[Asterisk-code-review] res_pjsip_messaging: Refactor outgoing URI processing (asterisk[16])
George Joseph
asteriskteam at digium.com
Tue Apr 27 12:27:48 CDT 2021
George Joseph has uploaded this change for review. ( https://gerrit.asterisk.org/c/asterisk/+/15828 )
Change subject: res_pjsip_messaging: Refactor outgoing URI processing
......................................................................
res_pjsip_messaging: Refactor outgoing URI processing
* Implemented the new "to" parameter of the MessageSend()
dialplan application. This allows a user to specify
a complete SIP "To" header separate from the Request URI.
* Completely refactored the get_outbound_endpoint() function
to actually handle all the destination combinations that
we advertized as supporting.
* Added lots of debugging.
ASTERISK-29404
Reported by Brian J. Murrell
Change-Id: I67a485196d9199916468f7f98bfb9a0b993a4cce
---
M res/res_pjsip_messaging.c
1 file changed, 444 insertions(+), 103 deletions(-)
git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/28/15828/1
diff --git a/res/res_pjsip_messaging.c b/res/res_pjsip_messaging.c
index 9287324..8a01e9c 100644
--- a/res/res_pjsip_messaging.c
+++ b/res/res_pjsip_messaging.c
@@ -25,15 +25,83 @@
/*** DOCUMENTATION
<info name="MessageDestinationInfo" language="en_US" tech="PJSIP">
- <para>Specifying a prefix of <literal>pjsip:</literal> will send the
- message as a SIP MESSAGE request.</para>
+ <para>The <literal>destination</literal> parameter is used to construct
+ the Request URI for an outgoing message. It can be in one of the following
+ formats, all prefixed with the <literal>pjsip:</literal> message tech.</para>
+ <para>
+ </para>
+ <enumlist>
+ <enum name="endpoint">
+ <para>Request URI comes from the endpoint's default aor and contact.</para>
+ </enum>
+ <enum name="endpoint/aor">
+ <para>Request URI comes from the specific aor/contact.</para>
+ </enum>
+ <enum name="endpoint at domain">
+ <para>Request URI from the endpoint's default aor and contact. The domain is discarded.</para>
+ </enum>
+ </enumlist>
+ <para>
+ </para>
+ <para>These all use the endpoint to send the message with the specified URI:</para>
+ <para>
+ </para>
+ <enumlist>
+ <enum name="endpoint/<sip[s]:host>>"/>
+ <enum name="endpoint/<sip[s]:user at host>"/>
+ <enum name="endpoint/"display name" <sip[s]:host>"/>
+ <enum name="endpoint/"display name" <sip[s]:user at host>"/>
+ <enum name="endpoint/sip[s]:host"/>
+ <enum name="endpoint/sip[s]:user at host"/>
+ <enum name="endpoint/host"/>
+ <enum name="endpoint/user at host"/>
+ </enumlist>
+ <para>
+ </para>
+ <para>These all use the default endpoint to send the message with the specified URI:</para>
+ <para>
+ </para>
+ <enumlist>
+ <enum name="<sip[s]:host>"/>
+ <enum name="<sip[s]:user at host>"/>
+ <enum name=""display name" <sip[s]:host>"/>
+ <enum name=""display name" <sip[s]:user at host>"/>
+ <enum name="sip[s]:host"/>
+ <enum name="sip[s]:user at host"/>
+ </enumlist>
+ <para>
+ </para>
+ <para>These use the default endpoint to send the message with the specified host:</para>
+ <para>
+ </para>
+ <enumlist>
+ <enum name="host"/>
+ <enum name="user at host"/>
+ </enumlist>
+ <para>
+ </para>
+ <para>In all cases URIs not already prefixed with <literal>sip:</literal> or
+ <literal>sips:</literal> will automatically be prefixed with <literal>sip:</literal>.</para>
+ <para>
+ </para>
</info>
<info name="MessageFromInfo" language="en_US" tech="PJSIP">
- <para>The <literal>from</literal> parameter can be a configured endpoint
- or in the form of "display-name" <URI>.</para>
+ <para>The <literal>from</literal> parameter is used to specity the <literal>From:</literal>
+ header in the outgoing SIP MESSAGE. It will override the value specified in
+ MESSAGE(from) which itself will override any <literal>from</literal> value from
+ an incoming SIP MESSAGE.
+ </para>
+ <para>
+ </para>
</info>
<info name="MessageToInfo" language="en_US" tech="PJSIP">
- <para>Ignored</para>
+ <para>The <literal>to</literal> parameter is used to specity the <literal>To:</literal>
+ header in the outgoing SIP MESSAGE. It will override the value specified in
+ MESSAGE(to) which itself will override any <literal>to</literal> value from
+ an incoming SIP MESSAGE.
+ </para>
+ <para>
+ </para>
</info>
***/
#include "asterisk.h"
@@ -47,6 +115,7 @@
#include "asterisk/res_pjsip.h"
#include "asterisk/res_pjsip_session.h"
#include "asterisk/taskprocessor.h"
+#include "asterisk/uri.h"
const pjsip_method pjsip_message_method = {PJSIP_OTHER_METHOD, {"MESSAGE", 7} };
@@ -114,133 +183,347 @@
/*!
* \internal
- * \brief Puts pointer past 'sip[s]:' string that should be at the
- * front of the given 'fromto' parameter
+ * \brief Retrieves an endpoint and URI from the "to" string.
*
- * \param fromto 'From' or 'To' field containing 'sip:'
- */
-static const char *skip_sip(const char *fromto)
-{
- const char *p;
-
- /* need to be one past 'sip:' or 'sips:' */
- if (!(p = strstr(fromto, "sip"))) {
- return fromto;
- }
-
- p += 3;
- if (*p == 's') {
- ++p;
- }
-
- return ++p;
-}
-
-/*!
- * \internal
- * \brief Retrieves an endpoint if specified in the given 'to'
+ * This URI is used as the Request URI.
*
* Expects the given 'to' to be in one of the following formats:
- * sip[s]:endpoint[/aor]
- * sip[s]:endpoint[/uri] - Where uri is: sip[s]:user at domain
- * sip[s]:endpoint[@domain]
- * sip[s]:unknown_user at domain <-- will use default outbound endpoint
+ * Why we allow so many is a mystery.
*
- * If an optional aor is given it will try to find an associated uri
- * to return. If an optional uri is given then that will be returned,
- * otherwise uri will be NULL.
+ * endpoint - We'll get URI from the default aor/contact
+ * endpoint/aor - We'll get the URI from the specific aor/contact
+ * endpoint at domain - We toss the domain part and just use the endpoint
*
- * \param to 'From' or 'To' field with possible endpoint
- * \param uri Optional uri to return
+ * These all use the endpoint and specified URI
+ * endpoint/<sip[s]:host>
+ * endpoint/<sip[s]:user at host>
+ * endpoint/"Bob" <sip[s]:host>
+ * endpoint/"Bob" <sip[s]:user at host>
+ * endpoint/sip[s]:host
+ * endpoint/sip[s]:user at host
+ * endpoint/host
+ * endpoint/user at host
+ *
+ * These all use the default endpoint and specified URI
+ * <sip[s]:host>
+ * <sip[s]:user at host>
+ * "Bob" <sip[s]:host>
+ * "Bob" <sip[s]:user at host>
+ * sip[s]:host
+ * sip[s]:user at host
+ *
+ * These use the default endpoint and specified host
+ * host
+ * user at host
+ *
+ * The ones that have the sip[s] scheme are the easiest to parse.
+ * The rest all have some issue.
+ *
+ * endpoint vs host : We have to test for endpoint first
+ * endpoint/aor vs endpoint/host : We have to test for aor first
+ * What if there's an aor with the same
+ * name as the host?
+ * endpoint at domain vs user at host : We have to test for endpoint first.
+ * What if there's an endpoint with the
+ * same name as the user?
+ *
+ * \param to 'To' field with possible endpoint
+ * \param uri Pointer to a char* which will be set to the URI.
+ * Must be ast_free'd by the caller.
+ *
+ * \note The logic below could probably be condensed but then it wouldn't be
+ * as clear.
*/
static struct ast_sip_endpoint *get_outbound_endpoint(const char *to, char **uri)
{
- char *name;
- char *aor_uri;
- struct ast_sip_endpoint *endpoint;
+ char *destination;
+ char *endpoint_name = NULL;
+ char *slash = NULL;
+ char *atsign = NULL;
+ char *scheme = NULL;
+ struct ast_sip_endpoint *endpoint = NULL;
- name = ast_strdupa(skip_sip(to));
+ destination = ast_strdupa(to);
- /* attempt to extract the endpoint name */
- if ((aor_uri = strchr(name, '/'))) {
- /* format was 'endpoint/(aor_name | uri)' */
- *aor_uri++ = '\0';
- } else if ((aor_uri = strchr(name, '@'))) {
- /* format was 'endpoint at domain' - discard the domain */
- *aor_uri = '\0';
+ slash = strchr(destination, '/');
+ atsign = strchr(destination, '@');
+ scheme = S_OR(strstr(destination, "sip:"), strstr(destination, "sips:"));
+
+ /* If it's just one word, it could be an endpoint name or just a hostname */
+ if (!slash && !atsign && !scheme) {
+ endpoint = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint", destination);
+ if (endpoint) {
+ /* It's an endpoint */
+ struct ast_sip_contact *contact = NULL;
+
+ endpoint_name = destination;
+ contact = ast_sip_location_retrieve_contact_from_aor_list(endpoint->aors);
+ if (!contact) {
+ /*
+ * We're getting the contact using the same method as
+ * ast_sip_create_request() so if there's no contact
+ * we can never send this message.
+ */
+ ao2_cleanup(endpoint);
+ endpoint = NULL;
+ *uri = NULL;
+ ast_debug(3, "Dest: FAIL: '%s' Found endpoint '%s' but didn't find an aor/contact for it\n",
+ to, endpoint_name);
+ } else {
+ ao2_cleanup(contact);
+ *uri = ast_strdup(contact->uri);
+ ast_debug(3, "Dest: '%s' Found endpoint '%s' and found contact with URI '%s'\n",
+ to, endpoint_name, *uri);
+ }
+ } else {
+ /*
+ * It's a hostname that we need to add a scheme to
+ */
+ char *temp_uri = ast_malloc(strlen(destination) + strlen("sip:") + 1);
+
+ sprintf(temp_uri, "sip:%s", destination);
+ *uri = temp_uri;
+ endpoint = ast_sip_default_outbound_endpoint();
+ ast_debug(3, "Dest: '%s' Didn't find endpoint so adding scheme and using URI '%s' with default endpoint\n",
+ to, *uri);
+ }
+ /* There's nothing left to parse so... */
+ return endpoint;
+ }
+
+ /* If there's a '/', it MUST be following an endpoint name */
+ if (slash) {
+ char *afterslash = slash + 1;
+ *slash = '\0';
+ endpoint_name = destination;
+ endpoint = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint", endpoint_name);
+ /*
+ * If there's a slash but there's isn't a valid endpoint name
+ * before it, there's nothing else we can do.
+ */
+ if (!endpoint) {
+ *uri = NULL;
+ ast_debug(3, "Dest: '%s' FAIL: Didn't find endpoint with name '%s' before the slash\n",
+ to, endpoint_name);
+
+ return NULL;
+ }
/*
- * We may want to match without any user options getting
- * in the way.
+ * If we found a scheme, then everything after the slash MUST be a URI.
+ * We don't need to do any further modification.
*/
- AST_SIP_USER_OPTIONS_TRUNCATE_CHECK(name);
+ if (scheme) {
+ *uri = ast_strdup(afterslash);
+ ast_debug(3, "Dest: '%s' Found endpoint '%s' and found URI '%s' after slash\n",
+ to, endpoint_name, *uri);
+ } else {
+ /* It now may be an aor or a URI without a scheme */
+ struct ast_sip_aor *aor;
+ struct ast_sip_contact *contact = NULL;
+
+ aor = ast_sip_location_retrieve_aor(afterslash);
+ if (aor) {
+ ast_debug(3, "Dest: '%s' Found endpoint '%s' and found aor '%s' after slash\n",
+ to, endpoint_name, ast_sorcery_object_get_id(aor));
+
+ contact = ast_sip_location_retrieve_first_aor_contact(aor);
+ if (contact) {
+ *uri = ast_strdup(contact->uri);
+ ast_debug(3, "Dest: '%s' Found endpoint '%s' and found contact with URI '%s' for aor '%s'\n",
+ to, endpoint_name, *uri, ast_sorcery_object_get_id(aor));
+ } else {
+ /*
+ * An aor without a contact is useless and since
+ * ast_sip_create_message() won't be able to find one
+ * either, we just need to bail.
+ */
+ ao2_cleanup(endpoint);
+ endpoint = NULL;
+ *uri = NULL;
+ ast_debug(3, "Dest: '%s' FAIL: Found endpoint '%s' but didn't find contact for aor '%s'\n",
+ to, endpoint_name, ast_sorcery_object_get_id(aor));
+ }
+ ao2_cleanup(contact);
+ ao2_cleanup(aor);
+ } else {
+ /*
+ * It's probably a uri without a scheme but we don't have a way to tell.
+ * We're going to assume it is and prepend it with a scheme.
+ */
+ *uri = ast_malloc(strlen(afterslash) + strlen("sip:") + 1);
+ sprintf(*uri, "sip:%s", afterslash);
+ ast_debug(3, "Dest: '%s' Found endpoint '%s' but didn't find aor after slash so using URI '%s'\n",
+ to, endpoint_name, *uri);
+ }
+ }
+
+ return endpoint;
}
- /* at this point, if name is not empty then it
- might be an endpoint, so try to retrieve it */
- if (ast_strlen_zero(name)
- || !(endpoint = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint",
- name))) {
- /* an endpoint was not found, so assume sending directly
- to a uri and use the default outbound endpoint */
- *uri = ast_strdup(to);
- return ast_sip_default_outbound_endpoint();
+ /*
+ * If there's an '@' but no scheme then it's either following an endpoint name
+ * and being followed by a domain name (which we discard).
+ * OR is's a user at host uri without a scheme. It's probably the latter but because
+ * endpoint at domain looks just line user at host, we'll test for endpoint first.
+ */
+ if (!endpoint && atsign && !scheme) {
+ char *afterat = atsign + 1;
+ *atsign = '\0';
+ endpoint_name = destination;
+
+ /* Apprently there may be ';<user_options>' after the endpoint name ??? */
+ AST_SIP_USER_OPTIONS_TRUNCATE_CHECK(endpoint_name);
+ endpoint = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint", endpoint_name);
+ /* If it's an endpoint, just return it because we discard the domain. */
+ if (endpoint) {
+ struct ast_sip_contact *contact =
+ ast_sip_location_retrieve_contact_from_aor_list(endpoint->aors);
+ if (!contact) {
+ /*
+ * We're getting the contact using the same method as
+ * ast_sip_create_request() so if there's no contact
+ * we can never send this message.
+ */
+ ao2_cleanup(endpoint);
+ endpoint = NULL;
+ *uri = NULL;
+ ast_debug(3, "Dest: '%s' FAIL: Found endpoint '%s' but didn't find contact\n",
+ to, endpoint_name);
+ } else {
+ *uri = ast_strdup(contact->uri);
+ ao2_cleanup(contact);
+ ast_debug(3, "Dest: '%s' Found endpoint '%s' and found contact with URI '%s' (discarding domain %s)\n",
+ to, endpoint_name, *uri, afterat);
+ }
+ } else {
+ /*
+ * It's probably a uri with a user but without a scheme but we don't have a way to tell.
+ * We're going to assume it is and prepend it with a scheme.
+ */
+ *uri = ast_malloc(strlen(to) + strlen("sip:") + 1);
+ sprintf(*uri, "sip:%s", to);
+ endpoint = ast_sip_default_outbound_endpoint();
+ ast_debug(3, "Dest: '%s' Didn't find endpoint before the '@' so using URI '%s' with default endpoint\n",
+ to, *uri);
+ }
+
+ return endpoint;
}
- if (ast_strlen_zero(aor_uri)) {
- *uri = NULL;
+ /*
+ * If all else fails, we assume it's a URI or just a hostname.
+ */
+ if (scheme) {
+ *uri = ast_strdup(destination);
+ ast_debug(3, "Dest: '%s' Didn't find an endpoint but did find a scheme so using URI '%s' with default endpoint\n",
+ to, *uri);
} else {
- struct ast_sip_aor *aor;
- struct ast_sip_contact *contact = NULL;
- char *end;
-
- /* Trim off any stray angle bracket that shouldn't be here */
- end = strchr(aor_uri, '>');
- if (end) {
- *end = '\0';
- }
-
- /*
- * if what's in 'uri' is a retrievable aor use the uri on it
- * instead, otherwise assume what's there is already a uri
- */
- aor = ast_sip_location_retrieve_aor(aor_uri);
- if (aor && (contact = ast_sip_location_retrieve_first_aor_contact(aor))) {
- aor_uri = (char *) contact->uri;
- }
- /* need to copy because underlying uri goes away */
- *uri = ast_strdup(aor_uri);
-
- ao2_cleanup(contact);
- ao2_cleanup(aor);
+ *uri = ast_malloc(strlen(destination) + strlen("sip:") + 1);
+ sprintf(*uri, "sip:%s", destination);
+ ast_debug(3, "Dest: '%s' Didn't find an endpoint and didn't find scheme so adding scheme and using URI '%s' with default endpoint\n",
+ to, *uri);
}
+ endpoint = ast_sip_default_outbound_endpoint();
return endpoint;
}
/*!
* \internal
- * \brief Overwrite fields in the outbound 'To' header
- *
- * Updates display name in an outgoing To header.
+ * \brief Replace the To URI in the tdata with the supplied one
*
* \param tdata the outbound message data structure
- * \param to info to copy into the header
+ * \param to URI to replace the To URI with
+ *
+ * \return 0: success, -1: failure
*/
-static void update_to(pjsip_tx_data *tdata, char *to)
+static int update_to_uri(pjsip_tx_data *tdata, char *to)
+{
+ pjsip_name_addr *parsed_name_addr;
+ pjsip_sip_uri *sip_uri;
+ pjsip_name_addr *tdata_name_addr;
+ pjsip_sip_uri *tdata_sip_uri;
+ char *buf = NULL;
+#define DEBUG_BUF_SIZE 256
+
+ parsed_name_addr = (pjsip_name_addr *) pjsip_parse_uri(tdata->pool, to, strlen(to),
+ PJSIP_PARSE_URI_AS_NAMEADDR);
+
+ if (!parsed_name_addr || (!PJSIP_URI_SCHEME_IS_SIP(parsed_name_addr->uri)
+ && !PJSIP_URI_SCHEME_IS_SIPS(parsed_name_addr->uri))) {
+ ast_log(LOG_WARNING, "To address '%s' is not a valid SIP/SIPS URI\n", to);
+ return -1;
+ }
+
+ sip_uri = pjsip_uri_get_uri(parsed_name_addr->uri);
+ if (DEBUG_ATLEAST(3)) {
+ buf = ast_alloca(DEBUG_BUF_SIZE);
+ pjsip_uri_print(PJSIP_URI_IN_FROMTO_HDR, sip_uri, buf, DEBUG_BUF_SIZE);
+ ast_debug(3, "Parsed To: %.*s %s\n", (int)parsed_name_addr->display.slen,
+ parsed_name_addr->display.ptr, buf);
+ }
+
+ tdata_name_addr = (pjsip_name_addr *) PJSIP_MSG_TO_HDR(tdata->msg)->uri;
+ if (!tdata_name_addr || (!PJSIP_URI_SCHEME_IS_SIP(tdata_name_addr->uri)
+ && !PJSIP_URI_SCHEME_IS_SIPS(tdata_name_addr->uri))) {
+ /* Highly unlikely but we have to check */
+ ast_log(LOG_WARNING, "tdata To address '%s' is not a valid SIP/SIPS URI\n", to);
+ return -1;
+ }
+
+ tdata_sip_uri = pjsip_uri_get_uri(tdata_name_addr->uri);
+ if (DEBUG_ATLEAST(3)) {
+ buf[0] = '\0';
+ pjsip_uri_print(PJSIP_URI_IN_FROMTO_HDR, tdata_sip_uri, buf, DEBUG_BUF_SIZE);
+ ast_debug(3, "Original tdata To: %.*s %s\n", (int)tdata_name_addr->display.slen,
+ tdata_name_addr->display.ptr, buf);
+ }
+
+ /* Replace the uri */
+ pjsip_sip_uri_assign(tdata->pool, tdata_sip_uri, sip_uri);
+ /* The display name isn't part of the URI so we need to replace it separately */
+ pj_strdup(tdata->pool, &tdata_name_addr->display, &parsed_name_addr->display);
+
+ if (DEBUG_ATLEAST(3)) {
+ buf[0] = '\0';
+ pjsip_uri_print(PJSIP_URI_IN_FROMTO_HDR, tdata_sip_uri, buf, 256);
+ ast_debug(3, "New tdata To: %.*s %s\n", (int)tdata_name_addr->display.slen,
+ tdata_name_addr->display.ptr, buf);
+ }
+
+ return 0;
+#undef DEBUG_BUF_SIZE
+}
+
+/*!
+ * \internal
+ * \brief Update the display name in the To uri in the tdata with the one from the supplied uri
+ *
+ * \param tdata the outbound message data structure
+ * \param to uri containing the display name to replace in the the To uri
+ *
+ * \return 0: success, -1: failure
+ */
+static int update_to_display_name(pjsip_tx_data *tdata, char *to)
{
pjsip_name_addr *parsed_name_addr;
parsed_name_addr = (pjsip_name_addr *) pjsip_parse_uri(tdata->pool, to, strlen(to),
PJSIP_PARSE_URI_AS_NAMEADDR);
+
if (parsed_name_addr) {
if (pj_strlen(&parsed_name_addr->display)) {
pjsip_name_addr *name_addr =
(pjsip_name_addr *) PJSIP_MSG_TO_HDR(tdata->msg)->uri;
pj_strdup(tdata->pool, &name_addr->display, &parsed_name_addr->display);
+
}
+ return 0;
}
+
+ return -1;
}
/*!
@@ -254,15 +537,17 @@
*
* \param tdata the outbound message data structure
* \param from info to copy into the header
+ *
+ * \return 0: success, -1: failure
*/
-static void update_from(pjsip_tx_data *tdata, char *from)
+static int update_from(pjsip_tx_data *tdata, char *from)
{
pjsip_name_addr *name_addr;
pjsip_sip_uri *uri;
pjsip_name_addr *parsed_name_addr;
if (ast_strlen_zero(from)) {
- return;
+ return 0;
}
name_addr = (pjsip_name_addr *) PJSIP_MSG_FROM_HDR(tdata->msg)->uri;
@@ -276,7 +561,7 @@
if (!PJSIP_URI_SCHEME_IS_SIP(parsed_name_addr->uri)
&& !PJSIP_URI_SCHEME_IS_SIPS(parsed_name_addr->uri)) {
ast_log(LOG_WARNING, "From address '%s' is not a valid SIP/SIPS URI\n", from);
- return;
+ return -1;
}
parsed_uri = pjsip_uri_get_uri(parsed_name_addr->uri);
@@ -285,9 +570,12 @@
pj_strdup(tdata->pool, &name_addr->display, &parsed_name_addr->display);
}
+ /* Unlike the To header, we only want to replace the user, host and port */
pj_strdup(tdata->pool, &uri->user, &parsed_uri->user);
pj_strdup(tdata->pool, &uri->host, &parsed_uri->host);
uri->port = parsed_uri->port;
+
+ return 0;
} else {
/* assume it is 'user[@domain]' format */
char *domain = strchr(from, '@');
@@ -302,7 +590,11 @@
} else {
pj_strdup2(tdata->pool, &uri->user, from);
}
+
+ return 0;
}
+
+ return -1;
}
/*!
@@ -618,13 +910,8 @@
}
++to;/* Now skip the ':' */
- /* Make sure we start with sip: */
- mdata->to = ast_begins_with(to, "sip:") ? ast_strdup(to) : ast_strdup(to - 4);
+ mdata->to = ast_strdup(to);
mdata->from = ast_strdup(from);
- if (!mdata->to || !mdata->from) {
- ao2_ref(mdata, -1);
- return NULL;
- }
/*
* Sometimes from URI can contain URI parameters, so remove them.
@@ -667,6 +954,25 @@
}
}
+/*!
+ * \internal
+ * \brief Send a MESSAGE
+ *
+ * \param mdata The outbound message data structure
+ *
+ * \return 0: success, -1: failure
+ *
+ * mdata contains the To and From specified in the call to the MessageSend
+ * dialplan app. It also contains the ast_msg object that contains the
+ * message body and may contain the To and From from the channel datastore,
+ * usually set with the MESSAGE or MESSAGE_DATA dialplan functions but
+ * could also come from an incoming sip MESSAGE.
+ *
+ * The mdata->to is always used as the basis for the Request URI
+ * while the mdata->msg->to is used for the To header. If
+ * mdata->msg->to isn't available, mdata->to is used for the To header.
+ *
+ */
static int msg_send(void *data)
{
struct msg_data *mdata = data; /* The caller holds a reference */
@@ -681,6 +987,9 @@
RAII_VAR(char *, uri, NULL, ast_free);
RAII_VAR(struct ast_sip_endpoint *, endpoint, NULL, ao2_cleanup);
+ ast_debug(3, "mdata From: %s msg From: %s mdata Destination: %s msg To: %s\n",
+ mdata->from, ast_msg_get_from(mdata->msg), mdata->to, ast_msg_get_to(mdata->msg));
+
endpoint = get_outbound_endpoint(mdata->to, &uri);
if (!endpoint) {
ast_log(LOG_ERROR,
@@ -689,13 +998,41 @@
return -1;
}
+ ast_debug(3, "Request URI: %s\n", uri);
+
if (ast_sip_create_request("MESSAGE", NULL, endpoint, uri, NULL, &tdata)) {
ast_log(LOG_WARNING, "PJSIP MESSAGE - Could not create request\n");
return -1;
}
- update_to(tdata, mdata->to);
- update_from(tdata, mdata->from);
+ /* If there was a To in the actual message, */
+ if (!ast_strlen_zero(ast_msg_get_to(mdata->msg))) {
+ char *msg_to = ast_strdupa(ast_msg_get_to(mdata->msg));
+
+ /*
+ * It's possible that the message To was copied from
+ * an incoming MESSAGE in which case it'll have the
+ * pjsip: tech prepended to it. We need to remove it.
+ */
+ if (ast_begins_with(msg_to, "pjsip:")) {
+ msg_to += 6;
+ }
+ update_to_uri(tdata, msg_to);
+ } else {
+ /*
+ * If there was no To in the message, it's still possible
+ * that there is a display name in the mdata To. If so,
+ * we'll copy the URI display name to the tdata To.
+ */
+ update_to_display_name(tdata, uri);
+ }
+
+ if (!ast_strlen_zero(mdata->from)) {
+ update_from(tdata, mdata->from);
+ } else if (!ast_strlen_zero(ast_msg_get_from(mdata->msg))) {
+ update_from(tdata, (char *)ast_msg_get_from(mdata->msg));
+ }
+
update_content_type(tdata, mdata->msg, &body);
if (ast_sip_add_body(tdata, &body)) {
@@ -704,10 +1041,14 @@
return -1;
}
+ /*
+ * This copies any headers set with MESSAGE_DATA() to the
+ * tdata.
+ */
vars_to_headers(mdata->msg, tdata);
ast_debug(1, "Sending message to '%s' (via endpoint %s) from '%s'\n",
- mdata->to, ast_sorcery_object_get_id(endpoint), mdata->from);
+ uri, ast_sorcery_object_get_id(endpoint), mdata->from);
if (ast_sip_send_request(tdata, NULL, endpoint, NULL, NULL)) {
ast_log(LOG_ERROR, "PJSIP MESSAGE - Could not send request\n");
--
To view, visit https://gerrit.asterisk.org/c/asterisk/+/15828
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Gerrit-Project: asterisk
Gerrit-Branch: 16
Gerrit-Change-Id: I67a485196d9199916468f7f98bfb9a0b993a4cce
Gerrit-Change-Number: 15828
Gerrit-PatchSet: 1
Gerrit-Owner: George Joseph <gjoseph at digium.com>
Gerrit-MessageType: newchange
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