[Asterisk-code-review] res_pjsip_messaging: Refactor outgoing URI processing (asterisk[master])
George Joseph
asteriskteam at digium.com
Tue Jun 8 14:33:17 CDT 2021
George Joseph has submitted this change. ( https://gerrit.asterisk.org/c/asterisk/+/15829 )
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.
* We now also accept a destination in the same format
as Dial()... PJSIP/number at endpoint
* Added lots of debugging.
ASTERISK-29404
Reported by Brian J. Murrell
Change-Id: I67a485196d9199916468f7f98bfb9a0b993a4cce
---
A doc/CHANGES-staging/res_pjsip_messaging.txt
M res/res_pjsip_messaging.c
2 files changed, 740 insertions(+), 101 deletions(-)
Approvals:
Kevin Harwell: Looks good to me, but someone else must approve
Joshua Colp: Looks good to me, but someone else must approve
George Joseph: Looks good to me, approved; Approved for Submit
diff --git a/doc/CHANGES-staging/res_pjsip_messaging.txt b/doc/CHANGES-staging/res_pjsip_messaging.txt
new file mode 100644
index 0000000..46874dc
--- /dev/null
+++ b/doc/CHANGES-staging/res_pjsip_messaging.txt
@@ -0,0 +1,7 @@
+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 at endpoint
diff --git a/res/res_pjsip_messaging.c b/res/res_pjsip_messaging.c
index 9287324..d895945 100644
--- a/res/res_pjsip_messaging.c
+++ b/res/res_pjsip_messaging.c
@@ -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 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>This form is similar to a dialstring:</para>
+ <para>
+ </para>
+ <enumlist>
+ <enum name="PJSIP/user at 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 at 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" <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 +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 @@
}
/*!
- * \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 fromto 'From' or 'To' field containing 'sip:'
+ * \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
+ *
+ * \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 at domain
- * sip[s]:endpoint[@domain]
- * sip[s]:unknown_user at domain <-- will use default outbound endpoint
+ * "to" could be one of the following:
+ * endpoint_name
+ * hostname
*
- * 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.
- *
- * \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;
- name = ast_strdupa(skip_sip(to));
+ /*
+ * If "to" is just one token, it could be an endpoint name
+ * or a hostname without a scheme.
+ */
- /* 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';
-
+ endpoint = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint", destination);
+ if (!endpoint) {
/*
- * We may want to match without any user options getting
- * in the way.
+ * We can only assume it's a hostname.
*/
- AST_SIP_USER_OPTIONS_TRUNCATE_CHECK(name);
+ 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;
}
- /* 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();
- }
+ /*
+ * It's an endpoint
+ */
- if (ast_strlen_zero(aor_uri)) {
+ 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.
+ */
+ 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;
- } else {
- struct ast_sip_aor *aor;
- struct ast_sip_contact *contact = NULL;
- char *end;
+ return NULL;
+ }
- /* Trim off any stray angle bracket that shouldn't be here */
- end = strchr(aor_uri, '>');
- if (end) {
- *end = '\0';
+ *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 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
+ *
+ * \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 at 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 at 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);
+ }
+
+ endpoint = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint", endpoint_name);
+ if (!endpoint) {
+ *uri = NULL;
+ 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 at 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);
- ao2_cleanup(aor);
+ 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 Overwrite fields in the outbound 'To' header
+ * \brief Get endpoint and URI when the destination contained a '@' but no '/' or scheme
*
- * Updates display name in an outgoing To header.
+ * "to" could be one of the following:
+ * <sip[s]:user at host>
+ * "Bob" <sip[s]:user at host>
+ * sip[s]:user at host
+ * user at 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 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 at 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 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
+ *
+ * This form is similar to a dialstring:
+ * PJSIP/user at 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 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 *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 at 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 at host uri without a scheme. It's probably the latter but because
+ * endpoint at domain looks just like user at 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 @@
*
* \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 @@
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 @@
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 @@
} else {
pj_strdup2(tdata->pool, &uri->user, from);
}
+
+ return 0;
}
+
+ return -1;
}
/*!
@@ -585,7 +1122,7 @@
struct msg_data {
struct ast_msg *msg;
- char *to;
+ char *destination;
char *from;
};
@@ -594,12 +1131,12 @@
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 @@
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 @@
}
}
+/*!
+ * \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 @@
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 @@
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 @@
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;
}
--
To view, visit https://gerrit.asterisk.org/c/asterisk/+/15829
To unsubscribe, or for help writing mail filters, visit https://gerrit.asterisk.org/settings
Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-Change-Id: I67a485196d9199916468f7f98bfb9a0b993a4cce
Gerrit-Change-Number: 15829
Gerrit-PatchSet: 5
Gerrit-Owner: George Joseph <gjoseph at digium.com>
Gerrit-Reviewer: Friendly Automation
Gerrit-Reviewer: George Joseph <gjoseph at digium.com>
Gerrit-Reviewer: Joshua Colp <jcolp at sangoma.com>
Gerrit-Reviewer: Kevin Harwell <kharwell at digium.com>
Gerrit-MessageType: merged
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-code-review/attachments/20210608/d5eb7b19/attachment-0001.html>
More information about the asterisk-code-review
mailing list