- added plugin_prefix
This commit is contained in:
Thomas Ries
+7
-2
@@ -34,13 +34,15 @@ pkglib_LTLIBRARIES = plugin_demo.la \
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plugin_logcall.la \
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plugin_defaulttarget.la \
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plugin_fix_bogus_via.la \
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plugin_stun.la
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plugin_stun.la \
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plugin_prefix.la
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DLOPENPLUGINS = -dlopen plugin_demo.la \
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-dlopen plugin_shortdial.la \
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-dlopen plugin_logcall.la \
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-dlopen plugin_defaulttarget.la \
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-dlopen plugin_fix_bogus_via.la \
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-dlopen plugin_stun.la
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-dlopen plugin_stun.la \
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-dlopen plugin_prefix.la
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#
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plugin_demo_la_SOURCES = plugin_demo.c
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plugin_demo_la_LDFLAGS = -module -avoid-version -shrext '.so'
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@@ -59,6 +61,9 @@ plugin_fix_bogus_via_la_LDFLAGS = -module -avoid-version -shrext '.so'
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#
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plugin_stun_la_SOURCES = plugin_stun.c
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plugin_stun_la_LDFLAGS = -module -avoid-version -shrext '.so'
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#
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plugin_prefix_la_SOURCES = plugin_prefix.c
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plugin_prefix_la_LDFLAGS = -module -avoid-version -shrext '.so'
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#
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@@ -0,0 +1,245 @@
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/*
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Copyright (C) 2002-2011 Thomas Ries <tries@gmx.net>
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This file is part of Siproxd.
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Siproxd is free software; you can redistribute it and/or modify
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it under the terms of the GNU General Public License as published by
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the Free Software Foundation; either version 2 of the License, or
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(at your option) any later version.
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Siproxd is distributed in the hope that it will be useful,
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but WITHOUT ANY WARRANTY; without even the implied warranty of
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MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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GNU General Public License for more details.
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You should have received a copy of the GNU General Public License
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along with Siproxd; if not, write to the Free Software
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Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
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*/
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/*
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* plugin_prefix is based on plugin_shortdial
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*
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* This plugin adds a configured prefix to outgoing calls.
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* Typical use is to add a dial-out code ('9' or '0') to
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* cover user experience with POTS dialing.
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*/
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/* must be defined before including <plugin.h> */
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#define PLUGIN_NAME plugin_prefix
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#include "config.h"
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#include <stdio.h>
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#include <stdlib.h>
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#include <string.h>
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#include <sys/types.h>
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#include <netinet/in.h>
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#include <arpa/inet.h>
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#include <osipparser2/osip_parser.h>
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#include "siproxd.h"
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#include "plugins.h"
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#include "log.h"
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static char const ident[]="$Id$";
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/* Plug-in identification */
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static char name[]="plugin_prefix";
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static char desc[]="Adda a dial-prefix as defined in config file";
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/* global configuration storage - required for config file location */
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extern struct siproxd_config configuration;
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/* plugin configuration storage */
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static struct plugin_config {
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char *prefix_akey;
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} plugin_cfg;
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/* Instructions for config parser */
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static cfgopts_t plugin_cfg_opts[] = {
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{ "plugin_prefix_akey", TYP_STRING, &plugin_cfg.prefix_akey, {0, NULL} },
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{0, 0, 0}
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};
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/* local prototypes */
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static int plugin_prefix_redirect(sip_ticket_t *ticket);
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static int plugin_prefix(sip_ticket_t *ticket);
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/*
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* Plugin API functions code
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*/
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/* Initialization */
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int PLUGIN_INIT(plugin_def_t *plugin_def) {
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plugin_def->api_version=SIPROXD_API_VERSION;
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plugin_def->name=name;
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plugin_def->desc=desc;
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plugin_def->exe_mask=PLUGIN_DETERMINE_TARGET;
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/* read the config file */
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if (read_config(configuration.configfile,
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configuration.config_search,
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plugin_cfg_opts, name) == STS_FAILURE) {
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ERROR("Plugin '%s': could not load config file", name);
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return STS_FAILURE;
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}
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return STS_SUCCESS;
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}
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/* Processing */
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int PLUGIN_PROCESS(int stage, sip_ticket_t *ticket){
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int sts;
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sts=plugin_prefix(ticket);
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return sts;
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}
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/* De-Initialization */
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int PLUGIN_END(plugin_def_t *plugin_def){
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return STS_SUCCESS;
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}
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/*
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* Workload code
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*/
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/* returns STS_SIP_SENT if processing is to be terminated,
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* otherwise STS_SUCCESS (go on with processing) */
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/* code (entry point) */
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static int plugin_prefix(sip_ticket_t *ticket) {
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int sts=STS_SUCCESS;
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osip_uri_t *req_url;
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osip_uri_t *to_url;
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osip_generic_param_t *r=NULL;
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/* plugin loaded and not configured, return with success */
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if (plugin_cfg.prefix_akey == NULL) return STS_SUCCESS;
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DEBUGC(DBCLASS_PLUGIN,"plugin entered");
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req_url=osip_message_get_uri(ticket->sipmsg);
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to_url=osip_to_get_url(ticket->sipmsg);
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/* only outgoing direction is handled */
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sip_find_direction(ticket, NULL);
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if (ticket->direction != DIR_OUTGOING)
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return STS_SUCCESS;
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/* only INVITE and ACK are handled */
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if (!MSG_IS_INVITE(ticket->sipmsg) && !MSG_IS_ACK(ticket->sipmsg))
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return STS_SUCCESS;
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/* REQ URI with username must exist, prefix string must exist */
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if (!req_url || !req_url->username || !plugin_cfg.prefix_akey)
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return STS_SUCCESS; /* ignore */
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/* Loop avoidance:
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* If this INVITE has already been redirected by a prior 302
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* moved response a "redirected" parameter should be present in the
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* URI.
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* Hopefully all UAs (Clients) do honor RFC3261 and copy the
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* *full* URI form the contact header into the new request header
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* upon a 3xx response.
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*/
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if (req_url) {
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osip_uri_param_get_byname(&(req_url->url_params), "redirected", &r);
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if (r && r->gvalue && strcmp(r->gvalue,"true")== 0) {
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DEBUGC(DBCLASS_PLUGIN,"Packet has already been redirected (ReqURI)");
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return STS_SUCCESS;
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}
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}
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if (to_url) {
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osip_uri_param_get_byname(&(to_url->url_params), "redirected", &r);
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if (r && r->gvalue && strcmp(r->gvalue,"true")== 0) {
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DEBUGC(DBCLASS_PLUGIN,"Packet has already been redirected (ToURI)");
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return STS_SUCCESS;
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}
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}
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/*
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* The called number is to be prefixed
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*/
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/* outgoing INVITE request */
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if (MSG_IS_INVITE(ticket->sipmsg)) {
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DEBUGC(DBCLASS_PLUGIN,"processing INVITE");
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sts=plugin_prefix_redirect(ticket);
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}
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/* outgoing ACK request: is result of a local 3xx answer (moved...) */
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else if (MSG_IS_ACK(ticket->sipmsg)) {
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/* make sure we only catch ACKs caused by myself (**02 -> *02 legitime) */
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DEBUGC(DBCLASS_PLUGIN,"processing ACK");
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sts=STS_SIP_SENT; /* eat up the ACK that was directed to myself */
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}
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return sts;
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}
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/* private plugin code */
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static int plugin_prefix_redirect(sip_ticket_t *ticket) {
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osip_uri_t *to_url=ticket->sipmsg->to->url;
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char *to_user=to_url->username;
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char *new_to_user=NULL;
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int i;
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size_t username_len;
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osip_contact_t *contact = NULL;
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/* including \0 + leading character(s) */
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username_len=strlen(to_user) + strlen(plugin_cfg.prefix_akey) + 1;
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new_to_user = osip_malloc(username_len); /* *_len excluding \0 */
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if (!new_to_user) return STS_SUCCESS;
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/* use a "302 Moved temporarily" response back to the client */
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/* new target is within the Contact Header */
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/* remove all Contact headers in message */
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for (i=0; (contact != NULL) || (i == 0); i++) {
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osip_message_get_contact(ticket->sipmsg, 0, &contact);
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if (contact) {
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osip_list_remove(&(ticket->sipmsg->contacts),0);
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osip_contact_free(contact);
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}
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} /* for i */
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/* insert one new Contact header containing the new target address */
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osip_contact_init(&contact);
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osip_uri_clone(to_url, &contact->url);
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/*
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* Add the 'redirected=true' parameter to URI. Required to figure out
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* if this INVITE has already been processed (redirected) and
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* does not need further attention by this plugin.
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* THIS IS REQUIRED TO AVOID A LOOP
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*/
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osip_uri_param_add(&(contact->url->url_params), osip_strdup("redirected"),
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osip_strdup("true"));
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/* only copy the part that really belongs to the username */
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snprintf(new_to_user, username_len, "%s%s",
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plugin_cfg.prefix_akey,to_user );
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/* strncpy may not terminate - do it manually to be sure */
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new_to_user[username_len-1]='\0';
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osip_list_add(&(ticket->sipmsg->contacts),contact,0);
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INFO("redirecting %s -> %s", to_user, new_to_user);
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/* USER part is always present */
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osip_free(contact->url->username);
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contact->url->username=new_to_user;
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/* sent redirect message back to local client */
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sip_gen_response(ticket, 302 /*Moved temporarily*/);
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return STS_SIP_SENT;
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}
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+1
-1
@@ -208,7 +208,7 @@ int call_plugins(int stage, sip_ticket_t *ticket) {
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if (sts == STS_SIP_SENT) {
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DEBUGC(DBCLASS_PLUGIN, "call_plugins: PLUGIN_DETERMINE_TARGET "
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"prematurely ending plugin processing in module "
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"%s sts=STS_SIP_SENT", cur->desc);
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"%s sts=STS_SIP_SENT", cur->name);
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return sts;
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}
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break;
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