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

Browse Source 
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
changes/97/197/1
Matthew Jordan 11 years ago
parent 9f4a357124
commit 3b9a245a29
5 changed files with 76 additions and 8 deletions
Show Stats Download Patch File Download Diff File

6
CHANGES
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@ -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

11
include/asterisk/res_pjsip.h
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@ -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.
*

48
res/res_pjsip.c
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@ -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)) {

1
res/res_pjsip/pjsip_configuration.c
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@ -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);

18
res/res_pjsip_caller_id.c
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@ -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);

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