[asterisk-commits] res pjsip: Add ignore uri user options option. (asterisk[master])
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Wed Sep 14 12:27:28 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, 276 insertions(+), 17 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 bf7bc75..38228e3 100644
--- a/CHANGES
+++ b/CHANGES
@@ -79,6 +79,17 @@
configure these options then you already had to do a reload after making
changes.
+ * 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.
+
app_confbridge
------------------
* Some sounds played into the bridge are played asynchronously. This, for
diff --git a/configs/samples/pjsip.conf.sample b/configs/samples/pjsip.conf.sample
index e6b3249..c6293b6 100644
--- a/configs/samples/pjsip.conf.sample
+++ b/configs/samples/pjsip.conf.sample
@@ -991,6 +991,22 @@
; If disabled then unsolicited mwi will start processing
; on the endpoint's next contact update.
+;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..1556521
--- /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: 7f3e21abe318
+Create Date: 2016-08-31 12:24:22.368956
+
+"""
+
+# revision identifiers, used by Alembic.
+revision = 'a6ef36f1309'
+down_revision = '7f3e21abe318'
+
+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 8a5ad29..92bdabb 100644
--- a/include/asterisk/res_pjsip.h
+++ b/include/asterisk/res_pjsip.h
@@ -2447,6 +2447,38 @@
unsigned int ast_sip_get_mwi_disable_initial_unsolicited(void);
/*!
+ * \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 7bb10c0..d1ebf64 100644
--- a/res/res_pjsip.c
+++ b/res/res_pjsip.c
@@ -1550,6 +1550,30 @@
</para>
</description>
</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 281630a..fc1227d 100644
--- a/res/res_pjsip/config_global.c
+++ b/res/res_pjsip/config_global.c
@@ -47,6 +47,7 @@
#define DEFAULT_MWI_TPS_QUEUE_HIGH AST_TASKPROCESSOR_HIGH_WATER_LEVEL
#define DEFAULT_MWI_TPS_QUEUE_LOW -1
#define DEFAULT_MWI_DISABLE_INITIAL_UNSOLICITED 0
+#define DEFAULT_IGNORE_URI_USER_OPTIONS 0
/*!
* \brief Cached global config object
@@ -100,6 +101,8 @@
/*! Nonzero to disable sending unsolicited mwi to all endpoints on startup */
unsigned int disable_initial_unsolicited;
} mwi;
+ /*! Nonzero if URI user field options are ignored. */
+ unsigned int ignore_uri_user_options;
};
static void global_destructor(void *obj)
@@ -384,6 +387,20 @@
return disable_initial_unsolicited;
}
+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
@@ -533,6 +550,9 @@
ast_sorcery_object_field_register(sorcery, "global", "mwi_disable_initial_unsolicited",
DEFAULT_MWI_DISABLE_INITIAL_UNSOLICITED ? "yes" : "no",
OPT_BOOL_T, 1, FLDSET(struct global_config, mwi.disable_initial_unsolicited));
+ 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 a282224..09fe155 100644
--- a/res/res_pjsip/pjsip_options.c
+++ b/res/res_pjsip/pjsip_options.c
@@ -750,8 +750,7 @@
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;
}
@@ -768,13 +767,20 @@
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 429cb61..16b19ec 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 82c3caa..301d9fc 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 6aa2c55..369cb62 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;
}
@@ -115,18 +116,25 @@
static struct ast_sip_endpoint *username_identify(pjsip_rx_data *rdata)
{
- char username[64], domain[64];
+ char username[64];
+ char domain[64];
struct ast_sip_endpoint *endpoint;
if (get_from_header(rdata, username, sizeof(username), domain, sizeof(domain))) {
return NULL;
}
+
+ /*
+ * We may want to be matched without any user options getting
+ * in the way.
+ */
+ AST_SIP_USER_OPTIONS_TRUNCATE_CHECK(username);
+
ast_debug(3, "Attempting identify by From username '%s' domain '%s'\n", username, domain);
endpoint = find_endpoint(rdata, username, domain);
if (!endpoint) {
ast_debug(3, "Endpoint not found for From username '%s' domain '%s'\n", username, domain);
- ao2_cleanup(endpoint);
return NULL;
}
if (!(endpoint->ident_method & AST_SIP_ENDPOINT_IDENTIFY_BY_USERNAME)) {
diff --git a/res/res_pjsip_messaging.c b/res/res_pjsip_messaging.c
index 7efb1a2..e63c825 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 2dde732..e170a75 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 fe16c61..015ef99 100644
--- a/res/res_pjsip_pubsub.c
+++ b/res/res_pjsip_pubsub.c
@@ -1378,6 +1378,12 @@
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",
@@ -2750,6 +2756,12 @@
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) {
@@ -2963,6 +2975,12 @@
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) {
ast_debug(1, "No 'inbound-publication' defined for resource '%s'\n", resource_name);
diff --git a/res/res_pjsip_refer.c b/res/res_pjsip_refer.c
index 19367bf..c1dee82 100644
--- a/res/res_pjsip_refer.c
+++ b/res/res_pjsip_refer.c
@@ -814,6 +814,13 @@
/* 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 fd87ef7..a8d2bdc 100644
--- a/res/res_pjsip_registrar.c
+++ b/res/res_pjsip_registrar.c
@@ -626,6 +626,12 @@
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);
+
aor_name = find_aor_name(username, domain_name, endpoint->aors);
if (aor_name) {
ast_debug(3, "Matched aor '%s' by To username\n", aor_name);
diff --git a/res/res_pjsip_session.c b/res/res_pjsip_session.c
index a26359f..7e885c3 100644
--- a/res/res_pjsip_session.c
+++ b/res/res_pjsip_session.c
@@ -1982,6 +1982,12 @@
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");
@@ -3095,6 +3101,13 @@
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/3856
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Gerrit-MessageType: merged
Gerrit-Change-Id: Ib42b0e940dd34d84c7b14bc2e90d1ba392624f62
Gerrit-PatchSet: 2
Gerrit-Project: asterisk
Gerrit-Branch: master
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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