dect
/
asterisk
Archived
13
0
Fork 0
This repository has been archived on 2022-02-17. You can view files and clone it, but cannot push or open issues or pull requests.
asterisk/channels/chan_gtalk.c

2145 lines
65 KiB
C
Raw Normal View History

/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 1999 - 2005, Digium, Inc.
*
* Matt O'Gorman <mogorman@digium.com>
*
* See http://www.asterisk.org for more information about
* the Asterisk project. Please do not directly contact
* any of the maintainers of this project for assistance;
* the project provides a web site, mailing lists and IRC
* channels for your use.
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
/*! \file
*
* \author Matt O'Gorman <mogorman@digium.com>
*
* \brief Gtalk Channel Driver, until google/libjingle works with jingle spec
*
* \ingroup channel_drivers
*/
/*** MODULEINFO
<depend>iksemel</depend>
<depend>res_jabber</depend>
<use>openssl</use>
***/
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include <sys/socket.h>
#include <fcntl.h>
#include <netdb.h>
#include <netinet/in.h>
#include <arpa/inet.h>
#include <sys/signal.h>
#include <iksemel.h>
#include <pthread.h>
#include <ctype.h>
#include "asterisk/lock.h"
#include "asterisk/channel.h"
#include "asterisk/config.h"
#include "asterisk/module.h"
#include "asterisk/pbx.h"
#include "asterisk/sched.h"
#include "asterisk/io.h"
#include "asterisk/rtp_engine.h"
#include "asterisk/stun.h"
#include "asterisk/acl.h"
#include "asterisk/callerid.h"
#include "asterisk/file.h"
#include "asterisk/cli.h"
#include "asterisk/app.h"
#include "asterisk/musiconhold.h"
#include "asterisk/manager.h"
#include "asterisk/stringfields.h"
#include "asterisk/utils.h"
#include "asterisk/causes.h"
#include "asterisk/astobj.h"
#include "asterisk/abstract_jb.h"
#include "asterisk/jabber.h"
#define GOOGLE_CONFIG "gtalk.conf"
#define GOOGLE_NS "http://www.google.com/session"
/*! Global jitterbuffer configuration - by default, jb is disabled */
static struct ast_jb_conf default_jbconf =
{
.flags = 0,
.max_size = -1,
.resync_threshold = -1,
.impl = "",
.target_extra = -1,
};
static struct ast_jb_conf global_jbconf;
enum gtalk_protocol {
AJI_PROTOCOL_UDP = 1,
AJI_PROTOCOL_SSLTCP = 2,
};
enum gtalk_connect_type {
AJI_CONNECT_STUN = 1,
AJI_CONNECT_LOCAL = 2,
AJI_CONNECT_RELAY = 3,
};
struct gtalk_pvt {
ast_mutex_t lock; /*!< Channel private lock */
time_t laststun;
struct gtalk *parent; /*!< Parent client */
char sid[100];
char us[AJI_MAX_JIDLEN];
char them[AJI_MAX_JIDLEN];
char ring[10]; /*!< Message ID of ring */
iksrule *ringrule; /*!< Rule for matching RING request */
int initiator; /*!< If we're the initiator */
int alreadygone;
int capability;
struct ast_codec_pref prefs;
struct gtalk_candidate *theircandidates;
struct gtalk_candidate *ourcandidates;
char cid_num[80]; /*!< Caller ID num */
char cid_name[80]; /*!< Caller ID name */
char exten[80]; /*!< Called extension */
struct ast_channel *owner; /*!< Master Channel */
struct ast_rtp_instance *rtp; /*!< RTP audio session */
struct ast_rtp_instance *vrtp; /*!< RTP video session */
format_t jointcapability; /*!< Supported capability at both ends (codecs ) */
format_t peercapability;
struct gtalk_pvt *next; /* Next entity */
};
struct gtalk_candidate {
char name[100];
enum gtalk_protocol protocol;
double preference;
char username[100];
char password[100];
enum gtalk_connect_type type;
char network[6];
int generation;
char ip[16];
int port;
int receipt;
struct gtalk_candidate *next;
};
struct gtalk {
ASTOBJ_COMPONENTS(struct gtalk);
struct aji_client *connection;
struct aji_buddy *buddy;
struct gtalk_pvt *p;
struct ast_codec_pref prefs;
int amaflags; /*!< AMA Flags */
char user[AJI_MAX_JIDLEN];
char context[AST_MAX_CONTEXT];
char parkinglot[AST_MAX_CONTEXT]; /*!< Parkinglot */
char accountcode[AST_MAX_ACCOUNT_CODE]; /*!< Account code */
format_t capability;
ast_group_t callgroup; /*!< Call group */
ast_group_t pickupgroup; /*!< Pickup group */
int callingpres; /*!< Calling presentation */
int allowguest;
char language[MAX_LANGUAGE]; /*!< Default language for prompts */
char musicclass[MAX_MUSICCLASS]; /*!< Music on Hold class */
};
struct gtalk_container {
ASTOBJ_CONTAINER_COMPONENTS(struct gtalk);
};
static const char desc[] = "Gtalk Channel";
static format_t global_capability = AST_FORMAT_ULAW | AST_FORMAT_ALAW | AST_FORMAT_GSM | AST_FORMAT_H263;
AST_MUTEX_DEFINE_STATIC(gtalklock); /*!< Protect the interface list (of gtalk_pvt's) */
/* Forward declarations */
static struct ast_channel *gtalk_request(const char *type, format_t format, const struct ast_channel *requestor, void *data, int *cause);
2007-01-19 18:06:03 +00:00
static int gtalk_digit(struct ast_channel *ast, char digit, unsigned int duration);
static int gtalk_sendtext(struct ast_channel *ast, const char *text);
2007-01-19 18:06:03 +00:00
static int gtalk_digit_begin(struct ast_channel *ast, char digit);
static int gtalk_digit_end(struct ast_channel *ast, char digit, unsigned int duration);
static int gtalk_call(struct ast_channel *ast, char *dest, int timeout);
static int gtalk_hangup(struct ast_channel *ast);
static int gtalk_answer(struct ast_channel *ast);
static int gtalk_action(struct gtalk *client, struct gtalk_pvt *p, const char *action);
static void gtalk_free_pvt(struct gtalk *client, struct gtalk_pvt *p);
static int gtalk_newcall(struct gtalk *client, ikspak *pak);
static struct ast_frame *gtalk_read(struct ast_channel *ast);
static int gtalk_write(struct ast_channel *ast, struct ast_frame *f);
static int gtalk_indicate(struct ast_channel *ast, int condition, const void *data, size_t datalen);
static int gtalk_fixup(struct ast_channel *oldchan, struct ast_channel *newchan);
static int gtalk_sendhtml(struct ast_channel *ast, int subclass, const char *data, int datalen);
static struct gtalk_pvt *gtalk_alloc(struct gtalk *client, const char *us, const char *them, const char *sid);
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85460 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-11 19:03:06 +00:00
static char *gtalk_do_reload(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static char *gtalk_show_channels(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
/*! \brief PBX interface structure for channel registration */
static const struct ast_channel_tech gtalk_tech = {
.type = "Gtalk",
.description = "Gtalk Channel Driver",
.capabilities = AST_FORMAT_AUDIO_MASK,
.requester = gtalk_request,
.send_text = gtalk_sendtext,
2007-01-19 18:06:03 +00:00
.send_digit_begin = gtalk_digit_begin,
.send_digit_end = gtalk_digit_end,
.bridge = ast_rtp_instance_bridge,
.call = gtalk_call,
.hangup = gtalk_hangup,
.answer = gtalk_answer,
.read = gtalk_read,
.write = gtalk_write,
.exception = gtalk_read,
.indicate = gtalk_indicate,
.fixup = gtalk_fixup,
.send_html = gtalk_sendhtml,
.properties = AST_CHAN_TP_WANTSJITTER | AST_CHAN_TP_CREATESJITTER
};
static struct sockaddr_in bindaddr = { 0, }; /*!< The address we bind to */
static struct sched_context *sched; /*!< The scheduling context */
static struct io_context *io; /*!< The IO context */
static struct in_addr __ourip;
static struct ast_cli_entry gtalk_cli[] = {
AST_CLI_DEFINE(gtalk_do_reload, "Reload GoogleTalk configuration"),
AST_CLI_DEFINE(gtalk_show_channels, "Show GoogleTalk channels"),
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85460 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-11 19:03:06 +00:00
};
static char externip[16];
static struct gtalk_container gtalk_list;
static void gtalk_member_destroy(struct gtalk *obj)
{
ast_free(obj);
}
static struct gtalk *find_gtalk(char *name, char *connection)
{
struct gtalk *gtalk = NULL;
char *domain = NULL , *s = NULL;
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85460 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-11 19:03:06 +00:00
if (strchr(connection, '@')) {
s = ast_strdupa(connection);
domain = strsep(&s, "@");
ast_verbose("OOOOH domain = %s\n", domain);
}
gtalk = ASTOBJ_CONTAINER_FIND(&gtalk_list, name);
if (!gtalk && strchr(name, '@'))
gtalk = ASTOBJ_CONTAINER_FIND_FULL(&gtalk_list, name, user,,, strcasecmp);
if (!gtalk) {
/* guest call */
ASTOBJ_CONTAINER_TRAVERSE(&gtalk_list, 1, {
ASTOBJ_RDLOCK(iterator);
if (!strcasecmp(iterator->name, "guest")) {
gtalk = iterator;
}
ASTOBJ_UNLOCK(iterator);
if (gtalk)
break;
});
}
return gtalk;
}
static int add_codec_to_answer(const struct gtalk_pvt *p, int codec, iks *dcodecs)
{
int res = 0;
char *format = ast_getformatname(codec);
if (!strcasecmp("ulaw", format)) {
iks *payload_eg711u, *payload_pcmu;
payload_pcmu = iks_new("payload-type");
payload_eg711u = iks_new("payload-type");
if(!payload_eg711u || !payload_pcmu) {
iks_delete(payload_pcmu);
iks_delete(payload_eg711u);
ast_log(LOG_WARNING,"Failed to allocate iks node");
return -1;
}
iks_insert_attrib(payload_pcmu, "id", "0");
iks_insert_attrib(payload_pcmu, "name", "PCMU");
iks_insert_attrib(payload_pcmu, "clockrate","8000");
iks_insert_attrib(payload_pcmu, "bitrate","64000");
iks_insert_attrib(payload_eg711u, "id", "100");
iks_insert_attrib(payload_eg711u, "name", "EG711U");
iks_insert_attrib(payload_eg711u, "clockrate","8000");
iks_insert_attrib(payload_eg711u, "bitrate","64000");
iks_insert_node(dcodecs, payload_pcmu);
iks_insert_node(dcodecs, payload_eg711u);
res ++;
}
if (!strcasecmp("alaw", format)) {
iks *payload_eg711a, *payload_pcma;
payload_pcma = iks_new("payload-type");
payload_eg711a = iks_new("payload-type");
if(!payload_eg711a || !payload_pcma) {
iks_delete(payload_eg711a);
iks_delete(payload_pcma);
ast_log(LOG_WARNING,"Failed to allocate iks node");
return -1;
}
iks_insert_attrib(payload_pcma, "id", "8");
iks_insert_attrib(payload_pcma, "name", "PCMA");
iks_insert_attrib(payload_pcma, "clockrate","8000");
iks_insert_attrib(payload_pcma, "bitrate","64000");
payload_eg711a = iks_new("payload-type");
iks_insert_attrib(payload_eg711a, "id", "101");
iks_insert_attrib(payload_eg711a, "name", "EG711A");
iks_insert_attrib(payload_eg711a, "clockrate","8000");
iks_insert_attrib(payload_eg711a, "bitrate","64000");
iks_insert_node(dcodecs, payload_pcma);
iks_insert_node(dcodecs, payload_eg711a);
res ++;
}
if (!strcasecmp("ilbc", format)) {
iks *payload_ilbc = iks_new("payload-type");
if(!payload_ilbc) {
ast_log(LOG_WARNING,"Failed to allocate iks node");
return -1;
}
iks_insert_attrib(payload_ilbc, "id", "97");
iks_insert_attrib(payload_ilbc, "name", "iLBC");
iks_insert_attrib(payload_ilbc, "clockrate","8000");
iks_insert_attrib(payload_ilbc, "bitrate","13300");
iks_insert_node(dcodecs, payload_ilbc);
res ++;
}
if (!strcasecmp("g723", format)) {
iks *payload_g723 = iks_new("payload-type");
if(!payload_g723) {
ast_log(LOG_WARNING,"Failed to allocate iks node");
return -1;
}
iks_insert_attrib(payload_g723, "id", "4");
iks_insert_attrib(payload_g723, "name", "G723");
iks_insert_attrib(payload_g723, "clockrate","8000");
iks_insert_attrib(payload_g723, "bitrate","6300");
iks_insert_node(dcodecs, payload_g723);
res ++;
}
if (!strcasecmp("speex", format)) {
iks *payload_speex = iks_new("payload-type");
if(!payload_speex) {
ast_log(LOG_WARNING,"Failed to allocate iks node");
return -1;
}
iks_insert_attrib(payload_speex, "id", "110");
iks_insert_attrib(payload_speex, "name", "speex");
iks_insert_attrib(payload_speex, "clockrate","8000");
iks_insert_attrib(payload_speex, "bitrate","11000");
iks_insert_node(dcodecs, payload_speex);
res++;
}
if (!strcasecmp("gsm", format)) {
iks *payload_gsm = iks_new("payload-type");
if(!payload_gsm) {
ast_log(LOG_WARNING,"Failed to allocate iks node");
return -1;
}
iks_insert_attrib(payload_gsm, "id", "103");
iks_insert_attrib(payload_gsm, "name", "gsm");
iks_insert_node(dcodecs, payload_gsm);
res++;
}
return res;
}
static int gtalk_invite(struct gtalk_pvt *p, char *to, char *from, char *sid, int initiator)
{
struct gtalk *client = p->parent;
iks *iq, *gtalk, *dcodecs, *payload_telephone, *transport;
int x;
int pref_codec = 0;
int alreadysent = 0;
int codecs_num = 0;
char *lowerto = NULL;
iq = iks_new("iq");
gtalk = iks_new("session");
dcodecs = iks_new("description");
transport = iks_new("transport");
payload_telephone = iks_new("payload-type");
if (!(iq && gtalk && dcodecs && transport && payload_telephone)){
iks_delete(iq);
iks_delete(gtalk);
iks_delete(dcodecs);
iks_delete(transport);
iks_delete(payload_telephone);
ast_log(LOG_ERROR, "Could not allocate iksemel nodes\n");
return 0;
}
iks_insert_attrib(dcodecs, "xmlns", "http://www.google.com/session/phone");
iks_insert_attrib(dcodecs, "xml:lang", "en");
for (x = 0; x < 64; x++) {
if (!(pref_codec = ast_codec_pref_index(&client->prefs, x)))
break;
if (!(client->capability & pref_codec))
continue;
if (alreadysent & pref_codec)
continue;
codecs_num = add_codec_to_answer(p, pref_codec, dcodecs);
alreadysent |= pref_codec;
}
if (codecs_num) {
/* only propose DTMF within an audio session */
iks_insert_attrib(payload_telephone, "id", "106");
iks_insert_attrib(payload_telephone, "name", "telephone-event");
iks_insert_attrib(payload_telephone, "clockrate", "8000");
}
iks_insert_attrib(transport,"xmlns","http://www.google.com/transport/p2p");
iks_insert_attrib(iq, "type", "set");
iks_insert_attrib(iq, "to", to);
iks_insert_attrib(iq, "from", from);
iks_insert_attrib(iq, "id", client->connection->mid);
ast_aji_increment_mid(client->connection->mid);
iks_insert_attrib(gtalk, "xmlns", "http://www.google.com/session");
iks_insert_attrib(gtalk, "type",initiator ? "initiate": "accept");
/* put the initiator attribute to lower case if we receive the call
* otherwise GoogleTalk won't establish the session */
if (!initiator) {
char c;
char *t = lowerto = ast_strdupa(to);
while (((c = *t) != '/') && (*t++ = tolower(c)));
}
iks_insert_attrib(gtalk, "initiator", initiator ? from : lowerto);
iks_insert_attrib(gtalk, "id", sid);
iks_insert_node(iq, gtalk);
iks_insert_node(gtalk, dcodecs);
iks_insert_node(gtalk, transport);
iks_insert_node(dcodecs, payload_telephone);
ast_aji_send(client->connection, iq);
iks_delete(payload_telephone);
iks_delete(transport);
iks_delete(dcodecs);
iks_delete(gtalk);
iks_delete(iq);
return 1;
}
static int gtalk_invite_response(struct gtalk_pvt *p, char *to , char *from, char *sid, int initiator)
{
iks *iq, *session, *transport;
char *lowerto = NULL;
iq = iks_new("iq");
session = iks_new("session");
transport = iks_new("transport");
if(!(iq && session && transport)) {
iks_delete(iq);
iks_delete(session);
iks_delete(transport);
ast_log(LOG_ERROR, " Unable to allocate IKS node\n");
return -1;
}
iks_insert_attrib(iq, "from", from);
iks_insert_attrib(iq, "to", to);
iks_insert_attrib(iq, "type", "set");
iks_insert_attrib(iq, "id",p->parent->connection->mid);
ast_aji_increment_mid(p->parent->connection->mid);
iks_insert_attrib(session, "type", "transport-accept");
iks_insert_attrib(session, "id", sid);
/* put the initiator attribute to lower case if we receive the call
* otherwise GoogleTalk won't establish the session */
if (!initiator) {
char c;
char *t = lowerto = ast_strdupa(to);
while (((c = *t) != '/') && (*t++ = tolower(c)));
}
iks_insert_attrib(session, "initiator", initiator ? from : lowerto);
iks_insert_attrib(session, "xmlns", "http://www.google.com/session");
iks_insert_attrib(transport, "xmlns", "http://www.google.com/transport/p2p");
iks_insert_node(iq,session);
iks_insert_node(session,transport);
ast_aji_send(p->parent->connection, iq);
iks_delete(transport);
iks_delete(session);
iks_delete(iq);
return 1;
}
static int gtalk_ringing_ack(void *data, ikspak *pak)
{
struct gtalk_pvt *p = data;
if (p->ringrule)
iks_filter_remove_rule(p->parent->connection->f, p->ringrule);
p->ringrule = NULL;
if (p->owner)
ast_queue_control(p->owner, AST_CONTROL_RINGING);
return IKS_FILTER_EAT;
}
static int gtalk_answer(struct ast_channel *ast)
{
struct gtalk_pvt *p = ast->tech_pvt;
int res = 0;
ast_debug(1, "Answer!\n");
ast_mutex_lock(&p->lock);
gtalk_invite(p, p->them, p->us,p->sid, 0);
manager_event(EVENT_FLAG_SYSTEM, "ChannelUpdate", "Channel: %s\r\nChanneltype: %s\r\nGtalk-SID: %s\r\n",
ast->name, "GTALK", p->sid);
ast_mutex_unlock(&p->lock);
return res;
}
static enum ast_rtp_glue_result gtalk_get_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
{
struct gtalk_pvt *p = chan->tech_pvt;
enum ast_rtp_glue_result res = AST_RTP_GLUE_RESULT_FORBID;
if (!p)
return res;
ast_mutex_lock(&p->lock);
if (p->rtp){
ao2_ref(p->rtp, +1);
*instance = p->rtp;
res = AST_RTP_GLUE_RESULT_LOCAL;
}
ast_mutex_unlock(&p->lock);
return res;
}
static format_t gtalk_get_codec(struct ast_channel *chan)
{
struct gtalk_pvt *p = chan->tech_pvt;
return p->peercapability;
}
static int gtalk_set_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance *rtp, struct ast_rtp_instance *vrtp, struct ast_rtp_instance *trtp, format_t codecs, int nat_active)
{
struct gtalk_pvt *p;
p = chan->tech_pvt;
if (!p)
return -1;
ast_mutex_lock(&p->lock);
/* if (rtp)
ast_rtp_get_peer(rtp, &p->redirip);
else
memset(&p->redirip, 0, sizeof(p->redirip));
p->redircodecs = codecs; */
/* Reset lastrtprx timer */
ast_mutex_unlock(&p->lock);
return 0;
}
static struct ast_rtp_glue gtalk_rtp_glue = {
.type = "Gtalk",
.get_rtp_info = gtalk_get_rtp_peer,
.get_codec = gtalk_get_codec,
.update_peer = gtalk_set_rtp_peer,
};
static int gtalk_response(struct gtalk *client, char *from, ikspak *pak, const char *reasonstr, const char *reasonstr2)
{
iks *response = NULL, *error = NULL, *reason = NULL;
int res = -1;
response = iks_new("iq");
if (response) {
iks_insert_attrib(response, "type", "result");
iks_insert_attrib(response, "from", from);
iks_insert_attrib(response, "to", iks_find_attrib(pak->x, "from"));
iks_insert_attrib(response, "id", iks_find_attrib(pak->x, "id"));
if (reasonstr) {
error = iks_new("error");
if (error) {
iks_insert_attrib(error, "type", "cancel");
reason = iks_new(reasonstr);
if (reason)
iks_insert_node(error, reason);
iks_insert_node(response, error);
}
}
ast_aji_send(client->connection, response);
res = 0;
}
iks_delete(reason);
iks_delete(error);
iks_delete(response);
return res;
}
static int gtalk_is_answered(struct gtalk *client, ikspak *pak)
{
struct gtalk_pvt *tmp;
char *from;
iks *codec;
char s1[BUFSIZ], s2[BUFSIZ], s3[BUFSIZ];
int peernoncodeccapability;
ast_log(LOG_DEBUG, "The client is %s\n", client->name);
/* Make sure our new call doesn't exist yet */
for (tmp = client->p; tmp; tmp = tmp->next) {
if (iks_find_with_attrib(pak->x, "session", "id", tmp->sid))
break;
}
/* codec points to the first <payload-type/> tag */
Merged revisions 185362 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185362 | dbrooks | 2009-03-31 11:37:12 -0500 (Tue, 31 Mar 2009) | 35 lines Fix incorrect parsing in chan_gtalk when xmpp contains extra whitespaces To drill into the xmpp to find the capabilities between channels, chan_gtalk calls iks_child() and iks_next(). iks_child() and iks_next() are functions in the iksemel xml parsing library that traverse xml nodes. The bug here is that both iks_child() and iks_next() will return the next iks_struct node *regardless* of type. chan_gtalk expects the next node to be of type IKS_TAG, which in most cases, it is, but in this case (a call being made from the Empathy IM client), there exists iks_struct nodes which are not IKS_TAG data (they are extraneous whitespaces), and chan_gtalk doesn't handle that case, so capabilities don't match, and a call cannot be made. iks_first_tag() and iks_next_tag(), on the other hand, will not return the very next iks_struct, but will check to see if the next iks_struct is of type IKS_TAG. If it isn't, it will be skipped, and the next struct of type IKS_TAG it finds will be returned. This assures that chan_gtalk will find the iks_struct it is looking for. This fix simply changes all calls to iks_child() and iks_next() to become calls to iks_first_tag() and iks_next_tag(), which resolves the capability matching. The following is a payload listing from Empathy, which, due to the extraneous whitespace, will not be parsed correctly by iksemel: <iq from='dbrooksjab@235-22-24-10/Telepathy' to='astjab@235-22-24-10/asterisk' type='set' id='542757715704'> <session xmlns='http://www.google.com/session' initiator='dbrooksjab@235-22-24-10/Telepathy' type='initiate' id='1837267342'> <description xmlns='http://www.google.com/session/phone'> <payload-type clockrate='16000' name='speex' id='96'/> <payload-type clockrate='8000' name='PCMA' id='8'/> <payload-type clockrate='8000' name='PCMU' id='0'/> <payload-type clockrate='90000' name='MPA' id='97'/> <payload-type clockrate='16000' name='SIREN' id='98'/> <payload-type clockrate='8000' name='telephone-event' id='99'/> </description> </session> </iq> Review: http://reviewboard.digium.com/r/181/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@185363 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-31 16:46:57 +00:00
codec = iks_first_tag(iks_first_tag(iks_first_tag(pak->x)));
while (codec) {
ast_rtp_codecs_payloads_set_m_type(ast_rtp_instance_get_codecs(tmp->rtp), tmp->rtp, atoi(iks_find_attrib(codec, "id")));
ast_rtp_codecs_payloads_set_rtpmap_type(ast_rtp_instance_get_codecs(tmp->rtp), tmp->rtp, atoi(iks_find_attrib(codec, "id")), "audio", iks_find_attrib(codec, "name"), 0);
Merged revisions 185362 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185362 | dbrooks | 2009-03-31 11:37:12 -0500 (Tue, 31 Mar 2009) | 35 lines Fix incorrect parsing in chan_gtalk when xmpp contains extra whitespaces To drill into the xmpp to find the capabilities between channels, chan_gtalk calls iks_child() and iks_next(). iks_child() and iks_next() are functions in the iksemel xml parsing library that traverse xml nodes. The bug here is that both iks_child() and iks_next() will return the next iks_struct node *regardless* of type. chan_gtalk expects the next node to be of type IKS_TAG, which in most cases, it is, but in this case (a call being made from the Empathy IM client), there exists iks_struct nodes which are not IKS_TAG data (they are extraneous whitespaces), and chan_gtalk doesn't handle that case, so capabilities don't match, and a call cannot be made. iks_first_tag() and iks_next_tag(), on the other hand, will not return the very next iks_struct, but will check to see if the next iks_struct is of type IKS_TAG. If it isn't, it will be skipped, and the next struct of type IKS_TAG it finds will be returned. This assures that chan_gtalk will find the iks_struct it is looking for. This fix simply changes all calls to iks_child() and iks_next() to become calls to iks_first_tag() and iks_next_tag(), which resolves the capability matching. The following is a payload listing from Empathy, which, due to the extraneous whitespace, will not be parsed correctly by iksemel: <iq from='dbrooksjab@235-22-24-10/Telepathy' to='astjab@235-22-24-10/asterisk' type='set' id='542757715704'> <session xmlns='http://www.google.com/session' initiator='dbrooksjab@235-22-24-10/Telepathy' type='initiate' id='1837267342'> <description xmlns='http://www.google.com/session/phone'> <payload-type clockrate='16000' name='speex' id='96'/> <payload-type clockrate='8000' name='PCMA' id='8'/> <payload-type clockrate='8000' name='PCMU' id='0'/> <payload-type clockrate='90000' name='MPA' id='97'/> <payload-type clockrate='16000' name='SIREN' id='98'/> <payload-type clockrate='8000' name='telephone-event' id='99'/> </description> </session> </iq> Review: http://reviewboard.digium.com/r/181/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@185363 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-31 16:46:57 +00:00
codec = iks_next_tag(codec);
}
/* Now gather all of the codecs that we are asked for */
ast_rtp_codecs_payload_formats(ast_rtp_instance_get_codecs(tmp->rtp), &tmp->peercapability, &peernoncodeccapability);
/* at this point, we received an awser from the remote Gtalk client,
which allows us to compare capabilities */
tmp->jointcapability = tmp->capability & tmp->peercapability;
if (!tmp->jointcapability) {
ast_log(LOG_WARNING, "Capabilities don't match : us - %s, peer - %s, combined - %s \n", ast_getformatname_multiple(s1, BUFSIZ, tmp->capability),
ast_getformatname_multiple(s2, BUFSIZ, tmp->peercapability),
ast_getformatname_multiple(s3, BUFSIZ, tmp->jointcapability));
/* close session if capabilities don't match */
ast_queue_hangup(tmp->owner);
return -1;
}
from = iks_find_attrib(pak->x, "to");
if(!from)
from = client->connection->jid->full;
if (tmp) {
if (tmp->owner)
ast_queue_control(tmp->owner, AST_CONTROL_ANSWER);
} else
ast_log(LOG_NOTICE, "Whoa, didn't find call!\n");
gtalk_response(client, from, pak, NULL, NULL);
return 1;
}
static int gtalk_is_accepted(struct gtalk *client, ikspak *pak)
{
struct gtalk_pvt *tmp;
char *from;
ast_log(LOG_DEBUG, "The client is %s\n", client->name);
/* find corresponding call */
for (tmp = client->p; tmp; tmp = tmp->next) {
if (iks_find_with_attrib(pak->x, "session", "id", tmp->sid))
break;
}
from = iks_find_attrib(pak->x, "to");
if(!from)
from = client->connection->jid->full;
if (!tmp)
ast_log(LOG_NOTICE, "Whoa, didn't find call!\n");
/* answer 'iq' packet to let the remote peer know that we're alive */
gtalk_response(client, from, pak, NULL, NULL);
return 1;
}
static int gtalk_handle_dtmf(struct gtalk *client, ikspak *pak)
{
struct gtalk_pvt *tmp;
iks *dtmfnode = NULL, *dtmfchild = NULL;
char *dtmf;
char *from;
/* Make sure our new call doesn't exist yet */
for (tmp = client->p; tmp; tmp = tmp->next) {
if (iks_find_with_attrib(pak->x, "session", "id", tmp->sid) || iks_find_with_attrib(pak->x, "gtalk", "sid", tmp->sid))
break;
}
from = iks_find_attrib(pak->x, "to");
if(!from)
from = client->connection->jid->full;
if (tmp) {
if(iks_find_with_attrib(pak->x, "dtmf-method", "method", "rtp")) {
gtalk_response(client, from, pak,
"feature-not-implemented xmlns='urn:ietf:params:xml:ns:xmpp-stanzas'",
"unsupported-dtmf-method xmlns='http://jabber.org/protocol/gtalk/info/dtmf#errors'");
return -1;
}
if ((dtmfnode = iks_find(pak->x, "dtmf"))) {
if((dtmf = iks_find_attrib(dtmfnode, "code"))) {
if(iks_find_with_attrib(pak->x, "dtmf", "action", "button-up")) {
struct ast_frame f = {AST_FRAME_DTMF_BEGIN, };
f.subclass.integer = dtmf[0];
ast_queue_frame(tmp->owner, &f);
ast_verbose("GOOGLE! DTMF-relay event received: %c\n", (int) f.subclass.integer);
} else if(iks_find_with_attrib(pak->x, "dtmf", "action", "button-down")) {
struct ast_frame f = {AST_FRAME_DTMF_END, };
f.subclass.integer = dtmf[0];
ast_queue_frame(tmp->owner, &f);
ast_verbose("GOOGLE! DTMF-relay event received: %c\n", (int) f.subclass.integer);
} else if(iks_find_attrib(pak->x, "dtmf")) { /* 250 millasecond default */
struct ast_frame f = {AST_FRAME_DTMF, };
f.subclass.integer = dtmf[0];
ast_queue_frame(tmp->owner, &f);
ast_verbose("GOOGLE! DTMF-relay event received: %c\n", (int) f.subclass.integer);
}
}
} else if ((dtmfnode = iks_find_with_attrib(pak->x, "gtalk", "action", "session-info"))) {
if((dtmfchild = iks_find(dtmfnode, "dtmf"))) {
if((dtmf = iks_find_attrib(dtmfchild, "code"))) {
if(iks_find_with_attrib(dtmfnode, "dtmf", "action", "button-up")) {
struct ast_frame f = {AST_FRAME_DTMF_END, };
f.subclass.integer = dtmf[0];
ast_queue_frame(tmp->owner, &f);
ast_verbose("GOOGLE! DTMF-relay event received: %c\n", (int) f.subclass.integer);
} else if(iks_find_with_attrib(dtmfnode, "dtmf", "action", "button-down")) {
struct ast_frame f = {AST_FRAME_DTMF_BEGIN, };
f.subclass.integer = dtmf[0];
ast_queue_frame(tmp->owner, &f);
ast_verbose("GOOGLE! DTMF-relay event received: %c\n", (int) f.subclass.integer);
}
}
}
}
gtalk_response(client, from, pak, NULL, NULL);
return 1;
} else
ast_log(LOG_NOTICE, "Whoa, didn't find call!\n");
gtalk_response(client, from, pak, NULL, NULL);
return 1;
}
static int gtalk_hangup_farend(struct gtalk *client, ikspak *pak)
{
struct gtalk_pvt *tmp;
char *from;
ast_debug(1, "The client is %s\n", client->name);
/* Make sure our new call doesn't exist yet */
for (tmp = client->p; tmp; tmp = tmp->next) {
if (iks_find_with_attrib(pak->x, "session", "id", tmp->sid))
break;
}
from = iks_find_attrib(pak->x, "to");
if(!from)
from = client->connection->jid->full;
if (tmp) {
tmp->alreadygone = 1;
if (tmp->owner)
ast_queue_hangup(tmp->owner);
} else
ast_log(LOG_NOTICE, "Whoa, didn't find call!\n");
gtalk_response(client, from, pak, NULL, NULL);
return 1;
}
static int gtalk_create_candidates(struct gtalk *client, struct gtalk_pvt *p, char *sid, char *from, char *to)
{
struct gtalk_candidate *tmp;
struct aji_client *c = client->connection;
struct gtalk_candidate *ours1 = NULL, *ours2 = NULL;
struct sockaddr_in sin = { 0, };
struct sockaddr_in dest;
struct in_addr us;
iks *iq, *gtalk, *candidate, *transport;
char user[17], pass[17], preference[5], port[7];
char *lowerfrom = NULL;
iq = iks_new("iq");
gtalk = iks_new("session");
candidate = iks_new("candidate");
transport = iks_new("transport");
if (!iq || !gtalk || !candidate || !transport) {
ast_log(LOG_ERROR, "Memory allocation error\n");
goto safeout;
}
ours1 = ast_calloc(1, sizeof(*ours1));
ours2 = ast_calloc(1, sizeof(*ours2));
if (!ours1 || !ours2)
goto safeout;
iks_insert_attrib(transport, "xmlns","http://www.google.com/transport/p2p");
iks_insert_node(iq, gtalk);
iks_insert_node(gtalk,transport);
iks_insert_node(transport, candidate);
for (; p; p = p->next) {
if (!strcasecmp(p->sid, sid))
break;
}
if (!p) {
ast_log(LOG_NOTICE, "No matching gtalk session - SID %s!\n", sid);
goto safeout;
}
ast_rtp_instance_get_local_address(p->rtp, &sin);
ast_find_ourip(&us, bindaddr);
if (!strcmp(ast_inet_ntoa(us), "127.0.0.1")) {
ast_log(LOG_WARNING, "Found a loopback IP on the system, check your network configuration or set the bindaddr attribute.");
}
/* Setup our gtalk candidates */
ast_copy_string(ours1->name, "rtp", sizeof(ours1->name));
ours1->port = ntohs(sin.sin_port);
ours1->preference = 1;
snprintf(user, sizeof(user), "%08lx%08lx", ast_random(), ast_random());
snprintf(pass, sizeof(pass), "%08lx%08lx", ast_random(), ast_random());
ast_copy_string(ours1->username, user, sizeof(ours1->username));
ast_copy_string(ours1->password, pass, sizeof(ours1->password));
ast_copy_string(ours1->ip, ast_inet_ntoa(us), sizeof(ours1->ip));
ours1->protocol = AJI_PROTOCOL_UDP;
ours1->type = AJI_CONNECT_LOCAL;
ours1->generation = 0;
p->ourcandidates = ours1;
if (!ast_strlen_zero(externip)) {
/* XXX We should really stun for this one not just go with externip XXX */
snprintf(user, sizeof(user), "%08lx%08lx", ast_random(), ast_random());
snprintf(pass, sizeof(pass), "%08lx%08lx", ast_random(), ast_random());
ast_copy_string(ours2->username, user, sizeof(ours2->username));
ast_copy_string(ours2->password, pass, sizeof(ours2->password));
ast_copy_string(ours2->ip, externip, sizeof(ours2->ip));
ast_copy_string(ours2->name, "rtp", sizeof(ours1->name));
ours2->port = ntohs(sin.sin_port);
ours2->preference = 0.9;
ours2->protocol = AJI_PROTOCOL_UDP;
ours2->type = AJI_CONNECT_STUN;
ours2->generation = 0;
ours1->next = ours2;
ours2 = NULL;
}
ours1 = NULL;
dest.sin_addr = __ourip;
dest.sin_port = sin.sin_port;
for (tmp = p->ourcandidates; tmp; tmp = tmp->next) {
snprintf(port, sizeof(port), "%d", tmp->port);
snprintf(preference, sizeof(preference), "%.2f", tmp->preference);
iks_insert_attrib(iq, "from", to);
iks_insert_attrib(iq, "to", from);
iks_insert_attrib(iq, "type", "set");
iks_insert_attrib(iq, "id", c->mid);
ast_aji_increment_mid(c->mid);
iks_insert_attrib(gtalk, "type", "transport-info");
iks_insert_attrib(gtalk, "id", sid);
/* put the initiator attribute to lower case if we receive the call
* otherwise GoogleTalk won't establish the session */
if (!p->initiator) {
char c;
char *t = lowerfrom = ast_strdupa(from);
while (((c = *t) != '/') && (*t++ = tolower(c)));
}
iks_insert_attrib(gtalk, "initiator", (p->initiator) ? to : lowerfrom);
iks_insert_attrib(gtalk, "xmlns", GOOGLE_NS);
iks_insert_attrib(candidate, "name", tmp->name);
iks_insert_attrib(candidate, "address", tmp->ip);
iks_insert_attrib(candidate, "port", port);
iks_insert_attrib(candidate, "username", tmp->username);
iks_insert_attrib(candidate, "password", tmp->password);
iks_insert_attrib(candidate, "preference", preference);
if (tmp->protocol == AJI_PROTOCOL_UDP)
iks_insert_attrib(candidate, "protocol", "udp");
if (tmp->protocol == AJI_PROTOCOL_SSLTCP)
iks_insert_attrib(candidate, "protocol", "ssltcp");
if (tmp->type == AJI_CONNECT_STUN)
iks_insert_attrib(candidate, "type", "stun");
if (tmp->type == AJI_CONNECT_LOCAL)
iks_insert_attrib(candidate, "type", "local");
if (tmp->type == AJI_CONNECT_RELAY)
iks_insert_attrib(candidate, "type", "relay");
iks_insert_attrib(candidate, "network", "0");
iks_insert_attrib(candidate, "generation", "0");
ast_aji_send(c, iq);
}
p->laststun = 0;
safeout:
if (ours1)
ast_free(ours1);
if (ours2)
ast_free(ours2);
iks_delete(iq);
iks_delete(gtalk);
iks_delete(candidate);
iks_delete(transport);
return 1;
}
static struct gtalk_pvt *gtalk_alloc(struct gtalk *client, const char *us, const char *them, const char *sid)
{
struct gtalk_pvt *tmp = NULL;
struct aji_resource *resources = NULL;
struct aji_buddy *buddy;
char idroster[200];
char *data, *exten = NULL;
ast_debug(1, "The client is %s for alloc\n", client->name);
if (!sid && !strchr(them, '/')) { /* I started call! */
if (!strcasecmp(client->name, "guest")) {
buddy = ASTOBJ_CONTAINER_FIND(&client->connection->buddies, them);
if (buddy)
resources = buddy->resources;
} else if (client->buddy)
resources = client->buddy->resources;
while (resources) {
if (resources->cap->jingle) {
break;
}
resources = resources->next;
}
if (resources)
snprintf(idroster, sizeof(idroster), "%s/%s", them, resources->resource);
else {
ast_log(LOG_ERROR, "no gtalk capable clients to talk to.\n");
return NULL;
}
}
if (!(tmp = ast_calloc(1, sizeof(*tmp)))) {
return NULL;
}
memcpy(&tmp->prefs, &client->prefs, sizeof(struct ast_codec_pref));
if (sid) {
ast_copy_string(tmp->sid, sid, sizeof(tmp->sid));
ast_copy_string(tmp->them, them, sizeof(tmp->them));
ast_copy_string(tmp->us, us, sizeof(tmp->us));
} else {
snprintf(tmp->sid, sizeof(tmp->sid), "%08lx%08lx", ast_random(), ast_random());
ast_copy_string(tmp->them, idroster, sizeof(tmp->them));
ast_copy_string(tmp->us, us, sizeof(tmp->us));
tmp->initiator = 1;
}
/* clear codecs */
if (!(tmp->rtp = ast_rtp_instance_new("asterisk", sched, &bindaddr, NULL))) {
ast_log(LOG_ERROR, "Failed to create a new RTP instance (possibly an invalid bindaddr?)\n");
ast_free(tmp);
return NULL;
}
ast_rtp_instance_set_prop(tmp->rtp, AST_RTP_PROPERTY_RTCP, 1);
ast_rtp_codecs_payloads_clear(ast_rtp_instance_get_codecs(tmp->rtp), tmp->rtp);
/* add user configured codec capabilites */
if (client->capability)
tmp->capability = client->capability;
else if (global_capability)
tmp->capability = global_capability;
tmp->parent = client;
if (!tmp->rtp) {
ast_log(LOG_WARNING, "Out of RTP sessions?\n");
ast_free(tmp);
return NULL;
}
/* Set CALLERID(name) to the full JID of the remote peer */
ast_copy_string(tmp->cid_name, tmp->them, sizeof(tmp->cid_name));
if(strchr(tmp->us, '/')) {
data = ast_strdupa(tmp->us);
exten = strsep(&data, "/");
} else
exten = tmp->us;
ast_copy_string(tmp->exten, exten, sizeof(tmp->exten));
ast_mutex_init(&tmp->lock);
ast_mutex_lock(&gtalklock);
tmp->next = client->p;
client->p = tmp;
ast_mutex_unlock(&gtalklock);
return tmp;
}
/*! \brief Start new gtalk channel */
static struct ast_channel *gtalk_new(struct gtalk *client, struct gtalk_pvt *i, int state, const char *title, const char *linkedid)
{
struct ast_channel *tmp;
int fmt;
int what;
const char *n2;
if (title)
n2 = title;
else
n2 = i->us;
tmp = ast_channel_alloc(1, state, i->cid_num, i->cid_name, linkedid, client->accountcode, i->exten, client->context, client->amaflags, "Gtalk/%s-%04lx", n2, ast_random() & 0xffff);
if (!tmp) {
ast_log(LOG_WARNING, "Unable to allocate Gtalk channel structure!\n");
return NULL;
}
tmp->tech = &gtalk_tech;
/* Select our native format based on codec preference until we receive
something from another device to the contrary. */
if (i->jointcapability)
what = i->jointcapability;
else if (i->capability)
what = i->capability;
else
what = global_capability;
/* Set Frame packetization */
if (i->rtp)
ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(i->rtp), i->rtp, &i->prefs);
tmp->nativeformats = ast_codec_choose(&i->prefs, what, 1) | (i->jointcapability & AST_FORMAT_VIDEO_MASK);
fmt = ast_best_codec(tmp->nativeformats);
if (i->rtp) {
ast_rtp_instance_set_prop(i->rtp, AST_RTP_PROPERTY_STUN, 1);
ast_channel_set_fd(tmp, 0, ast_rtp_instance_fd(i->rtp, 0));
ast_channel_set_fd(tmp, 1, ast_rtp_instance_fd(i->rtp, 1));
}
if (i->vrtp) {
ast_rtp_instance_set_prop(i->vrtp, AST_RTP_PROPERTY_STUN, 1);
ast_channel_set_fd(tmp, 2, ast_rtp_instance_fd(i->vrtp, 0));
ast_channel_set_fd(tmp, 3, ast_rtp_instance_fd(i->vrtp, 1));
}
if (state == AST_STATE_RING)
tmp->rings = 1;
tmp->adsicpe = AST_ADSI_UNAVAILABLE;
tmp->writeformat = fmt;
tmp->rawwriteformat = fmt;
tmp->readformat = fmt;
tmp->rawreadformat = fmt;
tmp->tech_pvt = i;
tmp->callgroup = client->callgroup;
tmp->pickupgroup = client->pickupgroup;
tmp->cid.cid_pres = client->callingpres;
if (!ast_strlen_zero(client->accountcode))
ast_string_field_set(tmp, accountcode, client->accountcode);
if (client->amaflags)
tmp->amaflags = client->amaflags;
if (!ast_strlen_zero(client->language))
ast_string_field_set(tmp, language, client->language);
if (!ast_strlen_zero(client->musicclass))
ast_string_field_set(tmp, musicclass, client->musicclass);
if (!ast_strlen_zero(client->parkinglot))
ast_string_field_set(tmp, parkinglot, client->parkinglot);
i->owner = tmp;
ast_module_ref(ast_module_info->self);
ast_copy_string(tmp->context, client->context, sizeof(tmp->context));
ast_copy_string(tmp->exten, i->exten, sizeof(tmp->exten));
if (!ast_strlen_zero(i->exten) && strcmp(i->exten, "s"))
tmp->cid.cid_dnid = ast_strdup(i->exten);
tmp->priority = 1;
if (i->rtp)
ast_jb_configure(tmp, &global_jbconf);
if (state != AST_STATE_DOWN && ast_pbx_start(tmp)) {
ast_log(LOG_WARNING, "Unable to start PBX on %s\n", tmp->name);
tmp->hangupcause = AST_CAUSE_SWITCH_CONGESTION;
ast_hangup(tmp);
tmp = NULL;
} else {
manager_event(EVENT_FLAG_SYSTEM, "ChannelUpdate",
"Channel: %s\r\nChanneltype: %s\r\nGtalk-SID: %s\r\n",
i->owner ? i->owner->name : "", "Gtalk", i->sid);
}
return tmp;
}
static int gtalk_action(struct gtalk *client, struct gtalk_pvt *p, const char *action)
{
iks *request, *session = NULL;
int res = -1;
char *lowerthem = NULL;
request = iks_new("iq");
if (request) {
iks_insert_attrib(request, "type", "set");
iks_insert_attrib(request, "from", p->us);
iks_insert_attrib(request, "to", p->them);
iks_insert_attrib(request, "id", client->connection->mid);
ast_aji_increment_mid(client->connection->mid);
session = iks_new("session");
if (session) {
iks_insert_attrib(session, "type", action);
iks_insert_attrib(session, "id", p->sid);
/* put the initiator attribute to lower case if we receive the call
* otherwise GoogleTalk won't establish the session */
if (!p->initiator) {
char c;
char *t = lowerthem = ast_strdupa(p->them);
while (((c = *t) != '/') && (*t++ = tolower(c)));
}
iks_insert_attrib(session, "initiator", p->initiator ? p->us : lowerthem);
iks_insert_attrib(session, "xmlns", "http://www.google.com/session");
iks_insert_node(request, session);
ast_aji_send(client->connection, request);
res = 0;
}
}
iks_delete(session);
iks_delete(request);
return res;
}
static void gtalk_free_candidates(struct gtalk_candidate *candidate)
{
struct gtalk_candidate *last;
while (candidate) {
last = candidate;
candidate = candidate->next;
ast_free(last);
}
}
static void gtalk_free_pvt(struct gtalk *client, struct gtalk_pvt *p)
{
struct gtalk_pvt *cur, *prev = NULL;
cur = client->p;
while (cur) {
if (cur == p) {
if (prev)
prev->next = p->next;
else
client->p = p->next;
break;
}
prev = cur;
cur = cur->next;
}
if (p->ringrule)
iks_filter_remove_rule(p->parent->connection->f, p->ringrule);
if (p->owner)
ast_log(LOG_WARNING, "Uh oh, there's an owner, this is going to be messy.\n");
if (p->rtp)
ast_rtp_instance_destroy(p->rtp);
if (p->vrtp)
ast_rtp_instance_destroy(p->vrtp);
gtalk_free_candidates(p->theircandidates);
ast_free(p);
}
static int gtalk_newcall(struct gtalk *client, ikspak *pak)
{
struct gtalk_pvt *p, *tmp = client->p;
struct ast_channel *chan;
int res;
iks *codec;
char *from = NULL;
char s1[BUFSIZ], s2[BUFSIZ], s3[BUFSIZ];
int peernoncodeccapability;
/* Make sure our new call doesn't exist yet */
from = iks_find_attrib(pak->x,"to");
if(!from)
from = client->connection->jid->full;
while (tmp) {
if (iks_find_with_attrib(pak->x, "session", "id", tmp->sid)) {
ast_log(LOG_NOTICE, "Ignoring duplicate call setup on SID %s\n", tmp->sid);
gtalk_response(client, from, pak, "out-of-order", NULL);
return -1;
}
tmp = tmp->next;
}
if (!strcasecmp(client->name, "guest")){
/* the guest account is not tied to any configured XMPP client,
let's set it now */
client->connection = ast_aji_get_client(from);
if (!client->connection) {
ast_log(LOG_ERROR, "No XMPP client to talk to, us (partial JID) : %s\n", from);
return -1;
}
}
p = gtalk_alloc(client, from, pak->from->full, iks_find_attrib(pak->query, "id"));
if (!p) {
ast_log(LOG_WARNING, "Unable to allocate gtalk structure!\n");
return -1;
}
chan = gtalk_new(client, p, AST_STATE_DOWN, pak->from->user, NULL);
if (!chan) {
gtalk_free_pvt(client, p);
return -1;
}
ast_mutex_lock(&p->lock);
ast_copy_string(p->them, pak->from->full, sizeof(p->them));
if (iks_find_attrib(pak->query, "id")) {
ast_copy_string(p->sid, iks_find_attrib(pak->query, "id"),
sizeof(p->sid));
}
/* codec points to the first <payload-type/> tag */
Merged revisions 185362 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185362 | dbrooks | 2009-03-31 11:37:12 -0500 (Tue, 31 Mar 2009) | 35 lines Fix incorrect parsing in chan_gtalk when xmpp contains extra whitespaces To drill into the xmpp to find the capabilities between channels, chan_gtalk calls iks_child() and iks_next(). iks_child() and iks_next() are functions in the iksemel xml parsing library that traverse xml nodes. The bug here is that both iks_child() and iks_next() will return the next iks_struct node *regardless* of type. chan_gtalk expects the next node to be of type IKS_TAG, which in most cases, it is, but in this case (a call being made from the Empathy IM client), there exists iks_struct nodes which are not IKS_TAG data (they are extraneous whitespaces), and chan_gtalk doesn't handle that case, so capabilities don't match, and a call cannot be made. iks_first_tag() and iks_next_tag(), on the other hand, will not return the very next iks_struct, but will check to see if the next iks_struct is of type IKS_TAG. If it isn't, it will be skipped, and the next struct of type IKS_TAG it finds will be returned. This assures that chan_gtalk will find the iks_struct it is looking for. This fix simply changes all calls to iks_child() and iks_next() to become calls to iks_first_tag() and iks_next_tag(), which resolves the capability matching. The following is a payload listing from Empathy, which, due to the extraneous whitespace, will not be parsed correctly by iksemel: <iq from='dbrooksjab@235-22-24-10/Telepathy' to='astjab@235-22-24-10/asterisk' type='set' id='542757715704'> <session xmlns='http://www.google.com/session' initiator='dbrooksjab@235-22-24-10/Telepathy' type='initiate' id='1837267342'> <description xmlns='http://www.google.com/session/phone'> <payload-type clockrate='16000' name='speex' id='96'/> <payload-type clockrate='8000' name='PCMA' id='8'/> <payload-type clockrate='8000' name='PCMU' id='0'/> <payload-type clockrate='90000' name='MPA' id='97'/> <payload-type clockrate='16000' name='SIREN' id='98'/> <payload-type clockrate='8000' name='telephone-event' id='99'/> </description> </session> </iq> Review: http://reviewboard.digium.com/r/181/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@185363 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-31 16:46:57 +00:00
codec = iks_first_tag(iks_first_tag(iks_first_tag(pak->x)));
while (codec) {
ast_rtp_codecs_payloads_set_m_type(ast_rtp_instance_get_codecs(p->rtp), p->rtp, atoi(iks_find_attrib(codec, "id")));
ast_rtp_codecs_payloads_set_rtpmap_type(ast_rtp_instance_get_codecs(p->rtp), p->rtp, atoi(iks_find_attrib(codec, "id")), "audio", iks_find_attrib(codec, "name"), 0);
Merged revisions 185362 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185362 | dbrooks | 2009-03-31 11:37:12 -0500 (Tue, 31 Mar 2009) | 35 lines Fix incorrect parsing in chan_gtalk when xmpp contains extra whitespaces To drill into the xmpp to find the capabilities between channels, chan_gtalk calls iks_child() and iks_next(). iks_child() and iks_next() are functions in the iksemel xml parsing library that traverse xml nodes. The bug here is that both iks_child() and iks_next() will return the next iks_struct node *regardless* of type. chan_gtalk expects the next node to be of type IKS_TAG, which in most cases, it is, but in this case (a call being made from the Empathy IM client), there exists iks_struct nodes which are not IKS_TAG data (they are extraneous whitespaces), and chan_gtalk doesn't handle that case, so capabilities don't match, and a call cannot be made. iks_first_tag() and iks_next_tag(), on the other hand, will not return the very next iks_struct, but will check to see if the next iks_struct is of type IKS_TAG. If it isn't, it will be skipped, and the next struct of type IKS_TAG it finds will be returned. This assures that chan_gtalk will find the iks_struct it is looking for. This fix simply changes all calls to iks_child() and iks_next() to become calls to iks_first_tag() and iks_next_tag(), which resolves the capability matching. The following is a payload listing from Empathy, which, due to the extraneous whitespace, will not be parsed correctly by iksemel: <iq from='dbrooksjab@235-22-24-10/Telepathy' to='astjab@235-22-24-10/asterisk' type='set' id='542757715704'> <session xmlns='http://www.google.com/session' initiator='dbrooksjab@235-22-24-10/Telepathy' type='initiate' id='1837267342'> <description xmlns='http://www.google.com/session/phone'> <payload-type clockrate='16000' name='speex' id='96'/> <payload-type clockrate='8000' name='PCMA' id='8'/> <payload-type clockrate='8000' name='PCMU' id='0'/> <payload-type clockrate='90000' name='MPA' id='97'/> <payload-type clockrate='16000' name='SIREN' id='98'/> <payload-type clockrate='8000' name='telephone-event' id='99'/> </description> </session> </iq> Review: http://reviewboard.digium.com/r/181/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@185363 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-31 16:46:57 +00:00
codec = iks_next_tag(codec);
}
/* Now gather all of the codecs that we are asked for */
ast_rtp_codecs_payload_formats(ast_rtp_instance_get_codecs(p->rtp), &p->peercapability, &peernoncodeccapability);
p->jointcapability = p->capability & p->peercapability;
ast_mutex_unlock(&p->lock);
ast_setstate(chan, AST_STATE_RING);
if (!p->jointcapability) {
ast_log(LOG_WARNING, "Capabilities don't match : us - %s, peer - %s, combined - %s \n", ast_getformatname_multiple(s1, BUFSIZ, p->capability),
ast_getformatname_multiple(s2, BUFSIZ, p->peercapability),
ast_getformatname_multiple(s3, BUFSIZ, p->jointcapability));
/* close session if capabilities don't match */
gtalk_action(client, p, "reject");
p->alreadygone = 1;
gtalk_hangup(chan);
Convert the ast_channel data structure over to the astobj2 framework. There is a lot that could be said about this, but the patch is a big improvement for performance, stability, code maintainability, and ease of future code development. The channel list is no longer an unsorted linked list. The main container for channels is an astobj2 hash table. All of the code related to searching for channels or iterating active channels has been rewritten. Let n be the number of active channels. Iterating the channel list has gone from O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1). Searching for a channel by extension is still O(n), but uses a new method for doing so, which is more efficient. The ast_channel object is now a reference counted object. The benefits here are plentiful. Some benefits directly related to issues in the previous code include: 1) When threads other than the channel thread owning a channel wanted access to a channel, it had to hold the lock on it to ensure that it didn't go away. This is no longer a requirement. Holding a reference is sufficient. 2) There are places that now require less dealing with channel locks. 3) There are places where channel locks are held for much shorter periods of time. 4) There are places where dealing with more than one channel at a time becomes _MUCH_ easier. ChanSpy is a great example of this. Writing code in the future that deals with multiple channels will be much easier. Some additional information regarding channel locking and reference count handling can be found in channel.h, where a new section has been added that discusses some of the rules associated with it. Mark Michelson also assisted with the development of this patch. He did the conversion of ChanSpy and introduced a new API, ast_autochan, which makes it much easier to deal with holding on to a channel pointer for an extended period of time and having it get automatically updated if the channel gets masqueraded. Mark was also a huge help in the code review process. Thanks to David Vossel for his assistance with this branch, as well. David did the conversion of the DAHDIScan application by making it become a wrapper for ChanSpy internally. The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch. Review: http://reviewboard.digium.com/r/203/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@190423 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-24 14:04:26 +00:00
ast_channel_release(chan);
return -1;
}
res = ast_pbx_start(chan);
switch (res) {
case AST_PBX_FAILED:
ast_log(LOG_WARNING, "Failed to start PBX :(\n");
gtalk_response(client, from, pak, "service-unavailable", NULL);
break;
case AST_PBX_CALL_LIMIT:
ast_log(LOG_WARNING, "Failed to start PBX (call limit reached) \n");
gtalk_response(client, from, pak, "service-unavailable", NULL);
break;
case AST_PBX_SUCCESS:
gtalk_response(client, from, pak, NULL, NULL);
gtalk_invite_response(p, p->them, p->us,p->sid, 0);
gtalk_create_candidates(client, p, p->sid, p->them, p->us);
/* nothing to do */
break;
}
return 1;
}
static int gtalk_update_stun(struct gtalk *client, struct gtalk_pvt *p)
{
struct gtalk_candidate *tmp;
struct hostent *hp;
struct ast_hostent ahp;
struct sockaddr_in sin = { 0, };
struct sockaddr_in aux = { 0, };
if (time(NULL) == p->laststun)
return 0;
tmp = p->theircandidates;
p->laststun = time(NULL);
while (tmp) {
char username[256];
/* Find the IP address of the host */
hp = ast_gethostbyname(tmp->ip, &ahp);
sin.sin_family = AF_INET;
memcpy(&sin.sin_addr, hp->h_addr, sizeof(sin.sin_addr));
sin.sin_port = htons(tmp->port);
snprintf(username, sizeof(username), "%s%s", tmp->username,
p->ourcandidates->username);
/* Find out the result of the STUN */
ast_rtp_instance_get_remote_address(p->rtp, &aux);
/* If the STUN result is different from the IP of the hostname,
lock on the stun IP of the hostname advertised by the
remote client */
if (aux.sin_addr.s_addr &&
aux.sin_addr.s_addr != sin.sin_addr.s_addr)
ast_rtp_instance_stun_request(p->rtp, &aux, username);
else
ast_rtp_instance_stun_request(p->rtp, &sin, username);
if (aux.sin_addr.s_addr) {
ast_debug(4, "Receiving RTP traffic from IP %s, matches with remote candidate's IP %s\n", ast_inet_ntoa(aux.sin_addr), tmp->ip);
ast_debug(4, "Sending STUN request to %s\n", tmp->ip);
}
tmp = tmp->next;
}
return 1;
}
static int gtalk_add_candidate(struct gtalk *client, ikspak *pak)
{
struct gtalk_pvt *p = NULL, *tmp = NULL;
struct aji_client *c = client->connection;
struct gtalk_candidate *newcandidate = NULL;
iks *traversenodes = NULL, *receipt = NULL;
char *from;
from = iks_find_attrib(pak->x,"to");
if(!from)
from = c->jid->full;
for (tmp = client->p; tmp; tmp = tmp->next) {
if (iks_find_with_attrib(pak->x, "session", "id", tmp->sid)) {
p = tmp;
break;
}
}
if (!p)
return -1;
traversenodes = pak->query;
while(traversenodes) {
if(!strcasecmp(iks_name(traversenodes), "session")) {
Merged revisions 185362 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185362 | dbrooks | 2009-03-31 11:37:12 -0500 (Tue, 31 Mar 2009) | 35 lines Fix incorrect parsing in chan_gtalk when xmpp contains extra whitespaces To drill into the xmpp to find the capabilities between channels, chan_gtalk calls iks_child() and iks_next(). iks_child() and iks_next() are functions in the iksemel xml parsing library that traverse xml nodes. The bug here is that both iks_child() and iks_next() will return the next iks_struct node *regardless* of type. chan_gtalk expects the next node to be of type IKS_TAG, which in most cases, it is, but in this case (a call being made from the Empathy IM client), there exists iks_struct nodes which are not IKS_TAG data (they are extraneous whitespaces), and chan_gtalk doesn't handle that case, so capabilities don't match, and a call cannot be made. iks_first_tag() and iks_next_tag(), on the other hand, will not return the very next iks_struct, but will check to see if the next iks_struct is of type IKS_TAG. If it isn't, it will be skipped, and the next struct of type IKS_TAG it finds will be returned. This assures that chan_gtalk will find the iks_struct it is looking for. This fix simply changes all calls to iks_child() and iks_next() to become calls to iks_first_tag() and iks_next_tag(), which resolves the capability matching. The following is a payload listing from Empathy, which, due to the extraneous whitespace, will not be parsed correctly by iksemel: <iq from='dbrooksjab@235-22-24-10/Telepathy' to='astjab@235-22-24-10/asterisk' type='set' id='542757715704'> <session xmlns='http://www.google.com/session' initiator='dbrooksjab@235-22-24-10/Telepathy' type='initiate' id='1837267342'> <description xmlns='http://www.google.com/session/phone'> <payload-type clockrate='16000' name='speex' id='96'/> <payload-type clockrate='8000' name='PCMA' id='8'/> <payload-type clockrate='8000' name='PCMU' id='0'/> <payload-type clockrate='90000' name='MPA' id='97'/> <payload-type clockrate='16000' name='SIREN' id='98'/> <payload-type clockrate='8000' name='telephone-event' id='99'/> </description> </session> </iq> Review: http://reviewboard.digium.com/r/181/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@185363 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-31 16:46:57 +00:00
traversenodes = iks_first_tag(traversenodes);
continue;
}
if(!strcasecmp(iks_name(traversenodes), "transport")) {
Merged revisions 185362 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185362 | dbrooks | 2009-03-31 11:37:12 -0500 (Tue, 31 Mar 2009) | 35 lines Fix incorrect parsing in chan_gtalk when xmpp contains extra whitespaces To drill into the xmpp to find the capabilities between channels, chan_gtalk calls iks_child() and iks_next(). iks_child() and iks_next() are functions in the iksemel xml parsing library that traverse xml nodes. The bug here is that both iks_child() and iks_next() will return the next iks_struct node *regardless* of type. chan_gtalk expects the next node to be of type IKS_TAG, which in most cases, it is, but in this case (a call being made from the Empathy IM client), there exists iks_struct nodes which are not IKS_TAG data (they are extraneous whitespaces), and chan_gtalk doesn't handle that case, so capabilities don't match, and a call cannot be made. iks_first_tag() and iks_next_tag(), on the other hand, will not return the very next iks_struct, but will check to see if the next iks_struct is of type IKS_TAG. If it isn't, it will be skipped, and the next struct of type IKS_TAG it finds will be returned. This assures that chan_gtalk will find the iks_struct it is looking for. This fix simply changes all calls to iks_child() and iks_next() to become calls to iks_first_tag() and iks_next_tag(), which resolves the capability matching. The following is a payload listing from Empathy, which, due to the extraneous whitespace, will not be parsed correctly by iksemel: <iq from='dbrooksjab@235-22-24-10/Telepathy' to='astjab@235-22-24-10/asterisk' type='set' id='542757715704'> <session xmlns='http://www.google.com/session' initiator='dbrooksjab@235-22-24-10/Telepathy' type='initiate' id='1837267342'> <description xmlns='http://www.google.com/session/phone'> <payload-type clockrate='16000' name='speex' id='96'/> <payload-type clockrate='8000' name='PCMA' id='8'/> <payload-type clockrate='8000' name='PCMU' id='0'/> <payload-type clockrate='90000' name='MPA' id='97'/> <payload-type clockrate='16000' name='SIREN' id='98'/> <payload-type clockrate='8000' name='telephone-event' id='99'/> </description> </session> </iq> Review: http://reviewboard.digium.com/r/181/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@185363 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-31 16:46:57 +00:00
traversenodes = iks_first_tag(traversenodes);
continue;
}
if(!strcasecmp(iks_name(traversenodes), "candidate")) {
newcandidate = ast_calloc(1, sizeof(*newcandidate));
if (!newcandidate)
return 0;
ast_copy_string(newcandidate->name, iks_find_attrib(traversenodes, "name"),
sizeof(newcandidate->name));
ast_copy_string(newcandidate->ip, iks_find_attrib(traversenodes, "address"),
sizeof(newcandidate->ip));
newcandidate->port = atoi(iks_find_attrib(traversenodes, "port"));
ast_copy_string(newcandidate->username, iks_find_attrib(traversenodes, "username"),
sizeof(newcandidate->username));
ast_copy_string(newcandidate->password, iks_find_attrib(traversenodes, "password"),
sizeof(newcandidate->password));
newcandidate->preference = atof(iks_find_attrib(traversenodes, "preference"));
if (!strcasecmp(iks_find_attrib(traversenodes, "protocol"), "udp"))
newcandidate->protocol = AJI_PROTOCOL_UDP;
if (!strcasecmp(iks_find_attrib(traversenodes, "protocol"), "ssltcp"))
newcandidate->protocol = AJI_PROTOCOL_SSLTCP;
if (!strcasecmp(iks_find_attrib(traversenodes, "type"), "stun"))
newcandidate->type = AJI_CONNECT_STUN;
if (!strcasecmp(iks_find_attrib(traversenodes, "type"), "local"))
newcandidate->type = AJI_CONNECT_LOCAL;
if (!strcasecmp(iks_find_attrib(traversenodes, "type"), "relay"))
newcandidate->type = AJI_CONNECT_RELAY;
ast_copy_string(newcandidate->network, iks_find_attrib(traversenodes, "network"),
sizeof(newcandidate->network));
newcandidate->generation = atoi(iks_find_attrib(traversenodes, "generation"));
newcandidate->next = NULL;
newcandidate->next = p->theircandidates;
p->theircandidates = newcandidate;
p->laststun = 0;
gtalk_update_stun(p->parent, p);
newcandidate = NULL;
}
Merged revisions 185362 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185362 | dbrooks | 2009-03-31 11:37:12 -0500 (Tue, 31 Mar 2009) | 35 lines Fix incorrect parsing in chan_gtalk when xmpp contains extra whitespaces To drill into the xmpp to find the capabilities between channels, chan_gtalk calls iks_child() and iks_next(). iks_child() and iks_next() are functions in the iksemel xml parsing library that traverse xml nodes. The bug here is that both iks_child() and iks_next() will return the next iks_struct node *regardless* of type. chan_gtalk expects the next node to be of type IKS_TAG, which in most cases, it is, but in this case (a call being made from the Empathy IM client), there exists iks_struct nodes which are not IKS_TAG data (they are extraneous whitespaces), and chan_gtalk doesn't handle that case, so capabilities don't match, and a call cannot be made. iks_first_tag() and iks_next_tag(), on the other hand, will not return the very next iks_struct, but will check to see if the next iks_struct is of type IKS_TAG. If it isn't, it will be skipped, and the next struct of type IKS_TAG it finds will be returned. This assures that chan_gtalk will find the iks_struct it is looking for. This fix simply changes all calls to iks_child() and iks_next() to become calls to iks_first_tag() and iks_next_tag(), which resolves the capability matching. The following is a payload listing from Empathy, which, due to the extraneous whitespace, will not be parsed correctly by iksemel: <iq from='dbrooksjab@235-22-24-10/Telepathy' to='astjab@235-22-24-10/asterisk' type='set' id='542757715704'> <session xmlns='http://www.google.com/session' initiator='dbrooksjab@235-22-24-10/Telepathy' type='initiate' id='1837267342'> <description xmlns='http://www.google.com/session/phone'> <payload-type clockrate='16000' name='speex' id='96'/> <payload-type clockrate='8000' name='PCMA' id='8'/> <payload-type clockrate='8000' name='PCMU' id='0'/> <payload-type clockrate='90000' name='MPA' id='97'/> <payload-type clockrate='16000' name='SIREN' id='98'/> <payload-type clockrate='8000' name='telephone-event' id='99'/> </description> </session> </iq> Review: http://reviewboard.digium.com/r/181/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@185363 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-31 16:46:57 +00:00
traversenodes = iks_next_tag(traversenodes);
}
receipt = iks_new("iq");
iks_insert_attrib(receipt, "type", "result");
iks_insert_attrib(receipt, "from", from);
iks_insert_attrib(receipt, "to", iks_find_attrib(pak->x, "from"));
iks_insert_attrib(receipt, "id", iks_find_attrib(pak->x, "id"));
ast_aji_send(c, receipt);
iks_delete(receipt);
return 1;
}
static struct ast_frame *gtalk_rtp_read(struct ast_channel *ast, struct gtalk_pvt *p)
{
struct ast_frame *f;
if (!p->rtp)
return &ast_null_frame;
f = ast_rtp_instance_read(p->rtp, 0);
gtalk_update_stun(p->parent, p);
if (p->owner) {
/* We already hold the channel lock */
if (f->frametype == AST_FRAME_VOICE) {
if (f->subclass.codec != (p->owner->nativeformats & AST_FORMAT_AUDIO_MASK)) {
ast_debug(1, "Oooh, format changed to %s\n", ast_getformatname(f->subclass.codec));
p->owner->nativeformats =
(p->owner->nativeformats & AST_FORMAT_VIDEO_MASK) | f->subclass.codec;
ast_set_read_format(p->owner, p->owner->readformat);
ast_set_write_format(p->owner, p->owner->writeformat);
}
/* if ((ast_test_flag(p, SIP_DTMF) == SIP_DTMF_INBAND) && p->vad) {
f = ast_dsp_process(p->owner, p->vad, f);
if (option_debug && f && (f->frametype == AST_FRAME_DTMF))
ast_debug(1, "* Detected inband DTMF '%c'\n", f->subclass);
} */
}
}
return f;
}
static struct ast_frame *gtalk_read(struct ast_channel *ast)
{
struct ast_frame *fr;
struct gtalk_pvt *p = ast->tech_pvt;
ast_mutex_lock(&p->lock);
fr = gtalk_rtp_read(ast, p);
ast_mutex_unlock(&p->lock);
return fr;
}
/*! \brief Send frame to media channel (rtp) */
static int gtalk_write(struct ast_channel *ast, struct ast_frame *frame)
{
struct gtalk_pvt *p = ast->tech_pvt;
int res = 0;
char buf[256];
switch (frame->frametype) {
case AST_FRAME_VOICE:
if (!(frame->subclass.codec & ast->nativeformats)) {
ast_log(LOG_WARNING,
"Asked to transmit frame type %s, while native formats is %s (read/write = %s/%s)\n",
ast_getformatname(frame->subclass.codec),
ast_getformatname_multiple(buf, sizeof(buf), ast->nativeformats),
ast_getformatname(ast->readformat),
ast_getformatname(ast->writeformat));
return 0;
}
if (p) {
ast_mutex_lock(&p->lock);
if (p->rtp) {
res = ast_rtp_instance_write(p->rtp, frame);
}
ast_mutex_unlock(&p->lock);
}
break;
case AST_FRAME_VIDEO:
if (p) {
ast_mutex_lock(&p->lock);
if (p->vrtp) {
res = ast_rtp_instance_write(p->vrtp, frame);
}
ast_mutex_unlock(&p->lock);
}
break;
case AST_FRAME_IMAGE:
return 0;
break;
default:
ast_log(LOG_WARNING, "Can't send %d type frames with Gtalk write\n",
frame->frametype);
return 0;
}
return res;
}
static int gtalk_fixup(struct ast_channel *oldchan, struct ast_channel *newchan)
{
struct gtalk_pvt *p = newchan->tech_pvt;
ast_mutex_lock(&p->lock);
if ((p->owner != oldchan)) {
ast_mutex_unlock(&p->lock);
return -1;
}
if (p->owner == oldchan)
p->owner = newchan;
ast_mutex_unlock(&p->lock);
return 0;
}
static int gtalk_indicate(struct ast_channel *ast, int condition, const void *data, size_t datalen)
{
int res = 0;
switch (condition) {
case AST_CONTROL_HOLD:
ast_moh_start(ast, data, NULL);
break;
case AST_CONTROL_UNHOLD:
ast_moh_stop(ast);
break;
default:
ast_log(LOG_NOTICE, "Don't know how to indicate condition '%d'\n", condition);
res = -1;
}
return res;
}
static int gtalk_sendtext(struct ast_channel *chan, const char *text)
{
int res = 0;
struct aji_client *client = NULL;
struct gtalk_pvt *p = chan->tech_pvt;
if (!p->parent) {
ast_log(LOG_ERROR, "Parent channel not found\n");
return -1;
}
if (!p->parent->connection) {
ast_log(LOG_ERROR, "XMPP client not found\n");
return -1;
}
client = p->parent->connection;
res = ast_aji_send_chat(client, p->them, text);
return res;
}
2007-01-19 18:06:03 +00:00
static int gtalk_digit_begin(struct ast_channel *chan, char digit)
{
return gtalk_digit(chan, digit, 0);
}
static int gtalk_digit_end(struct ast_channel *chan, char digit, unsigned int duration)
{
return gtalk_digit(chan, digit, duration);
}
static int gtalk_digit(struct ast_channel *ast, char digit, unsigned int duration)
{
struct gtalk_pvt *p = ast->tech_pvt;
struct gtalk *client = p->parent;
iks *iq, *gtalk, *dtmf;
char buffer[2] = {digit, '\0'};
char *lowerthem = NULL;
iq = iks_new("iq");
gtalk = iks_new("gtalk");
dtmf = iks_new("dtmf");
if(!iq || !gtalk || !dtmf) {
iks_delete(iq);
iks_delete(gtalk);
iks_delete(dtmf);
ast_log(LOG_ERROR, "Did not send dtmf do to memory issue\n");
return -1;
}
iks_insert_attrib(iq, "type", "set");
iks_insert_attrib(iq, "to", p->them);
iks_insert_attrib(iq, "from", p->us);
iks_insert_attrib(iq, "id", client->connection->mid);
ast_aji_increment_mid(client->connection->mid);
iks_insert_attrib(gtalk, "xmlns", "http://jabber.org/protocol/gtalk");
iks_insert_attrib(gtalk, "action", "session-info");
/* put the initiator attribute to lower case if we receive the call
* otherwise GoogleTalk won't establish the session */
if (!p->initiator) {
char c;
char *t = lowerthem = ast_strdupa(p->them);
while (((c = *t) != '/') && (*t++ = tolower(c)));
}
iks_insert_attrib(gtalk, "initiator", p->initiator ? p->us: lowerthem);
iks_insert_attrib(gtalk, "sid", p->sid);
iks_insert_attrib(dtmf, "xmlns", "http://jabber.org/protocol/gtalk/info/dtmf");
iks_insert_attrib(dtmf, "code", buffer);
iks_insert_node(iq, gtalk);
iks_insert_node(gtalk, dtmf);
ast_mutex_lock(&p->lock);
if (ast->dtmff.frametype == AST_FRAME_DTMF_BEGIN || duration == 0) {
iks_insert_attrib(dtmf, "action", "button-down");
} else if (ast->dtmff.frametype == AST_FRAME_DTMF_END || duration != 0) {
iks_insert_attrib(dtmf, "action", "button-up");
}
ast_aji_send(client->connection, iq);
iks_delete(iq);
iks_delete(gtalk);
iks_delete(dtmf);
ast_mutex_unlock(&p->lock);
return 0;
}
static int gtalk_sendhtml(struct ast_channel *ast, int subclass, const char *data, int datalen)
{
ast_log(LOG_NOTICE, "XXX Implement gtalk sendhtml XXX\n");
return -1;
}
/* Not in use right now.
static int gtalk_auto_congest(void *nothing)
{
struct gtalk_pvt *p = nothing;
ast_mutex_lock(&p->lock);
if (p->owner) {
if (!ast_channel_trylock(p->owner)) {
ast_log(LOG_NOTICE, "Auto-congesting %s\n", p->owner->name);
ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
ast_channel_unlock(p->owner);
}
}
ast_mutex_unlock(&p->lock);
return 0;
}
*/
/*! \brief Initiate new call, part of PBX interface
* dest is the dial string */
static int gtalk_call(struct ast_channel *ast, char *dest, int timeout)
{
struct gtalk_pvt *p = ast->tech_pvt;
if ((ast->_state != AST_STATE_DOWN) && (ast->_state != AST_STATE_RESERVED)) {
ast_log(LOG_WARNING, "gtalk_call called on %s, neither down nor reserved\n", ast->name);
return -1;
}
ast_setstate(ast, AST_STATE_RING);
if (!p->ringrule) {
ast_copy_string(p->ring, p->parent->connection->mid, sizeof(p->ring));
p->ringrule = iks_filter_add_rule(p->parent->connection->f, gtalk_ringing_ack, p,
IKS_RULE_ID, p->ring, IKS_RULE_DONE);
} else
ast_log(LOG_WARNING, "Whoa, already have a ring rule!\n");
gtalk_invite(p, p->them, p->us, p->sid, 1);
gtalk_create_candidates(p->parent, p, p->sid, p->them, p->us);
return 0;
}
/*! \brief Hangup a call through the gtalk proxy channel */
static int gtalk_hangup(struct ast_channel *ast)
{
struct gtalk_pvt *p = ast->tech_pvt;
struct gtalk *client;
ast_mutex_lock(&p->lock);
client = p->parent;
p->owner = NULL;
ast->tech_pvt = NULL;
if (!p->alreadygone)
gtalk_action(client, p, "terminate");
ast_mutex_unlock(&p->lock);
gtalk_free_pvt(client, p);
ast_module_unref(ast_module_info->self);
return 0;
}
/*! \brief Part of PBX interface */
static struct ast_channel *gtalk_request(const char *type, format_t format, const struct ast_channel *requestor, void *data, int *cause)
{
struct gtalk_pvt *p = NULL;
struct gtalk *client = NULL;
char *sender = NULL, *to = NULL, *s = NULL;
struct ast_channel *chan = NULL;
if (data) {
s = ast_strdupa(data);
if (s) {
sender = strsep(&s, "/");
if (sender && (sender[0] != '\0'))
to = strsep(&s, "/");
if (!to) {
ast_log(LOG_ERROR, "Bad arguments in Gtalk Dialstring: %s\n", (char*) data);
return NULL;
}
}
}
client = find_gtalk(to, sender);
if (!client) {
ast_log(LOG_WARNING, "Could not find recipient.\n");
return NULL;
}
if (!strcasecmp(client->name, "guest")){
/* the guest account is not tied to any configured XMPP client,
let's set it now */
client->connection = ast_aji_get_client(sender);
if (!client->connection) {
ast_log(LOG_ERROR, "No XMPP client to talk to, us (partial JID) : %s\n", sender);
ASTOBJ_UNREF(client, gtalk_member_destroy);
return NULL;
}
}
ASTOBJ_WRLOCK(client);
p = gtalk_alloc(client, strchr(sender, '@') ? sender : client->connection->jid->full, strchr(to, '@') ? to : client->user, NULL);
if (p)
chan = gtalk_new(client, p, AST_STATE_DOWN, to, requestor ? requestor->linkedid : NULL);
ASTOBJ_UNLOCK(client);
return chan;
}
/*! \brief CLI command "gtalk show channels" */
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85460 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-11 19:03:06 +00:00
static char *gtalk_show_channels(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
#define FORMAT "%-30.30s %-30.30s %-15.15s %-5.5s %-5.5s \n"
struct gtalk_pvt *p;
struct ast_channel *chan;
int numchans = 0;
char them[AJI_MAX_JIDLEN];
char *jid = NULL;
char *resource = NULL;
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85460 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-11 19:03:06 +00:00
switch (cmd) {
case CLI_INIT:
e->command = "gtalk show channels";
e->usage =
"Usage: gtalk show channels\n"
" Shows current state of the Gtalk channels.\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
if (a->argc != 3)
return CLI_SHOWUSAGE;
ast_mutex_lock(&gtalklock);
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85460 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-11 19:03:06 +00:00
ast_cli(a->fd, FORMAT, "Channel", "Jabber ID", "Resource", "Read", "Write");
ASTOBJ_CONTAINER_TRAVERSE(&gtalk_list, 1, {
ASTOBJ_WRLOCK(iterator);
p = iterator->p;
while(p) {
chan = p->owner;
ast_copy_string(them, p->them, sizeof(them));
jid = them;
resource = strchr(them, '/');
if (!resource)
resource = "None";
else {
*resource = '\0';
resource ++;
}
if (chan)
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85460 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-11 19:03:06 +00:00
ast_cli(a->fd, FORMAT,
chan->name,
jid,
resource,
ast_getformatname(chan->readformat),
ast_getformatname(chan->writeformat)
);
else
ast_log(LOG_WARNING, "No available channel\n");
numchans ++;
p = p->next;
}
ASTOBJ_UNLOCK(iterator);
});
ast_mutex_unlock(&gtalklock);
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85460 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-11 19:03:06 +00:00
ast_cli(a->fd, "%d active gtalk channel%s\n", numchans, (numchans != 1) ? "s" : "");
return CLI_SUCCESS;
#undef FORMAT
}
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85460 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-11 19:03:06 +00:00
/*! \brief CLI command "gtalk reload" */
static char *gtalk_do_reload(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85460 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-11 19:03:06 +00:00
switch (cmd) {
case CLI_INIT:
e->command = "gtalk reload";
e->usage =
"Usage: gtalk reload\n"
" Reload gtalk channel driver.\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
ast_verbose("IT DOES WORK!\n");
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85460 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-11 19:03:06 +00:00
return CLI_SUCCESS;
}
static int gtalk_parser(void *data, ikspak *pak)
{
struct gtalk *client = ASTOBJ_REF((struct gtalk *) data);
if (iks_find_attrib(pak->x, "type") && !strcmp(iks_find_attrib (pak->x, "type"),"error")) {
ast_log(LOG_NOTICE, "Remote peer reported an error, trying to establish the call anyway\n");
}
else if (iks_find_with_attrib(pak->x, "session", "type", "initiate")) {
/* New call */
gtalk_newcall(client, pak);
} else if (iks_find_with_attrib(pak->x, "session", "type", "candidates") || iks_find_with_attrib(pak->x, "session", "type", "transport-info")) {
ast_debug(3, "About to add candidate!\n");
gtalk_add_candidate(client, pak);
ast_debug(3, "Candidate Added!\n");
} else if (iks_find_with_attrib(pak->x, "session", "type", "accept")) {
gtalk_is_answered(client, pak);
} else if (iks_find_with_attrib(pak->x, "session", "type", "transport-accept")) {
gtalk_is_accepted(client, pak);
} else if (iks_find_with_attrib(pak->x, "session", "type", "content-info") || iks_find_with_attrib(pak->x, "gtalk", "action", "session-info")) {
gtalk_handle_dtmf(client, pak);
} else if (iks_find_with_attrib(pak->x, "session", "type", "terminate")) {
gtalk_hangup_farend(client, pak);
} else if (iks_find_with_attrib(pak->x, "session", "type", "reject")) {
gtalk_hangup_farend(client, pak);
}
ASTOBJ_UNREF(client, gtalk_member_destroy);
return IKS_FILTER_EAT;
}
/* Not using this anymore probably take out soon
static struct gtalk_candidate *gtalk_create_candidate(char *args)
{
char *name, *type, *preference, *protocol;
struct gtalk_candidate *res;
res = ast_calloc(1, sizeof(*res));
if (args)
name = args;
if ((args = strchr(args, ','))) {
*args = '\0';
args++;
preference = args;
}
if ((args = strchr(args, ','))) {
*args = '\0';
args++;
protocol = args;
}
if ((args = strchr(args, ','))) {
*args = '\0';
args++;
type = args;
}
if (name)
ast_copy_string(res->name, name, sizeof(res->name));
if (preference) {
res->preference = atof(preference);
}
if (protocol) {
if (!strcasecmp("udp", protocol))
res->protocol = AJI_PROTOCOL_UDP;
if (!strcasecmp("ssltcp", protocol))
res->protocol = AJI_PROTOCOL_SSLTCP;
}
if (type) {
if (!strcasecmp("stun", type))
res->type = AJI_CONNECT_STUN;
if (!strcasecmp("local", type))
res->type = AJI_CONNECT_LOCAL;
if (!strcasecmp("relay", type))
res->type = AJI_CONNECT_RELAY;
}
return res;
}
*/
static int gtalk_create_member(char *label, struct ast_variable *var, int allowguest,
struct ast_codec_pref prefs, char *context,
struct gtalk *member)
{
struct aji_client *client;
if (!member)
ast_log(LOG_WARNING, "Out of memory.\n");
ast_copy_string(member->name, label, sizeof(member->name));
ast_copy_string(member->user, label, sizeof(member->user));
ast_copy_string(member->context, context, sizeof(member->context));
member->allowguest = allowguest;
member->prefs = prefs;
while (var) {
#if 0
struct gtalk_candidate *candidate = NULL;
#endif
if (!strcasecmp(var->name, "username"))
ast_copy_string(member->user, var->value, sizeof(member->user));
else if (!strcasecmp(var->name, "disallow"))
ast_parse_allow_disallow(&member->prefs, &member->capability, var->value, 0);
else if (!strcasecmp(var->name, "allow"))
ast_parse_allow_disallow(&member->prefs, &member->capability, var->value, 1);
else if (!strcasecmp(var->name, "context"))
ast_copy_string(member->context, var->value, sizeof(member->context));
else if (!strcasecmp(var->name, "parkinglot"))
ast_copy_string(member->parkinglot, var->value, sizeof(member->parkinglot));
#if 0
else if (!strcasecmp(var->name, "candidate")) {
candidate = gtalk_create_candidate(var->value);
if (candidate) {
candidate->next = member->ourcandidates;
member->ourcandidates = candidate;
}
}
#endif
else if (!strcasecmp(var->name, "connection")) {
if ((client = ast_aji_get_client(var->value))) {
member->connection = client;
iks_filter_add_rule(client->f, gtalk_parser, member,
IKS_RULE_TYPE, IKS_PAK_IQ,
IKS_RULE_FROM_PARTIAL, member->user,
IKS_RULE_NS, "http://www.google.com/session",
IKS_RULE_DONE);
} else {
ast_log(LOG_ERROR, "connection referenced not found!\n");
return 0;
}
}
var = var->next;
}
if (member->connection && member->user)
member->buddy = ASTOBJ_CONTAINER_FIND(&member->connection->buddies, member->user);
else {
ast_log(LOG_ERROR, "No Connection or Username!\n");
}
return 1;
}
static int gtalk_load_config(void)
{
char *cat = NULL;
struct ast_config *cfg = NULL;
char context[AST_MAX_CONTEXT];
char parkinglot[AST_MAX_CONTEXT];
int allowguest = 1;
struct ast_variable *var;
struct gtalk *member;
struct ast_codec_pref prefs;
struct aji_client_container *clients;
struct gtalk_candidate *global_candidates = NULL;
struct hostent *hp;
struct ast_hostent ahp;
struct ast_flags config_flags = { 0 };
cfg = ast_config_load(GOOGLE_CONFIG, config_flags);
if (!cfg) {
return 0;
} else if (cfg == CONFIG_STATUS_FILEINVALID) {
ast_log(LOG_ERROR, "Config file %s is in an invalid format. Aborting.\n", GOOGLE_CONFIG);
return 0;
}
/* Copy the default jb config over global_jbconf */
memcpy(&global_jbconf, &default_jbconf, sizeof(struct ast_jb_conf));
cat = ast_category_browse(cfg, NULL);
for (var = ast_variable_browse(cfg, "general"); var; var = var->next) {
/* handle jb conf */
if (!ast_jb_read_conf(&global_jbconf, var->name, var->value))
continue;
if (!strcasecmp(var->name, "allowguest"))
allowguest =
(ast_true(ast_variable_retrieve(cfg, "general", "allowguest"))) ? 1 : 0;
else if (!strcasecmp(var->name, "disallow"))
ast_parse_allow_disallow(&prefs, &global_capability, var->value, 0);
else if (!strcasecmp(var->name, "allow"))
ast_parse_allow_disallow(&prefs, &global_capability, var->value, 1);
else if (!strcasecmp(var->name, "context"))
ast_copy_string(context, var->value, sizeof(context));
else if (!strcasecmp(var->name, "parkinglot"))
ast_copy_string(parkinglot, var->value, sizeof(parkinglot));
else if (!strcasecmp(var->name, "bindaddr")) {
if (!(hp = ast_gethostbyname(var->value, &ahp))) {
ast_log(LOG_WARNING, "Invalid address: %s\n", var->value);
} else {
memcpy(&bindaddr.sin_addr, hp->h_addr, sizeof(bindaddr.sin_addr));
}
}
/* Idea to allow for custom candidates */
/*
else if (!strcasecmp(var->name, "candidate")) {
candidate = gtalk_create_candidate(var->value);
if (candidate) {
candidate->next = global_candidates;
global_candidates = candidate;
}
}
*/
}
while (cat) {
if (strcasecmp(cat, "general")) {
var = ast_variable_browse(cfg, cat);
member = ast_calloc(1, sizeof(*member));
ASTOBJ_INIT(member);
ASTOBJ_WRLOCK(member);
if (!strcasecmp(cat, "guest")) {
ast_copy_string(member->name, "guest", sizeof(member->name));
ast_copy_string(member->user, "guest", sizeof(member->user));
ast_copy_string(member->context, context, sizeof(member->context));
ast_copy_string(member->parkinglot, parkinglot, sizeof(member->parkinglot));
member->allowguest = allowguest;
member->prefs = prefs;
while (var) {
if (!strcasecmp(var->name, "disallow"))
ast_parse_allow_disallow(&member->prefs, &member->capability,
var->value, 0);
else if (!strcasecmp(var->name, "allow"))
ast_parse_allow_disallow(&member->prefs, &member->capability,
var->value, 1);
else if (!strcasecmp(var->name, "context"))
ast_copy_string(member->context, var->value,
sizeof(member->context));
else if (!strcasecmp(var->name, "parkinglot"))
ast_copy_string(member->parkinglot, var->value,
sizeof(member->parkinglot));
/* Idea to allow for custom candidates */
/*
else if (!strcasecmp(var->name, "candidate")) {
candidate = gtalk_create_candidate(var->value);
if (candidate) {
candidate->next = member->ourcandidates;
member->ourcandidates = candidate;
}
}
*/
var = var->next;
}
ASTOBJ_UNLOCK(member);
clients = ast_aji_get_clients();
if (clients) {
ASTOBJ_CONTAINER_TRAVERSE(clients, 1, {
ASTOBJ_WRLOCK(iterator);
ASTOBJ_WRLOCK(member);
member->connection = NULL;
iks_filter_add_rule(iterator->f, gtalk_parser, member, IKS_RULE_TYPE, IKS_PAK_IQ, IKS_RULE_NS, "http://www.google.com/session", IKS_RULE_DONE);
iks_filter_add_rule(iterator->f, gtalk_parser, member, IKS_RULE_TYPE, IKS_PAK_IQ, IKS_RULE_NS, "http://jabber.org/protocol/gtalk", IKS_RULE_DONE);
ASTOBJ_UNLOCK(member);
ASTOBJ_UNLOCK(iterator);
});
ASTOBJ_CONTAINER_LINK(&gtalk_list, member);
ASTOBJ_UNREF(member, gtalk_member_destroy);
} else {
ASTOBJ_UNLOCK(member);
ASTOBJ_UNREF(member, gtalk_member_destroy);
}
} else {
ASTOBJ_UNLOCK(member);
if (gtalk_create_member(cat, var, allowguest, prefs, context, member))
ASTOBJ_CONTAINER_LINK(&gtalk_list, member);
ASTOBJ_UNREF(member, gtalk_member_destroy);
}
}
cat = ast_category_browse(cfg, cat);
}
gtalk_free_candidates(global_candidates);
return 1;
}
/*! \brief Load module into PBX, register channel */
static int load_module(void)
{
char *jabber_loaded = ast_module_helper("", "res_jabber.so", 0, 0, 0, 0);
free(jabber_loaded);
if (!jabber_loaded) {
/* If embedded, check for a different module name */
jabber_loaded = ast_module_helper("", "res_jabber", 0, 0, 0, 0);
free(jabber_loaded);
if (!jabber_loaded) {
ast_log(LOG_ERROR, "chan_gtalk.so depends upon res_jabber.so\n");
return AST_MODULE_LOAD_DECLINE;
}
}
ASTOBJ_CONTAINER_INIT(&gtalk_list);
if (!gtalk_load_config()) {
ast_log(LOG_ERROR, "Unable to read config file %s. Not loading module.\n", GOOGLE_CONFIG);
return 0;
}
sched = sched_context_create();
if (!sched)
ast_log(LOG_WARNING, "Unable to create schedule context\n");
io = io_context_create();
if (!io)
ast_log(LOG_WARNING, "Unable to create I/O context\n");
if (ast_find_ourip(&__ourip, bindaddr)) {
ast_log(LOG_WARNING, "Unable to get own IP address, Gtalk disabled\n");
return 0;
}
ast_rtp_glue_register(&gtalk_rtp_glue);
ast_cli_register_multiple(gtalk_cli, ARRAY_LEN(gtalk_cli));
/* Make sure we can register our channel type */
if (ast_channel_register(&gtalk_tech)) {
ast_log(LOG_ERROR, "Unable to register channel class %s\n", gtalk_tech.type);
return -1;
}
return 0;
}
/*! \brief Reload module */
static int reload(void)
{
return 0;
}
/*! \brief Unload the gtalk channel from Asterisk */
static int unload_module(void)
{
struct gtalk_pvt *privates = NULL;
ast_cli_unregister_multiple(gtalk_cli, ARRAY_LEN(gtalk_cli));
/* First, take us out of the channel loop */
ast_channel_unregister(&gtalk_tech);
ast_rtp_glue_unregister(&gtalk_rtp_glue);
if (!ast_mutex_lock(&gtalklock)) {
/* Hangup all interfaces if they have an owner */
ASTOBJ_CONTAINER_TRAVERSE(&gtalk_list, 1, {
ASTOBJ_WRLOCK(iterator);
privates = iterator->p;
while(privates) {
if (privates->owner)
ast_softhangup(privates->owner, AST_SOFTHANGUP_APPUNLOAD);
privates = privates->next;
}
iterator->p = NULL;
ASTOBJ_UNLOCK(iterator);
});
ast_mutex_unlock(&gtalklock);
} else {
ast_log(LOG_WARNING, "Unable to lock the monitor\n");
return -1;
}
ASTOBJ_CONTAINER_DESTROYALL(&gtalk_list, gtalk_member_destroy);
ASTOBJ_CONTAINER_DESTROY(&gtalk_list);
return 0;
}
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_DEFAULT, "Gtalk Channel Driver",
.load = load_module,
.unload = unload_module,
.reload = reload,
);