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Commit 8e2672d2 authored by George Joseph's avatar George Joseph
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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.

 * We now also accept a destination in the same format
   as Dial()...  PJSIP/number@endpoint

 * Added lots of debugging.

ASTERISK-29404
Reported by Brian J. Murrell

Change-Id: I67a485196d9199916468f7f98bfb9a0b993a4cce
parent 9106c9d1
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doc/CHANGES-staging/res_pjsip_messaging.txt 0 → 100644
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View file @ 8e2672d2
Subject: res_pjsip_messaging
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.
We now also accept a destination in the same format
as Dial()... PJSIP/number@endpoint
res/res_pjsip_messaging.c
+ 732
− 100
View file @ 8e2672d2
......@@ -25,15 +25,96 @@
/*** 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@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/&lt;sip[s]:host&gt;>"/>
<enum name="endpoint/&lt;sip[s]:user@host&gt;"/>
<enum name="endpoint/&quot;display name&quot; &lt;sip[s]:host&gt;"/>
<enum name="endpoint/&quot;display name&quot; &lt;sip[s]:user@host&gt;"/>
<enum name="endpoint/sip[s]:host"/>
<enum name="endpoint/sip[s]:user@host"/>
<enum name="endpoint/host"/>
<enum name="endpoint/user@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="&lt;sip[s]:host&gt;"/>
<enum name="&lt;sip[s]:user@host&gt;"/>
<enum name="&quot;display name&quot; &lt;sip[s]:host&gt;"/>
<enum name="&quot;display name&quot; &lt;sip[s]:user@host&gt;"/>
<enum name="sip[s]:host"/>
<enum name="sip[s]:user@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@host"/>
</enumlist>
<para>
</para>
<para>This form is similar to a dialstring:</para>
<para>
</para>
<enumlist>
<enum name="PJSIP/user@endpoint"/>
</enumlist>
<para>
</para>
<para>You still need to prefix the destination with
the <literal>pjsip:</literal> message technology prefix. For example:
<literal>pjsip:PJSIP/8005551212@myprovider</literal>.
The endpoint contact's URI will have the <literal>user</literal> inserted
into it and will become the Request URI. If the contact URI already has
a user specified, an error is returned.
</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" &lt;URI&gt;.</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 +128,8 @@
#include "asterisk/res_pjsip.h"
#include "asterisk/res_pjsip_session.h"
#include "asterisk/taskprocessor.h"
#include "asterisk/test.h"
#include "asterisk/uri.h"
const pjsip_method pjsip_message_method = {PJSIP_OTHER_METHOD, {"MESSAGE", 7} };
......@@ -113,134 +196,579 @@ static enum pjsip_status_code check_content_type_in_dialog(const pjsip_rx_data *
}
/*!
* \internal
* \brief Puts pointer past 'sip[s]:' string that should be at the
* front of the given 'fromto' parameter
* \brief Find a contact and insert a "user@" into its URI.
*
* \param to Original destination (for error messages only)
* \param endpoint_name Endpoint name (for error messages only)
* \param aors Command separated list of AORs
* \param user The user to insert in the contact URI
* \param uri Pointer to buffer in which to return the URI
*
* \param fromto 'From' or 'To' field containing 'sip:'
* \return 0 Success
* \return -1 Fail
*
* \note If the contact URI found for the endpoint already has a user in
* its URI, replacing it is probably not a good idea so an error is returned.
*/
static const char *skip_sip(const char *fromto)
static int insert_user_in_contact_uri(const char *to, const char *endpoint_name, const char *aors,
const char *user, char **uri)
{
const char *p;
char *atsign = NULL;
char *scheme = NULL;
char *contact_uri = NULL;
char *colon = NULL;
char *host;
struct ast_sip_contact *contact = NULL;
/* need to be one past 'sip:' or 'sips:' */
if (!(p = strstr(fromto, "sip"))) {
return fromto;
contact = ast_sip_location_retrieve_contact_from_aor_list(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.
*/
ast_log(LOG_WARNING, "Dest: '%s' MSG SEND FAIL: Couldn't find contact for endpoint '%s'\n",
to, endpoint_name);
return -1;
}
p += 3;
if (*p == 's') {
++p;
contact_uri = ast_strdupa(contact->uri);
ao2_cleanup(contact);
ast_debug(3, "Dest: '%s' User: '%s' Endpoint: '%s' ContactURI: '%s'\n", to, user, endpoint_name, contact_uri);
atsign = strchr(contact_uri, '@');
if (atsign) {
/*
* If there is already a username in the contact URI
* messing with it is probably NOT a good thing.
*/
ast_log(LOG_WARNING, "Dest: '%s' MSG SEND FAIL: There's already a username in endpoint %s's contact URI '%s'.\n",
to, endpoint_name, contact_uri);
return -1;
}
return ++p;
/*
* Contact URIs must have a scheme so we must insert the user between it and the host.
*/
colon = strchr(contact_uri, ':');
if (!colon) {
/* A contact URI without a scheme? Something's wrong. Bail */
ast_log(LOG_WARNING, "Dest: '%s' MSG SEND FAIL: There was no scheme in the contact URI '%s'\n",
to, contact_uri);
return -1;
}
host = colon + 1;
scheme = contact_uri;
*uri = ast_malloc(strlen(contact_uri) + strlen(user) + 2 /* One for the @ and one for the NULL */);
/*
* Need to set the NULL after the malloc or the length of contact_uri will be too short
* to hold the final result.
*/
*colon = '\0';
sprintf(*uri, "%s:%s@%s", scheme, user, host);
return 0;
}
/*!
* \internal
* \brief Retrieves an endpoint if specified in the given 'to'
* \brief Get endpoint and URI when the destination is only a single token
*
* 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@domain
* sip[s]:endpoint[@domain]
* sip[s]:unknown_user@domain <-- will use default outbound endpoint
*
* 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.
* "to" could be one of the following:
* endpoint_name
* hostname
*
* \param to 'From' or 'To' field with possible endpoint
* \param uri Optional uri to return
* \param to Destination specified in MessageSend
* \param uri Pointer to URI variable. Must be freed by caller
* \return endpoint
*/
static struct ast_sip_endpoint *get_outbound_endpoint(const char *to, char **uri)
{
char *name;
char *aor_uri;
struct ast_sip_endpoint *endpoint;
static struct ast_sip_endpoint *handle_single_token(const char *to, char *destination, char **uri) {
char *endpoint_name = NULL;
struct ast_sip_endpoint *endpoint = NULL;
struct ast_sip_contact *contact = NULL;
/*
* If "to" is just one token, it could be an endpoint name
* or a hostname without a scheme.
*/
name = ast_strdupa(skip_sip(to));
endpoint = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint", destination);
if (!endpoint) {
/*
* We can only assume it's a hostname.
*/
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);
return endpoint;
}
/* 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@domain' - discard the domain */
*aor_uri = '\0';
/*
* It's an endpoint
*/
endpoint_name = destination;
contact = ast_sip_location_retrieve_contact_from_aor_list(endpoint->aors);
if (!contact) {
/*
* We may want to match without any user options getting
* in the way.
* 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.
*/
AST_SIP_USER_OPTIONS_TRUNCATE_CHECK(name);
ast_log(LOG_WARNING, "Dest: '%s' MSG SEND FAIL: Found endpoint '%s' but didn't find an aor/contact for it\n",
to, endpoint_name);
ao2_cleanup(endpoint);
*uri = NULL;
return NULL;
}
/* 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();
*uri = ast_strdup(contact->uri);
ast_debug(3, "Dest: '%s' Found endpoint '%s' and found contact with URI '%s'\n",
to, endpoint_name, *uri);
ao2_cleanup(contact);
return endpoint;
}
/*!
* \internal
* \brief Get endpoint and URI when the destination contained a '/'
*
* "to" could be one of the following:
* endpoint/aor
* endpoint/<sip[s]:host>
* endpoint/<sip[s]:user@host>
* endpoint/"Bob" <sip[s]:host>
* endpoint/"Bob" <sip[s]:user@host>
* endpoint/sip[s]:host
* endpoint/sip[s]:user@host
* endpoint/host
* endpoint/user@host
*
* \param to Destination specified in MessageSend
* \param uri Pointer to URI variable. Must be freed by caller
* \return endpoint
*/
static struct ast_sip_endpoint *handle_slash(const char *to, char *destination, char **uri,
char *slash, char *atsign, char *scheme)
{
char *endpoint_name = NULL;
struct ast_sip_endpoint *endpoint = NULL;
struct ast_sip_contact *contact = NULL;
char *user = NULL;
char *afterslash = slash + 1;
struct ast_sip_aor *aor;
if (ast_begins_with(destination, "PJSIP/")) {
ast_debug(3, "Dest: '%s' Dialplan format'\n", to);
/*
* This has to be the form PJSIP/user@endpoint
*/
if (!atsign || strchr(afterslash, '/')) {
/*
* If there's no "user@" or there's a slash somewhere after
* "PJSIP/" then we go no further.
*/
*uri = NULL;
ast_log(LOG_WARNING,
"Dest: '%s' MSG SEND FAIL: Destinations beginning with 'PJSIP/' must be in the form of 'PJSIP/user@endpoint'\n",
to);
return NULL;
}
*atsign = '\0';
user = afterslash;
endpoint_name = atsign + 1;
ast_debug(3, "Dest: '%s' User: '%s' Endpoint: '%s'\n", to, user, endpoint_name);
} else {
/*
* Either...
* endpoint/aor
* endpoint/uri
*/
*slash = '\0';
endpoint_name = destination;
ast_debug(3, "Dest: '%s' Endpoint: '%s'\n", to, endpoint_name);
}
if (ast_strlen_zero(aor_uri)) {
endpoint = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint", endpoint_name);
if (!endpoint) {
*uri = NULL;
} 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';
}
ast_log(LOG_WARNING, "Dest: '%s' MSG SEND FAIL: Didn't find endpoint with name '%s'\n",
to, endpoint_name);
return NULL;
}
if (scheme) {
/*
* If we found a scheme, then everything after the slash MUST be a URI.
* We don't need to do any further modification.
*/
*uri = ast_strdup(afterslash);
ast_debug(3, "Dest: '%s' Found endpoint '%s' and found URI '%s' after '/'\n",
to, endpoint_name, *uri);
return endpoint;
}
if (user) {
/*
* This has to be the form PJSIP/user@endpoint
*/
int rc;
/*
* if what's in 'uri' is a retrievable aor use the uri on it
* instead, otherwise assume what's there is already a uri
* Set the return URI to be the endpoint's contact URI with the user
* portion set to the user that was specified before the endpoint name.
*/
aor = ast_sip_location_retrieve_aor(aor_uri);
if (aor && (contact = ast_sip_location_retrieve_first_aor_contact(aor))) {
aor_uri = (char *) contact->uri;
rc = insert_user_in_contact_uri(to, endpoint_name, endpoint->aors, user, uri);
if (rc != 0) {
/*
* insert_user_in_contact_uri prints the warning message.
*/
ao2_cleanup(endpoint);
endpoint = NULL;
*uri = NULL;
}
/* need to copy because underlying uri goes away */
*uri = ast_strdup(aor_uri);
ast_debug(3, "Dest: '%s' User: '%s' Endpoint: '%s' URI: '%s'\n", to, user,
endpoint_name, *uri);
ao2_cleanup(contact);
return endpoint;
}
/*
* We're now left with two possibilities...
* endpoint/aor
* endpoint/uri-without-scheme
*/
aor = ast_sip_location_retrieve_aor(afterslash);
if (!aor) {
/*
* It's probably a URI without a scheme but we don't have a way to tell
* for sure. 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 '/' so using URI '%s'\n",
to, endpoint_name, *uri);
return endpoint;
}
/*
* Only one possibility left... There was an aor name after the slash.
*/
ast_debug(3, "Dest: '%s' Found endpoint '%s' and found aor '%s' after '/'\n",
to, endpoint_name, ast_sorcery_object_get_id(aor));
contact = ast_sip_location_retrieve_first_aor_contact(aor);
if (!contact) {
/*
* 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.
*/
ast_log(LOG_WARNING, "Dest: '%s' MSG SEND FAIL: Found endpoint '%s' but didn't find contact for aor '%s'\n",
to, endpoint_name, ast_sorcery_object_get_id(aor));
ao2_cleanup(aor);
ao2_cleanup(endpoint);
*uri = NULL;
return NULL;
}
*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));
ao2_cleanup(contact);
ao2_cleanup(aor);
return endpoint;
}
/*!
* \internal
* \brief Get endpoint and URI when the destination contained a '@' but no '/' or scheme
*
* "to" could be one of the following:
* <sip[s]:user@host>
* "Bob" <sip[s]:user@host>
* sip[s]:user@host
* user@host
*
* \param to Destination specified in MessageSend
* \param uri Pointer to URI variable. Must be freed by caller
* \return endpoint
*/
static struct ast_sip_endpoint *handle_atsign(const char *to, char *destination, char **uri,
char *slash, char *atsign, char *scheme)
{
char *endpoint_name = NULL;
struct ast_sip_endpoint *endpoint = NULL;
struct ast_sip_contact *contact = NULL;
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 (!endpoint) {
/*
* 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;
}
/*
* OK, it's an endpoint and a domain (which we ignore)
*/
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_log(LOG_WARNING, "Dest: '%s' MSG SEND FAIL: Found endpoint '%s' but didn't find contact\n",
to, endpoint_name);
return NULL;
}
*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);
return endpoint;
}
/*!
* \internal
* \brief Overwrite fields in the outbound 'To' header
* \brief Retrieves an endpoint and URI from the "to" string.
*
* This URI is used as the Request URI.
*
* Expects the given 'to' to be in one of the following formats:
* Why we allow so many is a mystery.
*
* Basic:
* endpoint - We'll get URI from the default aor/contact
* endpoint/aor - We'll get the URI from the specific aor/contact
* endpoint@domain - We toss the domain part and just use the endpoint
*
* These all use the endpoint and specified URI:
* endpoint/<sip[s]:host>
* endpoint/<sip[s]:user@host>
* endpoint/"Bob" <sip[s]:host>
* endpoint/"Bob" <sip[s]:user@host>
* endpoint/sip[s]:host
* endpoint/sip[s]:user@host
* endpoint/host
* endpoint/user@host
*
* These all use the default endpoint and specified URI:
* <sip[s]:host>
* <sip[s]:user@host>
* "Bob" <sip[s]:host>
* "Bob" <sip[s]:user@host>
* sip[s]:host
* sip[s]:user@host
*
* Updates display name in an outgoing To header.
* These use the default endpoint and specified host:
* host
* user@host
*
* This form is similar to a dialstring:
* PJSIP/user@endpoint
* In this case, the user will be added to the endpoint contact's URI.
* If the contact URI already has a user, an error is returned.
*
* 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@domain vs user@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 *destination;
char *slash = NULL;
char *atsign = NULL;
char *scheme = NULL;
struct ast_sip_endpoint *endpoint = NULL;
destination = ast_strdupa(to);
slash = strchr(destination, '/');
atsign = strchr(destination, '@');
scheme = S_OR(strstr(destination, "sip:"), strstr(destination, "sips:"));
if (!slash && !atsign && !scheme) {
/*
* If there's only a single token, it can be either...
* endpoint
* host
*/
return handle_single_token(to, destination, uri);
}
if (slash) {
/*
* If there's a '/', then the form must be one of the following...
* PJSIP/user@endpoint
* endpoint/aor
* endpoint/uri
*/
return handle_slash(to, destination, uri, slash, atsign, scheme);
}
if (!endpoint && atsign && !scheme) {
/*
* 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@host uri without a scheme. It's probably the latter but because
* endpoint@domain looks just like user@host, we'll test for endpoint first.
*/
return handle_atsign(to, destination, uri, slash, atsign, scheme);
}
/*
* 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 {
*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 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 +782,17 @@ static void update_to(pjsip_tx_data *tdata, char *to)
*
* \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 +806,7 @@ static void update_from(pjsip_tx_data *tdata, char *from)
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 +815,12 @@ static void update_from(pjsip_tx_data *tdata, char *from)
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 +835,11 @@ static void update_from(pjsip_tx_data *tdata, char *from)
} else {
pj_strdup2(tdata->pool, &uri->user, from);
}
return 0;
}
return -1;
}
/*!
......@@ -585,7 +1122,7 @@ static enum pjsip_status_code rx_data_to_ast_msg(pjsip_rx_data *rdata, struct as
struct msg_data {
struct ast_msg *msg;
char *to;
char *destination;
char *from;
};
......@@ -594,12 +1131,12 @@ static void msg_data_destroy(void *obj)
struct msg_data *mdata = obj;
ast_free(mdata->from);
ast_free(mdata->to);
ast_free(mdata->destination);
ast_msg_destroy(mdata->msg);
}
static struct msg_data *msg_data_create(const struct ast_msg *msg, const char *to, const char *from)
static struct msg_data *msg_data_create(const struct ast_msg *msg, const char *destination, const char *from)
{
char *uri_params;
struct msg_data *mdata = ao2_alloc(sizeof(*mdata), msg_data_destroy);
......@@ -612,19 +1149,14 @@ static struct msg_data *msg_data_create(const struct ast_msg *msg, const char *t
mdata->msg = ast_msg_ref((struct ast_msg *) msg);
/* To starts with 'pjsip:' which needs to be removed. */
if (!(to = strchr(to, ':'))) {
if (!(destination = strchr(destination, ':'))) {
ao2_ref(mdata, -1);
return NULL;
}
++to;/* Now skip the ':' */
++destination;/* Now skip the ':' */
/* Make sure we start with sip: */
mdata->to = ast_begins_with(to, "sip:") ? ast_strdup(to) : ast_strdup(to - 4);
mdata->destination = ast_strdup(destination);
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 +1199,25 @@ static void update_content_type(pjsip_tx_data *tdata, struct ast_msg *msg, struc
}
}
/*!
* \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 */
......@@ -681,21 +1232,98 @@ static int msg_send(void *data)
RAII_VAR(char *, uri, NULL, ast_free);
RAII_VAR(struct ast_sip_endpoint *, endpoint, NULL, ao2_cleanup);
endpoint = get_outbound_endpoint(mdata->to, &uri);
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->destination, ast_msg_get_to(mdata->msg));
endpoint = get_outbound_endpoint(mdata->destination, &uri);
if (!endpoint) {
ast_log(LOG_ERROR,
"PJSIP MESSAGE - Could not find endpoint '%s' and no default outbound endpoint configured\n",
mdata->to);
mdata->destination);
ast_test_suite_event_notify("MSG_ENDPOINT_URI_FAIL",
"MdataFrom: %s\r\n"
"MsgFrom: %s\r\n"
"MdataDestination: %s\r\n"
"MsgTo: %s\r\n",
mdata->from,
ast_msg_get_from(mdata->msg),
mdata->destination,
ast_msg_get_to(mdata->msg));
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));
}
#ifdef TEST_FRAMEWORK
{
pjsip_name_addr *tdata_name_addr;
pjsip_sip_uri *tdata_sip_uri;
char touri[128];
char fromuri[128];
tdata_name_addr = (pjsip_name_addr *) PJSIP_MSG_TO_HDR(tdata->msg)->uri;
tdata_sip_uri = pjsip_uri_get_uri(tdata_name_addr->uri);
pjsip_uri_print(PJSIP_URI_IN_FROMTO_HDR, tdata_sip_uri, touri, sizeof(touri));
tdata_name_addr = (pjsip_name_addr *) PJSIP_MSG_FROM_HDR(tdata->msg)->uri;
tdata_sip_uri = pjsip_uri_get_uri(tdata_name_addr->uri);
pjsip_uri_print(PJSIP_URI_IN_FROMTO_HDR, tdata_sip_uri, fromuri, sizeof(fromuri));
ast_test_suite_event_notify("MSG_FROMTO_URI",
"MdataFrom: %s\r\n"
"MsgFrom: %s\r\n"
"MdataDestination: %s\r\n"
"MsgTo: %s\r\n"
"Endpoint: %s\r\n"
"RequestURI: %s\r\n"
"ToURI: %s\r\n"
"FromURI: %s\r\n",
mdata->from,
ast_msg_get_from(mdata->msg),
mdata->destination,
ast_msg_get_to(mdata->msg),
ast_sorcery_object_get_id(endpoint),
uri,
touri,
fromuri
);
}
#endif
update_content_type(tdata, mdata->msg, &body);
if (ast_sip_add_body(tdata, &body)) {
......@@ -704,10 +1332,14 @@ static int msg_send(void *data)
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");
......@@ -717,17 +1349,17 @@ static int msg_send(void *data)
return 0;
}
static int sip_msg_send(const struct ast_msg *msg, const char *to, const char *from)
static int sip_msg_send(const struct ast_msg *msg, const char *destination, const char *from)
{
struct msg_data *mdata;
int res;
if (ast_strlen_zero(to)) {
if (ast_strlen_zero(destination)) {
ast_log(LOG_ERROR, "SIP MESSAGE - a 'To' URI must be specified\n");
return -1;
}
mdata = msg_data_create(msg, to, from);
mdata = msg_data_create(msg, destination, from);
if (!mdata) {
return -1;
}
......
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