[svn-commits] mogorman: branch mogorman/asterisk-jabber r40950 - in
/team/mogorman/asterisk...
svn-commits at lists.digium.com
svn-commits at lists.digium.com
Wed Aug 23 12:41:39 MST 2006
Author: mogorman
Date: Wed Aug 23 14:41:38 2006
New Revision: 40950
URL: http://svn.digium.com/view/asterisk?rev=40950&view=rev
Log:
step one seperate jingle from gtalk as this
will make it easier to keep up and then phase out.
Added:
team/mogorman/asterisk-jabber/channels/chan_gtalk.c (with props)
team/mogorman/asterisk-jabber/configs/gtalk.conf.sample (with props)
Added: team/mogorman/asterisk-jabber/channels/chan_gtalk.c
URL: http://svn.digium.com/view/asterisk/team/mogorman/asterisk-jabber/channels/chan_gtalk.c?rev=40950&view=auto
==============================================================================
--- team/mogorman/asterisk-jabber/channels/chan_gtalk.c (added)
+++ team/mogorman/asterisk-jabber/channels/chan_gtalk.c Wed Aug 23 14:41:38 2006
@@ -1,0 +1,1733 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 1999 - 2005, Digium, Inc.
+ *
+ * Matt O'Gorman <mogorman at 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 at digium.com>
+ *
+ * \brief Gtalk Channel Driver
+ *
+ * \ingroup channel_drivers
+ */
+
+/*** MODULEINFO
+ <depend>iksemel</depend>
+ ***/
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include <unistd.h>
+#include <sys/socket.h>
+#include <errno.h>
+#include <stdlib.h>
+#include <fcntl.h>
+#include <netdb.h>
+#include <netinet/in.h>
+#include <arpa/inet.h>
+#include <sys/signal.h>
+#include <iksemel.h>
+
+#include "asterisk/lock.h"
+#include "asterisk/channel.h"
+#include "asterisk/config.h"
+#include "asterisk/logger.h"
+#include "asterisk/module.h"
+#include "asterisk/pbx.h"
+#include "asterisk/options.h"
+#include "asterisk/lock.h"
+#include "asterisk/sched.h"
+#include "asterisk/io.h"
+#include "asterisk/rtp.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_NODE "session"
+#define GOOGLE_NS "http://www.google.com/session"
+#define GOOGLE_SID "id"
+#define GOOGLE_ACCEPT "accept"
+#define GOOGLE_NEGOTIATE "candidates"
+
+
+/*! Global jitterbuffer configuration - by default, jb is disabled */
+static struct ast_jb_conf default_jbconf =
+{
+ .flags = 0,
+ .max_size = -1,
+ .resync_threshold = -1,
+ .impl = ""
+};
+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 from[100];
+ 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 *rtp; /*!< RTP audio session */
+ struct ast_rtp *vrtp; /*!< RTP video session */
+ int jointcapability; /*!< Supported capability at both ends (codecs ) */
+ int 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[100];
+ char context[100];
+ char accountcode[AST_MAX_ACCOUNT_CODE]; /*!< Account code */
+ int 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 const char type[] = "Gtalk";
+
+static int usecnt = 0;
+AST_MUTEX_DEFINE_STATIC(usecnt_lock);
+
+static int 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, int format, void *data, int *cause);
+static int gtalk_digit(struct ast_channel *ast, char digit);
+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_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 *from, const char *sid);
+/*----- RTP interface functions */
+static int gtalk_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp,
+ struct ast_rtp *vrtp, int codecs, int nat_active);
+static struct ast_rtp *gtalk_get_rtp_peer(struct ast_channel *chan);
+static int gtalk_get_codec(struct ast_channel *chan);
+
+/*! \brief PBX interface structure for channel registration */
+static const struct ast_channel_tech gtalk_tech = {
+ .type = type,
+ .description = "Gtalk Channel Driver",
+ .capabilities = ((AST_FORMAT_MAX_AUDIO << 1) - 1),
+ .requester = gtalk_request,
+ .send_digit = gtalk_digit,
+ .bridge = ast_rtp_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;
+
+
+/*! \brief RTP driver interface */
+static struct ast_rtp_protocol gtalk_rtp = {
+ type: "gtalk",
+ get_rtp_info: gtalk_get_rtp_peer,
+ set_rtp_peer: gtalk_set_rtp_peer,
+ get_codec: gtalk_get_codec,
+};
+
+static char externip[16];
+
+static struct gtalk_container gtalks;
+
+static void gtalk_member_destroy(struct gtalk *obj)
+{
+ free(obj);
+}
+
+static struct gtalk *find_gtalk(char *name, char *connection)
+{
+ struct gtalk *gtalk = NULL;
+
+ gtalk = ASTOBJ_CONTAINER_FIND(>alks, name);
+ if (!gtalk && strchr(name, '@'))
+ gtalk = ASTOBJ_CONTAINER_FIND_FULL(>alks, name, user,,, strcasecmp);
+
+ if (!gtalk) { /* guest call */
+ ASTOBJ_CONTAINER_TRAVERSE(>alks, 1, {
+ ASTOBJ_WRLOCK(iterator);
+ if (!strcasecmp(iterator->name, "guest")) {
+ if (!strcasecmp(iterator->connection->jid->partial, connection)) {
+ gtalk = iterator;
+ break;
+ } else if (!strcasecmp(iterator->connection->name, connection)) {
+ gtalk = iterator;
+ break;
+ }
+ }
+ ASTOBJ_UNLOCK(iterator);
+ });
+
+ }
+ return gtalk;
+}
+
+
+static void add_codec_to_answer(const struct gtalk_pvt *p, int codec, iks *dcodecs)
+{
+ char *format = ast_getformatname(codec);
+
+ if (!strcasecmp("ulaw", format)) {
+ iks *payload_eg711u, *payload_pcmu;
+ payload_pcmu = iks_new("payload-type");
+ iks_insert_attrib(payload_pcmu, "id", "0");
+ iks_insert_attrib(payload_pcmu, "name", "PCMU");
+ iks_insert_attrib(payload_pcmu, "xmlns", "http://www.google.com/session/phone");
+ payload_eg711u = iks_new("payload-type");
+ iks_insert_attrib(payload_eg711u, "id", "100");
+ iks_insert_attrib(payload_eg711u, "name", "EG711U");
+ iks_insert_attrib(payload_eg711u, "xmlns", "http://www.google.com/session/phone");
+ iks_insert_node(dcodecs, payload_pcmu);
+ iks_insert_node(dcodecs, payload_eg711u);
+ }
+ if (!strcasecmp("alaw", format)) {
+ iks *payload_eg711a;
+ iks *payload_pcma = iks_new("payload-type");
+ iks_insert_attrib(payload_pcma, "id", "8");
+ iks_insert_attrib(payload_pcma, "name", "PCMA");
+ iks_insert_attrib(payload_pcma, "xmlns", "http://www.google.com/session/phone");
+ 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, "xmlns", "http://www.google.com/session/phone");
+ iks_insert_node(dcodecs, payload_pcma);
+ iks_insert_node(dcodecs, payload_eg711a);
+ }
+ if (!strcasecmp("ilbc", format)) {
+ iks *payload_ilbc = iks_new("payload-type");
+ iks_insert_attrib(payload_ilbc, "id", "102");
+ iks_insert_attrib(payload_ilbc, "name", "iLBC");
+ iks_insert_attrib(payload_ilbc, "xmlns", "http://www.google.com/session/phone");
+ iks_insert_node(dcodecs, payload_ilbc);
+ }
+ if (!strcasecmp("g723", format)) {
+ iks *payload_g723 = iks_new("payload-type");
+ iks_insert_attrib(payload_g723, "id", "4");
+ iks_insert_attrib(payload_g723, "name", "G723");
+ iks_insert_attrib(payload_g723, "xmlns", "http://www.google.com/session/phone");
+ iks_insert_node(dcodecs, payload_g723);
+ }
+ ast_rtp_lookup_code(p->rtp, 1, codec);
+}
+
+static int gtalk_accept_call(struct gtalk *client, struct gtalk_pvt *p)
+{
+ struct gtalk_pvt *tmp = client->p;
+ struct aji_client *c = client->connection;
+ iks *iq, *gtalk, *dcodecs, *payload_red, *payload_audio, *payload_cn;
+ int x;
+ int pref_codec = 0;
+ int alreadysent = 0;
+
+ if (p->initiator)
+ return 1;
+
+ iq = iks_new("iq");
+ gtalk = iks_new(GOOGLE_NODE);
+ dcodecs = iks_new("description");
+ if (iq && gtalk && dcodecs) {
+ iks_insert_attrib(dcodecs, "xmlns", "http://www.google.com/session/phone");
+
+ for (x = 0; x < 32; x++) {
+ if (!(pref_codec = ast_codec_pref_index(&client->prefs, x)))
+ break;
+ if (!(client->capability & pref_codec))
+ continue;
+ if (alreadysent & pref_codec)
+ continue;
+ if (pref_codec <= AST_FORMAT_MAX_AUDIO)
+ add_codec_to_answer(p, pref_codec, dcodecs);
+ else
+ add_codec_to_answer(p, pref_codec, dcodecs);
+ alreadysent |= pref_codec;
+ }
+ payload_red = iks_new("payload-type");
+ iks_insert_attrib(payload_red, "id", "117");
+ iks_insert_attrib(payload_red, "name", "red");
+ iks_insert_attrib(payload_red, "xmlns", "http://www.google.com/session/phone");
+ payload_audio = iks_new("payload-type");
+ iks_insert_attrib(payload_audio, "id", "106");
+ iks_insert_attrib(payload_audio, "name", "audio/telephone-event");
+ iks_insert_attrib(payload_audio, "xmlns", "http://www.google.com/session/phone");
+ payload_cn = iks_new("payload-type");
+ iks_insert_attrib(payload_cn, "id", "13");
+ iks_insert_attrib(payload_cn, "name", "CN");
+ iks_insert_attrib(payload_cn, "xmlns", "http://www.google.com/session/phone");
+
+
+ iks_insert_attrib(iq, "type", "set");
+ iks_insert_attrib(iq, "to", (p->from) ? p->from : client->user);
+ 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", GOOGLE_ACCEPT);
+ iks_insert_attrib(gtalk, "initiator",
+ p->initiator ? client->connection->jid->full : p->from);
+ iks_insert_attrib(gtalk, GOOGLE_SID, tmp->sid);
+ iks_insert_node(iq, gtalk);
+ iks_insert_node(gtalk, dcodecs);
+ iks_insert_node(dcodecs, payload_red);
+ iks_insert_node(dcodecs, payload_audio);
+ iks_insert_node(dcodecs, payload_cn);
+
+ iks_send(c->p, iq);
+ iks_delete(payload_red);
+ iks_delete(payload_audio);
+ iks_delete(payload_cn);
+ iks_delete(dcodecs);
+ iks_delete(gtalk);
+ 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;
+ struct gtalk *client = p->parent;
+ int res = 0;
+
+ if (option_debug)
+ ast_log(LOG_DEBUG, "Answer!\n");
+ ast_mutex_lock(&p->lock);
+ gtalk_accept_call(client, p);
+ ast_mutex_unlock(&p->lock);
+ return res;
+}
+
+static struct ast_rtp *gtalk_get_rtp_peer(struct ast_channel *chan)
+{
+ struct gtalk_pvt *p = chan->tech_pvt;
+ struct ast_rtp *rtp = NULL;
+
+ if (!p)
+ return NULL;
+ ast_mutex_lock(&p->lock);
+ if (p->rtp)
+ rtp = p->rtp;
+ ast_mutex_unlock(&p->lock);
+ return rtp;
+}
+
+static int 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 *rtp, struct ast_rtp *vrtp, int 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 int gtalk_response(struct gtalk *client, 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", client->connection->jid->full);
+ 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);
+ }
+ }
+ iks_send(client->connection->p, response);
+ if (reason)
+ iks_delete(reason);
+ if (error)
+ iks_delete(error);
+ iks_delete(response);
+ res = 0;
+ }
+ return res;
+}
+
+static int gtalk_is_answered(struct gtalk *client, ikspak *pak)
+{
+ struct gtalk_pvt *tmp;
+
+ 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, GOOGLE_NODE, GOOGLE_SID, tmp->sid))
+ break;
+ }
+
+ 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, pak, NULL, NULL);
+ return 1;
+}
+
+static int gtalk_handle_dtmf(struct gtalk *client, ikspak *pak)
+{
+ struct gtalk_pvt *tmp;
+ iks *dtmfnode = NULL;
+ char *dtmf;
+ /* Make sure our new call doesn't exist yet */
+ for (tmp = client->p; tmp; tmp = tmp->next) {
+ if (iks_find_with_attrib(pak->x, GOOGLE_NODE, GOOGLE_SID, tmp->sid))
+ break;
+ }
+
+ if (tmp) {
+ if(iks_find_with_attrib(pak->x, "dtmf-method", "method", "rtp")) {
+ gtalk_response(client,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 = dtmf[0];
+ ast_queue_frame(tmp->owner, &f);
+ ast_verbose("GOOGLE! DTMF-relay event received: %c\n", f.subclass);
+ } else if(iks_find_with_attrib(pak->x, "dtmf", "action", "button-down")) {
+ struct ast_frame f = {AST_FRAME_DTMF_END, };
+ f.subclass = dtmf[0];
+ ast_queue_frame(tmp->owner, &f);
+ ast_verbose("GOOGLE! DTMF-relay event received: %c\n", f.subclass);
+ } else if(iks_find_attrib(pak->x, "dtmf")) { /* 250 millasecond default */
+ struct ast_frame f = {AST_FRAME_DTMF, };
+ f.subclass = dtmf[0];
+ ast_queue_frame(tmp->owner, &f);
+ ast_verbose("GOOGLE! DTMF-relay event received: %c\n", f.subclass);
+ }
+ }
+ }
+ gtalk_response(client, pak, NULL, NULL);
+ return 1;
+ } else
+ ast_log(LOG_NOTICE, "Whoa, didn't find call!\n");
+
+ gtalk_response(client, pak, NULL, NULL);
+ return 1;
+}
+
+
+static int gtalk_hangup_farend(struct gtalk *client, ikspak *pak)
+{
+ struct gtalk_pvt *tmp;
+
+ 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, GOOGLE_NODE, GOOGLE_SID, tmp->sid))
+ break;
+ }
+
+ if (tmp) {
+ tmp->alreadygone = 1;
+ ast_queue_hangup(tmp->owner);
+ } else
+ ast_log(LOG_NOTICE, "Whoa, didn't find call!\n");
+ gtalk_response(client, pak, NULL, NULL);
+ return 1;
+}
+
+static int gtalk_create_candidates(struct gtalk *client, struct gtalk_pvt *p, char *sid, char *from)
+{
+ struct gtalk_candidate *tmp;
+ struct aji_client *c = client->connection;
+ struct gtalk_candidate *ours1 = NULL, *ours2 = NULL;
+ struct sockaddr_in sin;
+ struct sockaddr_in dest;
+ struct in_addr us;
+ iks *iq, *gtalk, *candidate;
+ char user[17], pass[17], preference[5], port[7];
+
+
+ iq = iks_new("iq");
+ gtalk = iks_new(GOOGLE_NODE);
+ candidate = iks_new("candidate");
+ if (!iq || !gtalk || !candidate) {
+ 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_node(iq, gtalk);
+ iks_insert_node(gtalk, 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_get_us(p->rtp, &sin);
+ ast_find_ourip(&us, bindaddr);
+
+ /* 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", c->jid->full);
+ 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", "candidates");
+ iks_insert_attrib(gtalk, "id", sid);
+ iks_insert_attrib(gtalk, "initiator", (p->initiator) ? c->jid->full : from);
+ 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");
+ iks_send(c->p, iq);
+ }
+ p->laststun = 0;
+
+safeout:
+ if (ours1)
+ free(ours1);
+ if (ours2)
+ free(ours2);
+ if (iq)
+ iks_delete(iq);
+ if (gtalk)
+ iks_delete(gtalk);
+ if (candidate)
+ iks_delete(candidate);
+ return 1;
+}
+
+static struct gtalk_pvt *gtalk_alloc(struct gtalk *client, const char *from, const char *sid)
+{
+ struct gtalk_pvt *tmp = NULL;
+ struct aji_resource *resources = NULL;
+ struct aji_buddy *buddy;
+ char idroster[200];
+
+ if (option_debug)
+ ast_log(LOG_DEBUG, "The client is %s for alloc\n", client->name);
+ if (!sid && !strchr(from, '/')) { /* I started call! */
+ if (!strcasecmp(client->name, "guest")) {
+ buddy = ASTOBJ_CONTAINER_FIND(&client->connection->buddies, from);
+ if (buddy)
+ resources = buddy->resources;
+ } else
+ resources = client->buddy->resources;
+ while (resources) {
+ if (resources->cap->jingle) {
+ break;
+ }
+ resources = resources->next;
+ }
+ if (resources)
+ snprintf(idroster, sizeof(idroster), "%s/%s", from, 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;
+ }
+ if (sid) {
+ ast_copy_string(tmp->sid, sid, sizeof(tmp->sid));
+ ast_copy_string(tmp->from, from, sizeof(tmp->from));
+ } else {
+ snprintf(tmp->sid, sizeof(tmp->sid), "%08lx%08lx", ast_random(), ast_random());
+ ast_copy_string(tmp->from, idroster, sizeof(tmp->from));
+ tmp->initiator = 1;
+ }
+ tmp->rtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0, bindaddr.sin_addr);
+ tmp->parent = client;
+ if (!tmp->rtp) {
+ ast_log(LOG_WARNING, "Out of RTP sessions?\n");
+ free(tmp);
+ return NULL;
+ }
+ ast_copy_string(tmp->exten, "s", sizeof(tmp->exten));
+ ast_mutex_init(&tmp->lock);
+ ast_mutex_lock(>alklock);
+ tmp->next = client->p;
+ client->p = tmp;
+ ast_mutex_unlock(>alklock);
+ 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)
+{
+ struct ast_channel *tmp;
+ int fmt;
+ int what;
+
+ tmp = ast_channel_alloc(1);
+ if (!tmp) {
+ ast_log(LOG_WARNING, "Unable to allocate Gtalk channel structure!\n");
+ return NULL;
+ }
+ tmp->tech = >alk_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;
+ tmp->nativeformats = ast_codec_choose(&i->prefs, what, 1) | (i->jointcapability & AST_FORMAT_VIDEO_MASK);
+ fmt = ast_best_codec(tmp->nativeformats);
+
+ if (title)
+ ast_string_field_build(tmp, name, "Gtalk/%s-%04lx", title, ast_random() & 0xffff);
+ else
+ ast_string_field_build(tmp, name, "Gtalk/%s-%04lx", i->from, ast_random() & 0xffff);
+
+ if (i->rtp) {
+ tmp->fds[0] = ast_rtp_fd(i->rtp);
+ tmp->fds[1] = ast_rtcp_fd(i->rtp);
+ }
+ if (i->vrtp) {
+ tmp->fds[2] = ast_rtp_fd(i->vrtp);
+ tmp->fds[3] = ast_rtcp_fd(i->vrtp);
+ }
+ 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);
+ i->owner = tmp;
+ ast_mutex_lock(&usecnt_lock);
+ usecnt++;
+ ast_mutex_unlock(&usecnt_lock);
+ ast_copy_string(tmp->context, client->context, sizeof(tmp->context));
+ ast_copy_string(tmp->exten, i->exten, sizeof(tmp->exten));
+ ast_set_callerid(tmp, i->cid_num, i->cid_name, i->cid_num);
+ if (!ast_strlen_zero(i->exten) && strcmp(i->exten, "s"))
+ tmp->cid.cid_dnid = ast_strdup(i->exten);
+ tmp->priority = 1;
+ ast_setstate(tmp, state);
+ 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;
+ }
+
+ return tmp;
+}
+
+static int gtalk_action(struct gtalk *client, struct gtalk_pvt *p, const char *action)
+{
+ iks *request, *session = NULL;
+ int res = -1;
+
+ request = iks_new("iq");
+ if (request) {
+ iks_insert_attrib(request, "type", "set");
+ iks_insert_attrib(request, "from", client->connection->jid->full);
+ iks_insert_attrib(request, "to", p->from);
+ 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);
+ iks_insert_attrib(session, "initiator",
+ p->initiator ? client->connection->jid->full : p->from);
+ iks_insert_attrib(session, "xmlns", "http://www.google.com/session");
+ iks_insert_node(request, session);
+ iks_send(client->connection->p, request);
+ iks_delete(session);
+ res = 0;
+ }
+ 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;
+ 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_destroy(p->rtp);
+ if (p->vrtp)
+ ast_rtp_destroy(p->vrtp);
+ gtalk_free_candidates(p->theircandidates);
+ 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;
+
+ /* Make sure our new call doesn't exist yet */
+ while (tmp) {
+ if (iks_find_with_attrib(pak->x, GOOGLE_NODE, GOOGLE_SID, tmp->sid)) {
+ ast_log(LOG_NOTICE, "Ignoring duplicate call setup on SID %s\n", tmp->sid);
+ gtalk_response(client, pak, "out-of-order", NULL);
+ return -1;
+ }
+ tmp = tmp->next;
+ }
+
+ p = gtalk_alloc(client, pak->from->partial, iks_find_attrib(pak->query, GOOGLE_SID));
+ 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);
+ if (chan) {
+ ast_mutex_lock(&p->lock);
+ ast_copy_string(p->from, pak->from->full, sizeof(p->from));
+ if (iks_find_attrib(pak->query, GOOGLE_SID)) {
+ ast_copy_string(p->sid, iks_find_attrib(pak->query, GOOGLE_SID),
+ sizeof(p->sid));
+ }
+
+ codec = iks_child(iks_child(iks_child(pak->x)));
+ while (codec) {
+ ast_rtp_set_m_type(p->rtp, atoi(iks_find_attrib(codec, "id")));
+ ast_rtp_set_rtpmap_type(p->rtp, atoi(iks_find_attrib(codec, "id")), "audio",
+ iks_find_attrib(codec, "name"), 0);
+ codec = iks_next(codec);
+ }
+
+ ast_mutex_unlock(&p->lock);
+ ast_setstate(chan, AST_STATE_RING);
+ res = ast_pbx_start(chan);
+
+ switch (res) {
+ case AST_PBX_FAILED:
+ ast_log(LOG_WARNING, "Failed to start PBX :(\n");
+ gtalk_response(client, 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, pak, "service-unavailable", NULL);
+ break;
+ case AST_PBX_SUCCESS:
+ gtalk_response(client, pak, NULL, NULL);
+ gtalk_create_candidates(client, p,
+ iks_find_attrib(pak->query, GOOGLE_SID),
+ iks_find_attrib(pak->x, "from"));
+ /* nothing to do */
+ break;
+ }
+ } else {
+ gtalk_free_pvt(client, p);
+ }
+ 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;
+
+ if (time(NULL) == p->laststun)
+ return 0;
+
+ tmp = p->theircandidates;
+ p->laststun = time(NULL);
+ while (tmp) {
+ char username[256];
+ 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);
+
+ ast_rtp_stun_request(p->rtp, &sin, username);
+ 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;
+ newcandidate = ast_calloc(1, sizeof(*newcandidate));
+ if (!newcandidate)
+ return 0;
+ for (tmp = client->p; tmp; tmp = tmp->next) {
+ if (iks_find_with_attrib(pak->x, GOOGLE_NODE, GOOGLE_SID, tmp->sid)) {
+ p = tmp;
+ break;
+ }
+ }
+
+ if (!p)
+ return -1;
+
+ traversenodes = pak->query;
+ while(traversenodes) {
+ if(!strcasecmp(iks_name(traversenodes), "session")) {
+ traversenodes = iks_child(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;
+ }
+ traversenodes = iks_next(traversenodes);
+ }
+
+ receipt = iks_new("iq");
+ iks_insert_attrib(receipt, "type", "result");
+ iks_insert_attrib(receipt, "from", c->jid->full);
+ iks_insert_attrib(receipt, "to", iks_find_attrib(pak->x, "from"));
+ iks_insert_attrib(receipt, "id", iks_find_attrib(pak->x, "id"));
+ iks_send(c->p, 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_read(p->rtp);
+ gtalk_update_stun(p->parent, p);
+ if (p->owner) {
+ /* We already hold the channel lock */
+ if (f->frametype == AST_FRAME_VOICE) {
+ if (f->subclass != (p->owner->nativeformats & AST_FORMAT_AUDIO_MASK)) {
+ if (option_debug)
+ ast_log(LOG_DEBUG, "Oooh, format changed to %d\n", f->subclass);
+ p->owner->nativeformats =
+ (p->owner->nativeformats & AST_FORMAT_VIDEO_MASK) | f->subclass;
+ 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_log(LOG_DEBUG, "* 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;
+
+ switch (frame->frametype) {
+ case AST_FRAME_VOICE:
+ if (!(frame->subclass & ast->nativeformats)) {
+ ast_log(LOG_WARNING,
+ "Asked to transmit frame type %d, while native formats is %d (read/write = %d/%d)\n",
+ frame->subclass, ast->nativeformats, ast->readformat,
+ ast->writeformat);
+ return 0;
+ }
+ if (p) {
+ ast_mutex_lock(&p->lock);
+ if (p->rtp) {
+ res = ast_rtp_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_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_digit(struct ast_channel *ast, char digit)
+{
+ struct gtalk_pvt *p = ast->tech_pvt;
+ struct gtalk *client = p->parent;
+ iks *iq, *gtalk, *dtmf;
+ char buffer[2] = {digit, '\0'};
+ iq = iks_new("iq");
+ gtalk = iks_new("gtalk");
+ dtmf = iks_new("dtmf");
+ if(!iq || !gtalk || !dtmf) {
+ if(iq)
+ iks_delete(iq);
+ if(gtalk)
+ iks_delete(gtalk);
+ if(dtmf)
+ 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->from);
+ iks_insert_attrib(iq, "from", client->connection->jid->full);
+ 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", "content-info");
+ iks_insert_attrib(gtalk, "initiator", p->initiator ? client->connection->jid->full : p->from);
+ 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) {
+ ast_verbose("Sending 250ms dtmf!\n");
+ } else if (ast->dtmff.frametype == AST_FRAME_DTMF_BEGIN) {
+ iks_insert_attrib(dtmf, "action", "button-down");
+ } else if (ast->dtmff.frametype == AST_FRAME_DTMF_END) {
+ iks_insert_attrib(dtmf, "action", "button-up");
+ }
+ iks_send(client->connection->p, 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;
+}
+static int gtalk_transmit_invite(struct gtalk_pvt *p)
+{
+ struct gtalk *gtalk = NULL;
+ struct aji_client *client = NULL;
+ iks *iq, *desc, *session;
+ iks *payload_eg711u, *payload_pcmu;
+
+ gtalk = p->parent;
+ client = gtalk->connection;
+ iq = iks_new("iq");
+ desc = iks_new("description");
+ session = iks_new("session");
+ iks_insert_attrib(iq, "type", "set");
+ iks_insert_attrib(iq, "to", p->from);
+ iks_insert_attrib(iq, "from", client->jid->full);
+ iks_insert_attrib(iq, "id", client->mid);
[... 527 lines stripped ...]
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