[svn-commits] file: trunk r425804 - in /trunk: ./ contrib/ast-db-manage/config/versions/ in...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Fri Oct 17 06:34:44 CDT 2014
Author: file
Date: Fri Oct 17 06:30:23 2014
New Revision: 425804
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=425804
Log:
res_pjsip: Add 'user_eq_phone' option to add a 'user=phone' parameter when applicable.
This change adds a configuration option which adds a 'user=phone' parameter if the user
portion of the request URI or the From URI is determined to be a number.
Review: https://reviewboard.asterisk.org/r/4073/
Added:
trunk/contrib/ast-db-manage/config/versions/371a3bf4143e_add_user_eq_phone_option_to_pjsip.py (with props)
Modified:
trunk/CHANGES
trunk/include/asterisk/res_pjsip.h
trunk/res/res_pjsip.c
trunk/res/res_pjsip/pjsip_configuration.c
trunk/res/res_pjsip_caller_id.c
Modified: trunk/CHANGES
URL: http://svnview.digium.com/svn/asterisk/trunk/CHANGES?view=diff&rev=425804&r1=425803&r2=425804
==============================================================================
--- trunk/CHANGES (original)
+++ trunk/CHANGES Fri Oct 17 06:30:23 2014
@@ -21,6 +21,10 @@
ipaddress to bind the rtpengine to. For example, chan_sip might bind
to eth0 (10.0.0.2) but rtpengine to eth1 (192.168.1.10).
+chan_pjsip
+------------------
+ * New 'user_eq_phone' endpoint setting. This adds a 'user=phone' parameter
+ to the request URI and From URI if the user is determined to be a phone number.
Functions
------------------
Added: trunk/contrib/ast-db-manage/config/versions/371a3bf4143e_add_user_eq_phone_option_to_pjsip.py
URL: http://svnview.digium.com/svn/asterisk/trunk/contrib/ast-db-manage/config/versions/371a3bf4143e_add_user_eq_phone_option_to_pjsip.py?view=auto&rev=425804
==============================================================================
--- trunk/contrib/ast-db-manage/config/versions/371a3bf4143e_add_user_eq_phone_option_to_pjsip.py (added)
+++ trunk/contrib/ast-db-manage/config/versions/371a3bf4143e_add_user_eq_phone_option_to_pjsip.py Fri Oct 17 06:30:23 2014
@@ -1,0 +1,30 @@
+"""add user_eq_phone option to pjsip
+
+Revision ID: 371a3bf4143e
+Revises: 10aedae86a32
+Create Date: 2014-10-13 13:46:24.474675
+
+"""
+
+# revision identifiers, used by Alembic.
+revision = '371a3bf4143e'
+down_revision = '10aedae86a32'
+
+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_endpoints', sa.Column('user_eq_phone', yesno_values))
+
+def downgrade():
+ op.drop_column('ps_endpoints', 'user_eq_phone')
Propchange: trunk/contrib/ast-db-manage/config/versions/371a3bf4143e_add_user_eq_phone_option_to_pjsip.py
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: trunk/contrib/ast-db-manage/config/versions/371a3bf4143e_add_user_eq_phone_option_to_pjsip.py
------------------------------------------------------------------------------
svn:keywords = Author Date Id Revision
Propchange: trunk/contrib/ast-db-manage/config/versions/371a3bf4143e_add_user_eq_phone_option_to_pjsip.py
------------------------------------------------------------------------------
svn:mime-type = text/plain
Modified: trunk/include/asterisk/res_pjsip.h
URL: http://svnview.digium.com/svn/asterisk/trunk/include/asterisk/res_pjsip.h?view=diff&rev=425804&r1=425803&r2=425804
==============================================================================
--- trunk/include/asterisk/res_pjsip.h (original)
+++ trunk/include/asterisk/res_pjsip.h Fri Oct 17 06:30:23 2014
@@ -607,6 +607,8 @@
enum ast_sip_session_redirect redirect_method;
/*! Variables set on channel creation */
struct ast_variable *channel_vars;
+ /*! Whether to place a 'user=phone' parameter into the request URI if user is a number */
+ unsigned int usereqphone;
};
/*!
@@ -1482,6 +1484,15 @@
* \return The looked up endpoint
*/
struct ast_sip_endpoint *ast_pjsip_rdata_get_endpoint(pjsip_rx_data *rdata);
+
+/*!
+ * \brief Add 'user=phone' parameter to URI if enabled and user is a phone number.
+ *
+ * \param endpoint The endpoint to use for configuration
+ * \param pool The memory pool to allocate the parameter from
+ * \param uri The URI to check for user and to add parameter to
+ */
+void ast_sip_add_usereqphone(const struct ast_sip_endpoint *endpoint, pj_pool_t *pool, pjsip_uri *uri);
/*!
* \brief Retrieve any endpoints available to sorcery.
Modified: trunk/res/res_pjsip.c
URL: http://svnview.digium.com/svn/asterisk/trunk/res/res_pjsip.c?view=diff&rev=425804&r1=425803&r2=425804
==============================================================================
--- trunk/res/res_pjsip.c (original)
+++ trunk/res/res_pjsip.c Fri Oct 17 06:30:23 2014
@@ -35,6 +35,7 @@
#include "asterisk/taskprocessor.h"
#include "asterisk/uuid.h"
#include "asterisk/sorcery.h"
+#include "asterisk/file.h"
/*** MODULEINFO
<depend>pjproject</depend>
@@ -573,6 +574,9 @@
<configOption name="allow_transfer" default="yes">
<synopsis>Determines whether SIP REFER transfers are allowed for this endpoint</synopsis>
</configOption>
+ <configOption name="user_eq_phone" default="no">
+ <synopsis>Determines whether a user=phone parameter is placed into the request URI if the user is determined to be a phone number</synopsis>
+ </configOption>
<configOption name="sdp_owner" default="-">
<synopsis>String placed as the username portion of an SDP origin (o=) line.</synopsis>
</configOption>
@@ -1544,6 +1548,9 @@
</parameter>
<parameter name="AllowTransfer">
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='allow_transfer']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="UserEqPhone">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='user_eq_phone']/synopsis/node())"/></para>
</parameter>
<parameter name="SdpOwner">
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='sdp_owner']/synopsis/node())"/></para>
@@ -2104,6 +2111,41 @@
return 0;
}
+void ast_sip_add_usereqphone(const struct ast_sip_endpoint *endpoint, pj_pool_t *pool, pjsip_uri *uri)
+{
+ pjsip_sip_uri *sip_uri;
+ int i = 0;
+ pjsip_param *param;
+ const pj_str_t STR_USER = { "user", 4 };
+ const pj_str_t STR_PHONE = { "phone", 5 };
+
+ if (!endpoint || !endpoint->usereqphone || (!PJSIP_URI_SCHEME_IS_SIP(uri) && !PJSIP_URI_SCHEME_IS_SIPS(uri))) {
+ return;
+ }
+
+ sip_uri = pjsip_uri_get_uri(uri);
+
+ if (!pj_strlen(&sip_uri->user)) {
+ return;
+ }
+
+ /* Test URI user against allowed characters in AST_DIGIT_ANY */
+ for (; i < pj_strlen(&sip_uri->user); i++) {
+ if (!strchr(AST_DIGIT_ANYNUM, pj_strbuf(&sip_uri->user)[i])) {
+ break;
+ }
+ }
+
+ if (i < pj_strlen(&sip_uri->user)) {
+ return;
+ }
+
+ param = PJ_POOL_ALLOC_T(pool, pjsip_param);
+ param->name = STR_USER;
+ param->value = STR_PHONE;
+ pj_list_insert_before(&sip_uri->other_param, param);
+}
+
pjsip_dialog *ast_sip_create_dialog_uac(const struct ast_sip_endpoint *endpoint, const char *uri, const char *request_user)
{
char enclosed_uri[PJSIP_MAX_URL_SIZE];
@@ -2150,6 +2192,9 @@
pj_strdup2(dlg->pool, &sip_uri->user, request_user);
}
}
+
+ /* Add the user=phone parameter if applicable */
+ ast_sip_add_usereqphone(endpoint, dlg->pool, dlg->target);
/* We have to temporarily bump up the sess_count here so the dialog is not prematurely destroyed */
dlg->sess_count++;
@@ -2349,6 +2394,9 @@
pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
return -1;
}
+
+ /* Add the user=phone parameter if applicable */
+ ast_sip_add_usereqphone(endpoint, (*tdata)->pool, (*tdata)->msg->line.req.uri);
/* If an outbound proxy is specified on the endpoint apply it to this request */
if (endpoint && !ast_strlen_zero(endpoint->outbound_proxy) &&
Modified: trunk/res/res_pjsip/pjsip_configuration.c
URL: http://svnview.digium.com/svn/asterisk/trunk/res/res_pjsip/pjsip_configuration.c?view=diff&rev=425804&r1=425803&r2=425804
==============================================================================
--- trunk/res/res_pjsip/pjsip_configuration.c (original)
+++ trunk/res/res_pjsip/pjsip_configuration.c Fri Oct 17 06:30:23 2014
@@ -1732,6 +1732,7 @@
ast_sorcery_object_field_register(sip_sorcery, "endpoint", "record_on_feature", "automixmon", OPT_STRINGFIELD_T, 0, STRFLDSET(struct ast_sip_endpoint, info.recording.onfeature));
ast_sorcery_object_field_register(sip_sorcery, "endpoint", "record_off_feature", "automixmon", OPT_STRINGFIELD_T, 0, STRFLDSET(struct ast_sip_endpoint, info.recording.offfeature));
ast_sorcery_object_field_register(sip_sorcery, "endpoint", "allow_transfer", "yes", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, allowtransfer));
+ ast_sorcery_object_field_register(sip_sorcery, "endpoint", "user_eq_phone", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, usereqphone));
ast_sorcery_object_field_register(sip_sorcery, "endpoint", "sdp_owner", "-", OPT_STRINGFIELD_T, 0, STRFLDSET(struct ast_sip_endpoint, media.sdpowner));
ast_sorcery_object_field_register(sip_sorcery, "endpoint", "sdp_session", "Asterisk", OPT_STRINGFIELD_T, 0, STRFLDSET(struct ast_sip_endpoint, media.sdpsession));
ast_sorcery_object_field_register_custom(sip_sorcery, "endpoint", "tos_audio", "0", tos_handler, tos_audio_to_str, NULL, 0, 0);
Modified: trunk/res/res_pjsip_caller_id.c
URL: http://svnview.digium.com/svn/asterisk/trunk/res/res_pjsip_caller_id.c?view=diff&rev=425804&r1=425803&r2=425804
==============================================================================
--- trunk/res/res_pjsip_caller_id.c (original)
+++ trunk/res/res_pjsip_caller_id.c Fri Oct 17 06:30:23 2014
@@ -669,11 +669,7 @@
ast_party_id_copy(&connected_id, &effective_id);
ast_channel_unlock(session->channel);
- if (session->inv_session->state < PJSIP_INV_STATE_CONFIRMED &&
- ast_strlen_zero(session->endpoint->fromuser) &&
- (session->endpoint->id.trust_outbound ||
- ((connected_id.name.presentation & AST_PRES_RESTRICTION) == AST_PRES_ALLOWED &&
- (connected_id.number.presentation & AST_PRES_RESTRICTION) == AST_PRES_ALLOWED))) {
+ if (session->inv_session->state < PJSIP_INV_STATE_CONFIRMED) {
/* Only change the From header on the initial outbound INVITE. Switching it
* mid-call might confuse some UAs.
*/
@@ -683,8 +679,16 @@
from = pjsip_msg_find_hdr(tdata->msg, PJSIP_H_FROM, tdata->msg->hdr.next);
dlg = session->inv_session->dlg;
- modify_id_header(tdata->pool, from, &connected_id);
- modify_id_header(dlg->pool, dlg->local.info, &connected_id);
+ if (ast_strlen_zero(session->endpoint->fromuser) &&
+ (session->endpoint->id.trust_outbound ||
+ ((connected_id.name.presentation & AST_PRES_RESTRICTION) == AST_PRES_ALLOWED &&
+ (connected_id.number.presentation & AST_PRES_RESTRICTION) == AST_PRES_ALLOWED))) {
+ modify_id_header(tdata->pool, from, &connected_id);
+ modify_id_header(dlg->pool, dlg->local.info, &connected_id);
+ }
+
+ ast_sip_add_usereqphone(session->endpoint, tdata->pool, from->uri);
+ ast_sip_add_usereqphone(session->endpoint, dlg->pool, dlg->local.info->uri);
}
add_id_headers(session, tdata, &connected_id);
ast_party_id_free(&connected_id);
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