the semicolon
+ * showing up on their displays even though the RFC
+ * allows the semicolon.
+ */
+ pj_strset(&uri_user, (char *) pj_strbuf(&uri_user), semi - pj_strbuf(&uri_user));
+ }
+
+ if (pj_strlen(&uri_user)) {
update->number = 1;
data->number.valid = 1;
- set_redirecting_value(&data->number.str, &uri->user);
+ set_redirecting_value(&data->number.str, &uri_user);
}
if (pj_strlen(&name_addr->display)) {
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: merged
Gerrit-Change-Id: Ib42b0e940dd34d84c7b14bc2e90d1ba392624f62
Gerrit-PatchSet: 2
Gerrit-Project: asterisk
Gerrit-Branch: certified/13.8
Gerrit-Owner: Richard Mudgett <rmudgett at digium.com>
Gerrit-Reviewer: Anonymous Coward #1000019
Gerrit-Reviewer: George Joseph <gjoseph at digium.com>
Gerrit-Reviewer: Mark Michelson <mmichelson at digium.com>
Gerrit-Reviewer: Scott Griepentrog <sgriepentrog at digium.com>
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