res_pjsip: Add 'user_eq_phone' option to add a 'user=phone' parameter when applicable.

Note that this is a backport of r425804 from trunk.

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/

ASTERISK-24643 #close

........

Merged revisions 430083 from http://svn.asterisk.org/svn/asterisk/branches/13

git-svn-id: https://origsvn.digium.com/svn/asterisk/certified/branches/13.1@430085 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/CHANGES b/CHANGES
index f32487f386fd28b8dfedf07e1f18199abbceb79a..e70d81485882ad31a5dd3f91fadcd48a66bb97ba 100644 (file)
--- a/CHANGES
+++ b/CHANGES
@@ -11,6 +11,11 @@
 --- Functionality changes from Asterisk 13.1.0 to Asterisk 13.1-cert1 --------
 ------------------------------------------------------------------------------
 
+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.
+
 ARI
 ------------------
  * The Originate operation now takes in an originator channel. The linked ID of
@@ -106,7 +111,6 @@ included in Asterisk 13.
 Finally, all users upgrading to Asterisk 13 should read the UPGRADE.txt file
 delivered with this release.
 
-
 Build System
 ------------------
  * Sample config files have been moved from configs/ to a sub-folder of that

diff --git a/include/asterisk/res_pjsip.h b/include/asterisk/res_pjsip.h
index 2617e955022c473a5347a74d3bb4c9cb235f774a..95dec46ad9be59cfd18bc7bb0ea25fa5f2722929 100644 (file)
--- a/include/asterisk/res_pjsip.h
+++ b/include/asterisk/res_pjsip.h
@@ -609,6 +609,8 @@ struct ast_sip_endpoint {
        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;
 };
 
 /*!
@@ -1486,6 +1488,15 @@ void ast_copy_pj_str(char *dest, const pj_str_t *src, size_t size);
  */
 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.
  *

diff --git a/res/res_pjsip.c b/res/res_pjsip.c
index f597f27a46f4fabfe59b3708bea3a0ed51b2cb37..481d3b0a44b2133e8868720fbf04e5439b55e891 100644 (file)
--- a/res/res_pjsip.c
+++ b/res/res_pjsip.c
@@ -35,6 +35,7 @@
 #include "asterisk/taskprocessor.h"
 #include "asterisk/uuid.h"
 #include "asterisk/sorcery.h"
+#include "asterisk/file.h"
 
 /*** MODULEINFO
        <depend>pjproject</depend>
@@ -580,6 +581,9 @@
                                <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>
@@ -1568,6 +1572,9 @@
                                <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>
                                </parameter>
@@ -2127,6 +2134,41 @@ static int sip_get_tpselector_from_endpoint(const struct ast_sip_endpoint *endpo
        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];
@@ -2174,6 +2216,9 @@ pjsip_dialog *ast_sip_create_dialog_uac(const struct ast_sip_endpoint *endpoint,
                }
        }
 
+       /* 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++;
 
@@ -2374,6 +2419,9 @@ static int create_out_of_dialog_request(const pjsip_method *method, struct ast_s
                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) &&
                ast_sip_set_outbound_proxy((*tdata), endpoint->outbound_proxy)) {

diff --git a/res/res_pjsip/pjsip_configuration.c b/res/res_pjsip/pjsip_configuration.c
index 63e63fb1405b7880ce44770bc983f5d6f9f15a09..3b58cea49d2abf6545aa0756ef93130b234f550e 100644 (file)
--- a/res/res_pjsip/pjsip_configuration.c
+++ b/res/res_pjsip/pjsip_configuration.c
@@ -1740,6 +1740,7 @@ int ast_res_pjsip_initialize_configuration(const struct ast_module_info *ast_mod
        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);

diff --git a/res/res_pjsip_caller_id.c b/res/res_pjsip_caller_id.c
index e22ce6a09ea46cdb517c62a08faac7719f2c1055..c3757e06f1ee44a03cb14556469cf377ee7c6e2f 100644 (file)
--- a/res/res_pjsip_caller_id.c
+++ b/res/res_pjsip_caller_id.c
@@ -669,11 +669,7 @@ static void caller_id_outgoing_request(struct ast_sip_session *session, pjsip_tx
        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 @@ static void caller_id_outgoing_request(struct ast_sip_session *session, pjsip_tx
                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);