adds support for t38 pass through patch brought up to trunk from

bug 5090 by josh colp.  Thanks to everyone who help get this patch through
especially to the author Steven Underwood.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@33890 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2 files changed, 873 insertions, 17 deletions

diff --git a/channel.c b/channel.c
index 398bf59ab..30a1ad073 100644
--- a/channel.c
+++ b/channel.c
@@ -2383,6 +2383,10 @@ int ast_write(struct ast_channel *chan, struct ast_frame *fr)
 		res = (chan->tech->write_video == NULL) ? 0 :
 			chan->tech->write_video(chan, fr);
 		break;
+	case AST_FRAME_MODEM:
+		res = (chan->tech->write == NULL) ? 0 :
+			chan->tech->write(chan, fr);
+		break;
 	case AST_FRAME_VOICE:
 		if (chan->tech->write == NULL)
 			break;	/*! \todo XXX should return 0 maybe ? */
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 5f600814b..49b8aac87 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -118,6 +118,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #include "asterisk/sched.h"
 #include "asterisk/io.h"
 #include "asterisk/rtp.h"
+#include "asterisk/udptl.h"
 #include "asterisk/acl.h"
 #include "asterisk/manager.h"
 #include "asterisk/callerid.h"
@@ -718,9 +719,13 @@ struct sip_auth {
 #define SIP_PAGE2_ALLOWOVERLAP		(1 << 11)	/*!< Allow overlap dialing ? */
 #define SIP_PAGE2_SUBSCRIBEMWIONLY	(1 << 12)	/*!< Only issue MWI notification if subscribed to */
 #define SIP_PAGE2_INC_RINGING		(1 << 13)	/*!< Did this connection increment the counter of in-use calls? */
+#define SIP_PAGE2_T38SUPPORT		(7 << 14)	/*!< T38 Fax Passthrough Support */
+#define SIP_PAGE2_T38SUPPORT_UDPTL	(1 << 14)	/*!< 14: T38 Fax Passthrough Support */
+#define SIP_PAGE2_T38SUPPORT_RTP	(2 << 14)	/*!< 15: T38 Fax Passthrough Support */
+#define SIP_PAGE2_T38SUPPORT_TCP	(4 << 14)	/*!< 16: T38 Fax Passthrough Support */
 
 #define SIP_PAGE2_FLAGS_TO_COPY \
-	(SIP_PAGE2_ALLOWSUBSCRIBE | SIP_PAGE2_ALLOWOVERLAP | SIP_PAGE2_VIDEOSUPPORT)
+	(SIP_PAGE2_ALLOWSUBSCRIBE | SIP_PAGE2_ALLOWOVERLAP | SIP_PAGE2_VIDEOSUPPORT | SIP_PAGE2_T38SUPPORT)
 
 /* SIP packet flags */
 #define SIP_PKT_DEBUG		(1 << 0)	/*!< Debug this packet */
@@ -729,10 +734,55 @@ struct sip_auth {
 #define SIP_PKT_IGNORE_RESP	(1 << 3)	/*!< Resp ignore - ??? */
 #define SIP_PKT_IGNORE_REQ	(1 << 4)	/*!< Req ignore - ??? */
 
+/* T.38 set of flags */
+#define T38FAX_FILL_BIT_REMOVAL		(1 << 0)	/*!< Default: 0 (unset)*/
+#define T38FAX_TRANSCODING_MMR			(1 << 1)	/*!< Default: 0 (unset)*/
+#define T38FAX_TRANSCODING_JBIG		(1 << 2)	/*!< Default: 0 (unset)*/
+/* Rate management */
+#define T38FAX_RATE_MANAGEMENT_TRANSFERED_TCF	(0 << 3)
+#define T38FAX_RATE_MANAGEMENT_LOCAL_TCF	(1 << 3)	/*!< Unset for transferredTCF (UDPTL), set for localTCF (TPKT) */
+/* UDP Error correction */
+#define T38FAX_UDP_EC_NONE			(0 << 4)	/*!< two bits, if unset NO t38UDPEC field in T38 SDP*/
+#define T38FAX_UDP_EC_FEC			(1 << 4)	/*!< Set for t38UDPFEC */
+#define T38FAX_UDP_EC_REDUNDANCY		(2 << 4)	/*!< Set for t38UDPRedundancy */
+/* T38 Spec version */
+#define T38FAX_VERSION				(3 << 6)	/*!< two bits, 2 values so far, up to 4 values max */
+#define T38FAX_VERSION_0			(0 << 6)	/*!< Version 0 */
+#define T38FAX_VERSION_1			(1 << 6)	/*!< Version 1 */
+/* Maximum Fax Rate */
+#define T38FAX_RATE_2400			(1 << 8)	/*!< 2400 bps t38FaxRate */
+#define T38FAX_RATE_4800			(1 << 9)	/*!< 4800 bps t38FaxRate */
+#define T38FAX_RATE_7200			(1 << 10)	/*!< 7200 bps t38FaxRate */
+#define T38FAX_RATE_9600			(1 << 11)	/*!< 9600 bps t38FaxRate */
+#define T38FAX_RATE_12000			(1 << 12)	/*!< 12000 bps t38FaxRate */
+#define T38FAX_RATE_14400			(1 << 13)	/*!< 14400 bps t38FaxRate */
+
+/*!< This is default: NO MMR and JBIG trancoding, NO fill bit removal, transferredTCF TCF, UDP FEC, Version 0 and 9600 max fax rate */
+static int global_t38_capability = T38FAX_VERSION_0 | T38FAX_RATE_2400 | T38FAX_RATE_4800 | T38FAX_RATE_7200 | T38FAX_RATE_9600;
+
 #define sipdebug		ast_test_flag(&global_flags[1], SIP_PAGE2_DEBUG)
 #define sipdebug_config		ast_test_flag(&global_flags[1], SIP_PAGE2_DEBUG_CONFIG)
 #define sipdebug_console	ast_test_flag(&global_flags[1], SIP_PAGE2_DEBUG_CONSOLE)
 
+/*! \brief T38 Sates for a call */
+enum t38state {
+	T38_DISABLED = 0,		/*! Not enabled */
+	T38_LOCAL_DIRECT,		/*! Offered from local */
+	T38_LOCAL_REINVITE,		/*! Offered from local - REINVITE */
+	T38_PEER_DIRECT,		/*! Offered from peer */
+	T38_PEER_REINVITE,		/*! Offered from peer - REINVITE */
+	T38_ENABLED			/*! Negotiated (enabled) */
+};
+
+/*! \brief T.38 channel settings (at some point we need to make this alloc'ed */
+struct t38properties {
+	struct ast_flags t38support;	/*!< Flag for udptl, rtp or tcp support for this session */
+	int capability;			/*!< Our T38 capability */
+	int peercapability;		/*!< Peers T38 capability */
+	int jointcapability;		/*!< Supported T38 capability at both ends */
+	enum t38state state;		/*!< T.38 state */
+};
+
 /*! \brief Parameters to know status of transfer */
 enum referstatus {
 	REFER_IDLE,		/*!< No REFER is in progress */
@@ -837,6 +887,9 @@ static struct sip_pvt {
 	int noncodeccapability;			/*!< DTMF RFC2833 telephony-event */
 	int redircodecs;			/*!< Redirect codecs */
 	int maxcallbitrate;			/*!< Maximum Call Bitrate for Video Calls */	
+	struct t38properties t38;		/*!< T38 settings */
+	struct sockaddr_in udptlredirip;	/*!< Where our T.38 UDPTL should be going if not to us */
+	struct ast_udptl *udptl;		/*!< T.38 UDPTL session */
 	int callingpres;			/*!< Calling presentation */
 	int authtries;				/*!< Times we've tried to authenticate */
 	int expiry;				/*!< How long we take to expire */
@@ -1409,8 +1462,14 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struc
 static struct ast_rtp *sip_get_rtp_peer(struct ast_channel *chan);
 static struct ast_rtp *sip_get_vrtp_peer(struct ast_channel *chan);
 static int sip_get_codec(struct ast_channel *chan);
-static struct ast_frame *sip_rtp_read(struct ast_channel *ast, struct sip_pvt *p);
+static struct ast_frame *sip_rtp_read(struct ast_channel *ast, struct sip_pvt *p, int *faxdetect);
 
+/*------ T38 Support --------- */
+static int sip_handle_t38_reinvite(struct ast_channel *chan, struct sip_pvt *pvt, int reinvite); /*!< T38 negotiation helper function */
+static int transmit_response_with_t38_sdp(struct sip_pvt *p, char *msg, struct sip_request *req, int retrans);
+static int transmit_reinvite_with_t38_sdp(struct sip_pvt *p);
+static struct ast_udptl *sip_get_udptl_peer(struct ast_channel *chan);
+static int sip_set_udptl_peer(struct ast_channel *chan, struct ast_udptl *udptl);
 
 /*! \brief Definition of this channel for PBX channel registration */
 static const struct ast_channel_tech sip_tech = {
@@ -1452,6 +1511,13 @@ static struct ast_rtp_protocol sip_rtp = {
 	get_codec: sip_get_codec,
 };
 
+/*! \brief Interface structure with callbacks used to connect to UDPTL module*/
+static struct ast_udptl_protocol sip_udptl = {
+	type: "SIP",
+	get_udptl_info: sip_get_udptl_peer,
+	set_udptl_peer: sip_set_udptl_peer,
+};
+
 /*! \brief Convert transfer status to string */
 static char *referstatus2str(enum referstatus rstatus)
 {
@@ -2387,6 +2453,24 @@ static int create_addr_from_peer(struct sip_pvt *r, struct sip_peer *peer)
 		r->vrtp = NULL;
 	}
 	r->prefs = peer->prefs;
+	if (ast_test_flag(&r->flags[1], SIP_PAGE2_T38SUPPORT)) {
+		r->t38.capability = global_t38_capability;
+		if (r->udptl) {
+			if (ast_udptl_get_error_correction_scheme(r->udptl) == UDPTL_ERROR_CORRECTION_FEC )
+				r->t38.capability |= T38FAX_UDP_EC_FEC;
+			else if (ast_udptl_get_error_correction_scheme(r->udptl) == UDPTL_ERROR_CORRECTION_REDUNDANCY )
+				r->t38.capability |= T38FAX_UDP_EC_REDUNDANCY;
+			else if (ast_udptl_get_error_correction_scheme(r->udptl) == UDPTL_ERROR_CORRECTION_NONE )
+				r->t38.capability |= T38FAX_UDP_EC_NONE;
+			r->t38.capability |= T38FAX_RATE_MANAGEMENT_TRANSFERED_TCF;
+			if (option_debug > 1)
+				ast_log(LOG_DEBUG,"Our T38 capability (%d)\n", r->t38.capability);
+		}
+		r->t38.jointcapability = r->t38.capability;
+	} else if (r->udptl) {
+		ast_udptl_destroy(r->udptl);
+		r->udptl = NULL;
+	}
 	natflags = ast_test_flag(&r->flags[0], SIP_NAT) & SIP_NAT_ROUTE;
 	if (r->rtp) {
 		if (option_debug)
@@ -2400,6 +2484,11 @@ static int create_addr_from_peer(struct sip_pvt *r, struct sip_peer *peer)
 		ast_rtp_setnat(r->vrtp, natflags);
 		ast_rtp_setdtmf(r->vrtp, 0);
 	}
+	if (r->udptl) {
+		if (option_debug)
+			ast_log(LOG_DEBUG, "Setting NAT on UDPTL to %s\n", natflags ? "On" : "Off");
+		ast_udptl_setnat(r->udptl, natflags);
+	}
 	ast_string_field_set(r, peername, peer->username);
 	ast_string_field_set(r, authname, peer->username);
 	ast_string_field_set(r, username, peer->username);
@@ -2563,7 +2652,12 @@ static int sip_call(struct ast_channel *ast, char *dest, int timeout)
 		} else if (!strcasecmp(ast_var_name(current),"SIPTRANSFER_REPLACES")) {
 			/* We're replacing a call. */
 			p->options->replaces = ast_var_value(current);
+		} else if (!strcasecmp(ast_var_name(current),"T38CALL")) {
+			p->t38.state = T38_LOCAL_DIRECT;
+			if (option_debug)
+				ast_log(LOG_DEBUG,"T38State change to %d on channel %s\n", p->t38.state, ast->name);
 		}
+
 	}
 	
 	res = 0;
@@ -2588,6 +2682,9 @@ static int sip_call(struct ast_channel *ast, char *dest, int timeout)
 	if ( res != -1 ) {
 		p->callingpres = ast->cid.cid_pres;
 		p->jointcapability = p->capability;
+		p->t38.jointcapability = p->t38.capability;
+		if (option_debug)
+			ast_log(LOG_DEBUG,"Our T38 capability (%d), joint T38 capability (%d)\n", p->t38.capability, p->t38.jointcapability);
 		transmit_invite(p, SIP_INVITE, 1, 2);
 		if (p->maxtime) {
 			/* Initialize auto-congest time */
@@ -2655,6 +2752,8 @@ static void __sip_destroy(struct sip_pvt *p, int lockowner)
 		ast_rtp_destroy(p->rtp);
 	if (p->vrtp)
 		ast_rtp_destroy(p->vrtp);
+	if (p->udptl)
+		ast_udptl_destroy(p->udptl);
 	if (p->refer)
 		free(p->refer);
 	if (p->route) {
@@ -3182,7 +3281,14 @@ static int sip_answer(struct ast_channel *ast)
 		ast_setstate(ast, AST_STATE_UP);
 		if (option_debug)
 			ast_log(LOG_DEBUG, "SIP answering channel: %s\n", ast->name);
-		res = transmit_response_with_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL);
+		if (p->t38.state == T38_PEER_DIRECT) {
+			p->t38.state = T38_ENABLED;
+			if (option_debug > 1)
+				ast_log(LOG_DEBUG,"T38State change to %d on channel %s\n", p->t38.state, ast->name);
+			res = transmit_response_with_t38_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL);
+		} else {
+			res = transmit_response_with_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL);
+		}
 	}
 	ast_mutex_unlock(&p->lock);
 	return res;
@@ -3237,6 +3343,21 @@ static int sip_write(struct ast_channel *ast, struct ast_frame *frame)
 	case AST_FRAME_IMAGE:
 		return 0;
 		break;
+	case AST_FRAME_MODEM:
+		if (p) {
+			ast_mutex_lock(&p->lock);
+			if (p->udptl) {
+				if ((ast->_state != AST_STATE_UP) &&
+					!ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) && 
+				    !ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
+					transmit_response_with_t38_sdp(p, "183 Session Progress", &p->initreq, XMIT_RELIABLE);
+					ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
+				}
+				res = ast_udptl_write(p->udptl, frame);
+			}
+			ast_mutex_unlock(&p->lock);
+		}
+		break;
 	default: 
 		ast_log(LOG_WARNING, "Can't send %d type frames with SIP write\n", frame->frametype);
 		return 0;
@@ -3504,6 +3625,9 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit
 		tmp->fds[2] = ast_rtp_fd(i->vrtp);
 		tmp->fds[3] = ast_rtcp_fd(i->vrtp);
 	}
+	if (i->udptl) {
+		tmp->fds[5] = ast_udptl_fd(i->udptl);
+	}
 	if (state == AST_STATE_RING)
 		tmp->rings = 1;
 	tmp->adsicpe = AST_ADSI_UNAVAILABLE;
@@ -3697,7 +3821,7 @@ static const char *get_header(const struct sip_request *req, const char *name)
 }
 
 /*! \brief Read RTP from network */
-static struct ast_frame *sip_rtp_read(struct ast_channel *ast, struct sip_pvt *p)
+static struct ast_frame *sip_rtp_read(struct ast_channel *ast, struct sip_pvt *p, int *faxdetect)
 {
 	/* Retrieve audio/etc from channel.  Assumes p->lock is already held. */
 	struct ast_frame *f;
@@ -3720,6 +3844,9 @@ static struct ast_frame *sip_rtp_read(struct ast_channel *ast, struct sip_pvt *p
 	case 3:
 		f = ast_rtcp_read(p->vrtp);	/* RTCP Control Channel for video */
 		break;
+	case 5:
+		f = ast_udptl_read(p->udptl);	/* UDPTL for T.38 */
+		break;
 	default:
 		f = &ast_null_frame;
 	}
@@ -3740,8 +3867,15 @@ static struct ast_frame *sip_rtp_read(struct ast_channel *ast, struct sip_pvt *p
 			}
 			if ((ast_test_flag(&p->flags[0], 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);
+				if (f && f->frametype == AST_FRAME_DTMF) {
+					if (ast_test_flag(&p->t38.t38support, SIP_PAGE2_T38SUPPORT_UDPTL) && f->subclass == 'f') {
+						if (option_debug)
+							ast_log(LOG_DEBUG, "Fax CNG detected on %s\n", ast->name);
+						*faxdetect = 1;
+					} else if (option_debug) {
+						ast_log(LOG_DEBUG, "* Detected inband DTMF '%c'\n", f->subclass);
+					}
+				}
 			}
 		}
 	}
@@ -3753,10 +3887,31 @@ static struct ast_frame *sip_read(struct ast_channel *ast)
 {
 	struct ast_frame *fr;
 	struct sip_pvt *p = ast->tech_pvt;
+	int faxdetected = 0;
 
 	ast_mutex_lock(&p->lock);
-	fr = sip_rtp_read(ast, p);
+	fr = sip_rtp_read(ast, p, &faxdetected);
 	p->lastrtprx = time(NULL);
+
+	/* If we are NOT bridged to another channel, and we have detected fax tone we issue T38 re-invite to a peer */
+	/* If we are bridged then it is the responsibility of the SIP device to issue T38 re-invite if it detects CNG or fax preamble */
+	if (faxdetected && ast_test_flag(&p->t38.t38support, SIP_PAGE2_T38SUPPORT_UDPTL) && (p->t38.state == T38_DISABLED) && !(ast_bridged_channel(ast))) {
+		if (!ast_test_flag(&p->flags[0], SIP_GOTREFER)) {
+			if (!p->pendinginvite) {
+				if (option_debug > 2)
+					ast_log(LOG_DEBUG, "Sending reinvite on SIP (%s) for T.38 negotiation.\n",ast->name);
+				p->t38.state = T38_LOCAL_REINVITE;
+				transmit_reinvite_with_t38_sdp(p);
+				if (option_debug > 1)
+					ast_log(LOG_DEBUG, "T38 state changed to %d on channel %s", p->t38.state, ast->name);
+			}
+		} else if (!ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
+			if (option_debug > 2)
+				ast_log(LOG_DEBUG, "Deferring reinvite on SIP (%s) - it will be re-negotiated for T.38\n", ast->name);
+			ast_set_flag(&p->flags[0], SIP_NEEDREINVITE);
+		}
+	}
+
 	ast_mutex_unlock(&p->lock);
 	return fr;
 }
@@ -3850,6 +4005,8 @@ static struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *si
 		/* If the global videosupport flag is on, we always create a RTP interface for video */
 		if (ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT))
 			p->vrtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0, bindaddr.sin_addr);
+		if (ast_test_flag(&p->flags[1], SIP_PAGE2_T38SUPPORT))
+			p->udptl = ast_udptl_new_with_bindaddr(sched, io, 0, bindaddr.sin_addr);
 		if (!p->rtp || (ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) && !p->vrtp)) {
 			ast_log(LOG_WARNING, "Unable to create RTP audio %s session: %s\n",
 				ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) ? "and video" : "", strerror(errno));
@@ -3867,6 +4024,9 @@ static struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *si
 			ast_rtp_settos(p->vrtp, global_tos_video);
 			ast_rtp_setdtmf(p->vrtp, 0);
 		}
+		if (p->udptl) {
+			ast_udptl_settos(p->udptl, global_tos_audio);
+		}
 		p->rtptimeout = global_rtptimeout;
 		p->rtpholdtimeout = global_rtpholdtimeout;
 		p->rtpkeepalive = global_rtpkeepalive;
@@ -3883,6 +4043,8 @@ static struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *si
 			ast_rtp_setnat(p->rtp, natflags);
 		if (p->vrtp)
 			ast_rtp_setnat(p->vrtp, natflags);
+		if (p->udptl)
+			ast_udptl_setnat(p->udptl, natflags);
 	}
 
 	if (p->method != SIP_REGISTER)
@@ -3899,6 +4061,17 @@ static struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *si
 	if ((ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833) ||
 	    (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO))
 		p->noncodeccapability |= AST_RTP_DTMF;
+	if (p->udptl) {
+		p->t38.capability = global_t38_capability;
+		if (ast_udptl_get_error_correction_scheme(p->udptl) == UDPTL_ERROR_CORRECTION_REDUNDANCY)
+			p->t38.capability |= T38FAX_UDP_EC_REDUNDANCY;
+		else if (ast_udptl_get_error_correction_scheme(p->udptl) == UDPTL_ERROR_CORRECTION_FEC)
+			p->t38.capability |= T38FAX_UDP_EC_FEC;
+		else if (ast_udptl_get_error_correction_scheme(p->udptl) == UDPTL_ERROR_CORRECTION_NONE)
+			p->t38.capability |= T38FAX_UDP_EC_NONE;
+		p->t38.capability |= T38FAX_RATE_MANAGEMENT_TRANSFERED_TCF;
+		p->t38.jointcapability = p->t38.capability;
+	}
 	ast_string_field_set(p, context, default_context);
 
 	/* Add to active dialog list */
@@ -4256,6 +4429,10 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
 	int len = -1;
 	int portno = -1;		/*!< RTP Audio port number */
 	int vportno = -1;		/*!< RTP Video port number */
+	int udptlportno = -1;
+	int peert38capability = 0;
+	char s[256];
+	int old = 0;
 
 	/* Peer capability is the capability in the SDP, non codec is RFC2833 DTMF (101) */	
 	int peercapability, peernoncodeccapability;
@@ -4366,6 +4543,20 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
 					ast_verbose("Found RTP video format %d\n", codec);
 				ast_rtp_set_m_type(newvideortp, codec);
 			}
+		} else if (p->udptl && (sscanf(m, "image %d udptl t38 %n", &x, &len) == 1)) {
+			if (debug)
+				ast_verbose("Got T.38 offer in SDP\n");
+			udptlportno = x;
+			
+			if (p->owner && p->lastinvite) {
+				p->t38.state = T38_PEER_REINVITE; /* T38 Offered in re-invite from remote party */
+				if (option_debug > 1)
+					ast_log(LOG_DEBUG, "T38 state changed to %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>" );
+			} else {
+				p->t38.state = T38_PEER_DIRECT; /* T38 Offered directly from peer in first invite */
+				if (option_debug > 1)
+					ast_log(LOG_DEBUG, "T38 state changed to %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>");
+			}
 		} else 
 			ast_log(LOG_WARNING, "Unsupported SDP media type in offer: %s\n", m);
 		if (numberofports > 1)
@@ -4388,7 +4579,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
 
 		}
 	}
-	if (portno == -1 && vportno == -1)
+	if (portno == -1 && vportno == -1 && udptlportno == -1)
 		/* No acceptable offer found in SDP  - we have no ports */
 		/* Do not change RTP or VRTP if this is a re-invite */
 		return -2;
@@ -4404,13 +4595,36 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
 	if (vhp)
 		memcpy(&vsin.sin_addr, vhp->h_addr, sizeof(vsin.sin_addr));
 		
-
-	/* Setup audio port number */
-	sin.sin_port = htons(portno);
+	if (p->rtp) {
+		if (portno > 0) {
+			sin.sin_port = htons(portno);
+			ast_rtp_set_peer(p->rtp, &sin);
+			if (debug)
+				ast_verbose("Peer audio RTP is at port %s:%d\n", ast_inet_ntoa(iabuf,sizeof(iabuf), sin.sin_addr), ntohs(sin.sin_port));
+		} else {
+			ast_rtp_stop(p->rtp);
+			if (debug)
+				ast_verbose("Peer doesn't provide audio\n");
+		}
+	}
 	/* Setup video port number */
 	if (vportno != -1)
 		vsin.sin_port = htons(vportno);
 
+	/* Setup UDPTL port number */
+	if (p->udptl) {
+		if (udptlportno > 0) {
+			sin.sin_port = htons(udptlportno);
+			ast_udptl_set_peer(p->udptl, &sin);
+			if (debug)
+				ast_log(LOG_DEBUG,"Peer T.38 UDPTL is at port %s:%d\n",ast_inet_ntoa(iabuf, sizeof(iabuf), sin.sin_addr), ntohs(sin.sin_port));
+		} else {
+			ast_udptl_stop(p->udptl);
+			if (debug)
+				ast_log(LOG_DEBUG, "Peer doesn't provide T.38 UDPTL\n");
+		}
+	}
+
 	/* Next, scan through each "a=rtpmap:" line, noting each
 	 * specified RTP payload type (with corresponding MIME subtype):
 	 */
@@ -4474,6 +4688,123 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
 		if (p->vrtp)
 			ast_rtp_set_rtpmap_type(newvideortp, codec, "video", mimeSubtype);
 	}
+	
+	if (udptlportno != -1) {
+		int found = 0, x;
+		
+		old = 0;
+		
+		/* Scan trough the a= lines for T38 attributes and set apropriate fileds */
+		iterator = req->sdp_start;
+		while ((a = get_sdp_iterate(&iterator, req, "a"))[0] != '\0') {
+			if ((sscanf(a, "T38FaxMaxBuffer:%d", &x) == 1)) {
+				found = 1;
+				if (option_debug > 2)
+					ast_log(LOG_DEBUG,"MaxBufferSize:%d\n",x);
+			}
+			if ((sscanf(a, "T38MaxBitRate:%d", &x) == 1)) {
+				found = 1;
+				if (option_debug > 2)
+					ast_log(LOG_DEBUG,"T38MaxBitRate: %d\n",x);
+				switch (x) {
+				case 14400:
+					peert38capability |= T38FAX_RATE_14400 | T38FAX_RATE_12000 | T38FAX_RATE_9600 | T38FAX_RATE_7200 | T38FAX_RATE_4800 | T38FAX_RATE_2400;
+					break;
+				case 12000:
+					peert38capability |= T38FAX_RATE_12000 | T38FAX_RATE_9600 | T38FAX_RATE_7200 | T38FAX_RATE_4800 | T38FAX_RATE_2400;
+					break;
+				case 9600:
+					peert38capability |= T38FAX_RATE_9600 | T38FAX_RATE_7200 | T38FAX_RATE_4800 | T38FAX_RATE_2400;
+					break;
+				case 7200:
+					peert38capability |= T38FAX_RATE_7200 | T38FAX_RATE_4800 | T38FAX_RATE_2400;
+					break;
+				case 4800:
+					peert38capability |= T38FAX_RATE_4800 | T38FAX_RATE_2400;
+					break;
+				case 2400:
+					peert38capability |= T38FAX_RATE_2400;
+					break;
+				}
+			}
+			if ((sscanf(a, "T38FaxVersion:%d", &x) == 1)) {
+				found = 1;
+				if (option_debug > 2)
+					ast_log(LOG_DEBUG,"FaxVersion: %d\n",x);
+				if (x == 0)
+					peert38capability |= T38FAX_VERSION_0;
+				else if (x == 1)
+					peert38capability |= T38FAX_VERSION_1;
+			}
+			if ((sscanf(a, "T38FaxMaxDatagram:%d", &x) == 1)) {
+				found = 1;
+				if (option_debug > 2)
+					ast_log(LOG_DEBUG,"FaxMaxDatagram: %d\n",x);
+				ast_udptl_set_far_max_datagram(p->udptl, x);
+				ast_udptl_set_local_max_datagram(p->udptl, x);
+			}
+			if ((sscanf(a, "T38FaxFillBitRemoval:%d", &x) == 1)) {
+				found = 1;
+				if (option_debug > 2)
+					ast_log(LOG_DEBUG,"FillBitRemoval: %d\n",x);
+				if (x == 1)
+					peert38capability |= T38FAX_FILL_BIT_REMOVAL;
+			}
+			if ((sscanf(a, "T38FaxTranscodingMMR:%d", &x) == 1)) {
+				found = 1;
+				if (option_debug > 2)
+					ast_log(LOG_DEBUG,"Transcoding MMR: %d\n",x);
+				if (x == 1)
+					peert38capability |= T38FAX_TRANSCODING_MMR;
+			}
+			if ((sscanf(a, "T38FaxTranscodingJBIG:%d", &x) == 1)) {
+				found = 1;
+				if (option_debug > 2)
+					ast_log(LOG_DEBUG,"Transcoding JBIG: %d\n",x);
+				if (x == 1)
+					peert38capability |= T38FAX_TRANSCODING_JBIG;
+			}
+			if ((sscanf(a, "T38FaxRateManagement:%s", s) == 1)) {
+				found = 1;
+				if (option_debug > 2)
+					ast_log(LOG_DEBUG,"RateMangement: %s\n", s);
+				if (!strcasecmp(s, "localTCF"))
+					peert38capability |= T38FAX_RATE_MANAGEMENT_LOCAL_TCF;
+				else if (!strcasecmp(s, "transferredTCF"))
+					peert38capability |= T38FAX_RATE_MANAGEMENT_TRANSFERED_TCF;
+			}
+			if ((sscanf(a, "T38FaxUdpEC:%s", s) == 1)) {
+				found = 1;
+				if (option_debug > 2)
+					ast_log(LOG_DEBUG,"UDP EC: %s\n", s);
+				if (!strcasecmp(s, "t38UDPRedundancy")) {
+					peert38capability |= T38FAX_UDP_EC_REDUNDANCY;
+					ast_udptl_set_error_correction_scheme(p->udptl, UDPTL_ERROR_CORRECTION_REDUNDANCY);
+				} else if (!strcasecmp(s, "t38UDPFEC")) {
+					peert38capability |= T38FAX_UDP_EC_FEC;
+					ast_udptl_set_error_correction_scheme(p->udptl, UDPTL_ERROR_CORRECTION_FEC);
+				} else {
+					peert38capability |= T38FAX_UDP_EC_NONE;
+					ast_udptl_set_error_correction_scheme(p->udptl, UDPTL_ERROR_CORRECTION_NONE);
+				}
+			}
+		}
+		if (found) { /* Some cisco equipment returns nothing beside c= and m= lines in 200 OK T38 SDP */
+			p->t38.peercapability = peert38capability;
+			p->t38.jointcapability = (peert38capability & 255); /* Put everything beside supported speeds settings */
+			peert38capability &= (T38FAX_RATE_14400 | T38FAX_RATE_12000 | T38FAX_RATE_9600 | T38FAX_RATE_7200 | T38FAX_RATE_4800 | T38FAX_RATE_2400);
+			p->t38.jointcapability |= (peert38capability & p->t38.capability); /* Put the lower of our's and peer's speed */
+		}
+		if (debug)
+			ast_log(LOG_DEBUG,"Our T38 capability = (%d), peer T38 capability (%d), joint T38 capability (%d)\n",
+				p->t38.capability,
+				p->t38.peercapability,
+				p->t38.jointcapability);
+	} else {
+		p->t38.state = T38_DISABLED;
+		if (option_debug > 1)
+			ast_log(LOG_DEBUG, "T38 state changed to %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>");
+	}
 
 	/* Now gather all of the codecs that we are asked for: */
 	ast_rtp_get_current_formats(newaudiortp, &peercapability, &peernoncodeccapability);
@@ -5209,6 +5540,142 @@ static void add_codec_to_sdp(const struct sip_pvt *p, int codec, int sample_rate
 		ast_build_string(a_buf, a_size, "a=fmtp:%d annexb=no\r\n", rtp_code);
 }
 
+/*! \brief Get Max T.38 Transmision rate from T38 capabilities */
+static int t38_get_rate(int t38cap)
+{
+	int maxrate = (t38cap & (T38FAX_RATE_14400 | T38FAX_RATE_12000 | T38FAX_RATE_9600 | T38FAX_RATE_7200 | T38FAX_RATE_4800 | T38FAX_RATE_2400));
+	
+	if (maxrate & T38FAX_RATE_14400) {
+		if (option_debug > 1)
+			ast_log(LOG_DEBUG, "T38MaxFaxRate 14400 found\n");
+		return 14400;
+	} else if (maxrate & T38FAX_RATE_12000) {
+		if (option_debug > 1)
+			ast_log(LOG_DEBUG, "T38MaxFaxRate 12000 found\n");
+		return 12000;
+	} else if (maxrate & T38FAX_RATE_9600) {
+		if (option_debug > 1)
+			ast_log(LOG_DEBUG, "T38MaxFaxRate 9600 found\n");
+		return 9600;
+	} else if (maxrate & T38FAX_RATE_7200) {
+		if (option_debug > 1)
+			ast_log(LOG_DEBUG, "T38MaxFaxRate 7200 found\n");
+		return 7200;
+	} else if (maxrate & T38FAX_RATE_4800) {
+		if (option_debug > 1)
+			ast_log(LOG_DEBUG, "T38MaxFaxRate 4800 found\n");
+		return 4800;
+	} else if (maxrate & T38FAX_RATE_2400) {
+		if (option_debug > 1)
+			ast_log(LOG_DEBUG, "T38MaxFaxRate 2400 found\n");
+		return 2400;
+	} else {
+		if (option_debug > 1)
+			ast_log(LOG_DEBUG, "Strange, T38MaxFaxRate NOT found in peers T38 SDP.\n");
+		return 0;
+	}
+}
+
+/*! \brief Add T.38 Session Description Protocol message */
+static int add_t38_sdp(struct sip_request *resp, struct sip_pvt *p)
+{
+	int len = 0;
+	int x = 0;
+	struct sockaddr_in udptlsin;
+	char v[256] = "";
+	char s[256] = "";
+	char o[256] = "";
+	char c[256] = "";
+	char t[256] = "";
+	char m_modem[256];
+	char a_modem[1024];
+	char *m_modem_next = m_modem;
+	size_t m_modem_left = sizeof(m_modem);
+	char *a_modem_next = a_modem;
+	size_t a_modem_left = sizeof(a_modem);
+	char iabuf[INET_ADDRSTRLEN];
+	struct sockaddr_in udptldest = { 0, };
+	int debug;
+	
+	debug = sip_debug_test_pvt(p);
+	len = 0;
+	if (!p->udptl) {
+		ast_log(LOG_WARNING, "No way to add SDP without an UDPTL structure\n");
+		return -1;
+	}
+	
+	if (!p->sessionid) {
+		p->sessionid = getpid();
+		p->sessionversion = p->sessionid;
+	} else
+		p->sessionversion++;
+	
+	/* Our T.38 end is */
+	ast_udptl_get_us(p->udptl, &udptlsin);
+	
+	/* Determine T.38 UDPTL destination */
+	if (p->udptlredirip.sin_addr.s_addr) {
+		udptldest.sin_port = p->udptlredirip.sin_port;
+		udptldest.sin_addr = p->udptlredirip.sin_addr;
+	} else {
+		udptldest.sin_addr = p->ourip;
+		udptldest.sin_port = udptlsin.sin_port;
+	}
+	
+	if (debug) {
+		ast_log(LOG_DEBUG, "T.38 UDPTL is at %s port %d\n", ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip), ntohs(udptlsin.sin_port));
+	}
+	
+	/* We break with the "recommendation" and send our IP, in order that our
+	   peer doesn't have to ast_gethostbyname() us */
+	
+	if (debug) {
+		ast_log(LOG_DEBUG, "Our T38 capability (%d), peer T38 capability (%d), joint capability (%d)\n",
+			p->t38.capability,
+			p->t38.peercapability,
+			p->t38.jointcapability);
+	}
+	snprintf(v, sizeof(v), "v=0\r\n");
+	snprintf(o, sizeof(o), "o=root %d %d IN IP4 %s\r\n", p->sessionid, p->sessionversion, ast_inet_ntoa(iabuf, sizeof(iabuf), udptldest.sin_addr));
+	snprintf(s, sizeof(s), "s=session\r\n");
+	snprintf(c, sizeof(c), "c=IN IP4 %s\r\n", ast_inet_ntoa(iabuf, sizeof(iabuf), udptldest.sin_addr));
+	snprintf(t, sizeof(t), "t=0 0\r\n");
+	ast_build_string(&m_modem_next, &m_modem_left, "m=image %d udptl t38\r\n", ntohs(udptldest.sin_port));
+	
+	if ((p->t38.jointcapability & T38FAX_VERSION) == T38FAX_VERSION_0)
+		ast_build_string(&a_modem_next, &a_modem_left, "a=T38FaxVersion:0\r\n");
+	if ((p->t38.jointcapability & T38FAX_VERSION) == T38FAX_VERSION_1)
+		ast_build_string(&a_modem_next, &a_modem_left, "a=T38FaxVersion:1\r\n");
+	if ((x = t38_get_rate(p->t38.jointcapability)))
+		ast_build_string(&a_modem_next, &a_modem_left, "a=T38MaxBitRate:%d\r\n",x);
+	ast_build_string(&a_modem_next, &a_modem_left, "a=T38FaxFillBitRemoval:%d\r\n", (p->t38.jointcapability & T38FAX_FILL_BIT_REMOVAL) ? 1 : 0);
+	ast_build_string(&a_modem_next, &a_modem_left, "a=T38FaxTranscodingMMR:%d\r\n", (p->t38.jointcapability & T38FAX_TRANSCODING_MMR) ? 1 : 0);
+	ast_build_string(&a_modem_next, &a_modem_left, "a=T38FaxTranscodingJBIG:%d\r\n", (p->t38.jointcapability & T38FAX_TRANSCODING_JBIG) ? 1 : 0);
+	ast_build_string(&a_modem_next, &a_modem_left, "a=T38FaxRateManagement:%s\r\n", (p->t38.jointcapability & T38FAX_RATE_MANAGEMENT_LOCAL_TCF) ? "localTCF" : "transferredTCF");
+	x = ast_udptl_get_local_max_datagram(p->udptl);
+	ast_build_string(&a_modem_next, &a_modem_left, "a=T38FaxMaxBuffer:%d\r\n",x);
+	ast_build_string(&a_modem_next, &a_modem_left, "a=T38FaxMaxDatagram:%d\r\n",x);
+	if (p->t38.jointcapability != T38FAX_UDP_EC_NONE)
+		ast_build_string(&a_modem_next, &a_modem_left, "a=T38FaxUdpEC:%s\r\n", (p->t38.jointcapability & T38FAX_UDP_EC_REDUNDANCY) ? "t38UDPRedundancy" : "t38UDPFEC");
+	if (p->udptl)
+		len = strlen(m_modem) + strlen(a_modem);
+	add_header(resp, "Content-Type", "application/sdp");
+	add_header_contentLength(resp, len);
+	add_line(resp, v);
+	add_line(resp, o);
+	add_line(resp, s);
+	add_line(resp, c);
+	add_line(resp, t);
+	add_line(resp, m_modem);
+	add_line(resp, a_modem);
+	
+	/* Update lastrtprx when we send our SDP */
+	p->lastrtprx = p->lastrtptx = time(NULL);
+	
+	return 0;
+}
+
+
 /*! \brief Add RFC 2833 DTMF offer to SDP */
 static void add_noncodec_to_sdp(const struct sip_pvt *p, int format, int sample_rate,
 				char **m_buf, size_t *m_size, char **a_buf, size_t *a_size,
@@ -5306,6 +5773,11 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p)
 		ast_log(LOG_DEBUG, "** Our capability: %s Video flag: %s\n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), capability), ast_test_flag(&p->flags[0], SIP_NOVIDEO) ? "True" : "False");
 		ast_log(LOG_DEBUG, "** Our prefcodec: %s \n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), p->prefcodec));
 	}
+	
+	if ((ast_test_flag(&p->t38.t38support, SIP_PAGE2_T38SUPPORT_RTP))) {
+		ast_build_string(&m_audio_next, &m_audio_left, " %d", 191);
+		ast_build_string(&a_audio_next, &a_audio_left, "a=rtpmap:%d %s/%d\r\n", 191, "t38", 8000);
+	}
 
 	/* Check if we need video in this call */
 	if((capability & AST_FORMAT_VIDEO_MASK) && !ast_test_flag(&p->flags[0], SIP_NOVIDEO)) {
@@ -5494,6 +5966,26 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p)
 	return 0;
 }
 
+/*--- transmit_response_with_t38_sdp: Used for 200 OK and 183 early media ---*/
+static int transmit_response_with_t38_sdp(struct sip_pvt *p, char *msg, struct sip_request *req, int retrans)
+{
+	struct sip_request resp;
+	int seqno;
+	
+	if (sscanf(get_header(req, "CSeq"), "%d ", &seqno) != 1) {
+		ast_log(LOG_WARNING, "Unable to get seqno from '%s'\n", get_header(req, "CSeq"));
+		return -1;
+	}
+	respprep(&resp, p, msg, req);
+	if (p->udptl) {
+		ast_udptl_offered_from_local(p->udptl, 0);
+		add_t38_sdp(&resp, p);
+	} else {
+		ast_log(LOG_ERROR, "Can't add SDP to response, since we have no UDPTL session allocated. Call-ID %s\n", p->callid);
+	}
+	return send_response(p, &resp, retrans, seqno);
+}
+
 /*! \brief copy SIP request (mostly used to save request for responses) */
 static void copy_request(struct sip_request *dst, const struct sip_request *src)
 {
@@ -5595,6 +6087,31 @@ static int transmit_reinvite_with_sdp(struct sip_pvt *p)
 	return send_request(p, &req, 1, p->ocseq);
 }
 
+/*--- transmit_reinvite_with_t38_sdp: Transmit reinvite with T38 SDP ---*/
+/*     A re-invite is basically a new INVITE with the same CALL-ID and TAG as the
+       INVITE that opened the SIP dialogue
+       We reinvite so that the T38 processing can take place.
+       SIP Signalling stays with * in the path.
+*/
+static int transmit_reinvite_with_t38_sdp(struct sip_pvt *p)
+{
+	struct sip_request req;
+
+	reqprep(&req, p, ast_test_flag(&p->flags[0], SIP_REINVITE_UPDATE) ?  SIP_UPDATE : SIP_INVITE, 0, 1);
+	
+	add_header(&req, "Allow", ALLOWED_METHODS);
+	add_header(&req, "Supported", SUPPORTED_EXTENSIONS);
+	if (sipdebug)
+		add_header(&req, "X-asterisk-info", "SIP re-invite (T38 switchover)");
+	ast_udptl_offered_from_local(p->udptl, 1);
+	add_t38_sdp(&req, p);
+	/* Use this as the basis */
+	initialize_initreq(p, &req);
+	p->lastinvite = p->ocseq;
+	ast_set_flag(&p->flags[0], SIP_OUTGOING);
+	return send_request(p, &req, 1, p->ocseq);
+}
+
 /*! \brief Check Contact: URI of SIP message */
 static void extract_uri(struct sip_pvt *p, struct sip_request *req)
 {
@@ -5924,8 +6441,15 @@ static int transmit_invite(struct sip_pvt *p, int sipmethod, int sdp, int init)
 			}
 		}
 	}
-	if (sdp && p->rtp) {
-		add_sdp(&req, p);
+	if (sdp) {
+		if (p->udptl && p->t38.state == T38_LOCAL_DIRECT) {
+			ast_udptl_offered_from_local(p->udptl, 1);
+			if (option_debug)
+				ast_log(LOG_DEBUG, "T38 is in state %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>");
+			add_t38_sdp(&req, p);
+		} else if (p->rtp) {
+			add_sdp(&req, p);
+		}
 	} else {
 		add_header_contentLength(&req, 0);
 	}
@@ -8077,6 +8601,11 @@ static enum check_auth_result check_user_full(struct sip_pvt *p, struct sip_requ
 				ast_log(LOG_DEBUG, "Setting NAT on VRTP to %s\n", usenatroute ? "On" : "Off");
 			ast_rtp_setnat(p->vrtp, usenatroute);
 		}
+		if (p->udptl) {
+			if (option_debug)
+				ast_log(LOG_DEBUG, "Setting NAT on UDPTL to %s\n", usenatroute ? "On" : "Off");
+			ast_udptl_setnat(p->udptl, usenatroute);
+		}
 		if (!(res = check_auth(p, req, user->name, user->secret, user->md5secret, sipmethod, uri, reliable, ast_test_flag(req, SIP_PKT_IGNORE)))) {
 			sip_cancel_destroy(p);
 			ast_copy_flags(&p->flags[0], &user->flags[0], SIP_FLAGS_TO_COPY);
@@ -8123,7 +8652,8 @@ static enum check_auth_result check_user_full(struct sip_pvt *p, struct sip_requ
 				p->noncodeccapability |= AST_RTP_DTMF;
 			else
 				p->noncodeccapability &= ~AST_RTP_DTMF;
-
+			if (p->t38.peercapability)
+				p->t38.jointcapability &= p->t38.peercapability;
 			p->maxcallbitrate = user->maxcallbitrate;
 			/* If we do not support video, remove video from call structure */
 			if (!ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) && p->vrtp) {
@@ -8184,6 +8714,10 @@ static enum check_auth_result check_user_full(struct sip_pvt *p, struct sip_requ
 				ast_log(LOG_DEBUG, "Setting NAT on VRTP to %s\n", usenatroute ? "On" : "Off");
 				ast_rtp_setnat(p->vrtp, usenatroute);
 			}
+			if (p->udptl) {
+				ast_log(LOG_DEBUG, "Setting NAT on UDPTL to %s\n", usenatroute ? "On" : "Off");
+				ast_udptl_setnat(p->udptl, usenatroute);
+			}
 			ast_string_field_set(p, peersecret, peer->secret);
 			ast_string_field_set(p, peermd5secret, peer->md5secret);
 			ast_string_field_set(p, subscribecontext, peer->subscribecontext);
@@ -8255,6 +8789,8 @@ static enum check_auth_result check_user_full(struct sip_pvt *p, struct sip_requ
 					p->noncodeccapability |= AST_RTP_DTMF;
 				else
 					p->noncodeccapability &= ~AST_RTP_DTMF;
+				if (p->t38.peercapability)
+					p->t38.jointcapability &= p->t38.peercapability;
 			}
 			ASTOBJ_UNREF(peer, sip_destroy_peer);
 		} else { 
@@ -9055,6 +9591,9 @@ static int _sip_show_peer(int type, int fd, struct mansession *s, struct message
 		ast_cli(fd, "  Insecure     : %s\n", insecure2str(ast_test_flag(&peer->flags[0], SIP_INSECURE_PORT), ast_test_flag(&peer->flags[0], SIP_INSECURE_INVITE)));
 		ast_cli(fd, "  Nat          : %s\n", nat2str(ast_test_flag(&peer->flags[0], SIP_NAT)));
 		ast_cli(fd, "  ACL          : %s\n", (peer->ha?"Yes":"No"));
+		ast_cli(fd, "  T38 pt UDPTL : %s\n", ast_test_flag(&peer->flags[1], SIP_PAGE2_T38SUPPORT_UDPTL)?"Yes":"No");
+		ast_cli(fd, "  T38 pt RTP   : %s\n", ast_test_flag(&peer->flags[1], SIP_PAGE2_T38SUPPORT_RTP)?"Yes":"No");
+		ast_cli(fd, "  T38 pt TCP   : %s\n", ast_test_flag(&peer->flags[1], SIP_PAGE2_T38SUPPORT_TCP)?"Yes":"No");
 		ast_cli(fd, "  CanReinvite  : %s\n", ast_test_flag(&peer->flags[0], SIP_CAN_REINVITE)?"Yes":"No");
 		ast_cli(fd, "  PromiscRedir : %s\n", ast_test_flag(&peer->flags[0], SIP_PROMISCREDIR)?"Yes":"No");
 		ast_cli(fd, "  User=Phone   : %s\n", ast_test_flag(&peer->flags[0], SIP_USEREQPHONE)?"Yes":"No");
@@ -9324,6 +9863,9 @@ static int sip_show_settings(int fd, int argc, char *argv[])
 	ast_cli(fd, "  IP ToS SIP:             %s\n", ast_tos2str(global_tos_sip));
 	ast_cli(fd, "  IP ToS RTP audio:       %s\n", ast_tos2str(global_tos_audio));
 	ast_cli(fd, "  IP ToS RTP video:       %s\n", ast_tos2str(global_tos_video));
+	ast_cli(fd, "  T38 fax pt UDPTL:       %s\n", ast_test_flag(&global_flags[1], SIP_PAGE2_T38SUPPORT_UDPTL) ? "Yes" : "No");
+	ast_cli(fd, "  T38 fax pt RTP:         %s\n", ast_test_flag(&global_flags[1], SIP_PAGE2_T38SUPPORT_RTP) ? "Yes" : "No");
+	ast_cli(fd, "  T38 fax pt TCP:         %s\n", ast_test_flag(&global_flags[1], SIP_PAGE2_T38SUPPORT_TCP) ? "Yes" : "No");
 	if (!realtimepeers && !realtimeusers)
 		ast_cli(fd, "  SIP realtime:           Disabled\n" );
 	else
@@ -10476,6 +11018,11 @@ static int function_sipchaninfo_read(struct ast_channel *chan, char *cmd, char *
 		ast_copy_string(buf, p->useragent, len);
 	} else  if (!strcasecmp(data, "peername")) {
 		ast_copy_string(buf, p->peername, len);
+	} else if (!strcasecmp(data, "t38passthrough")) {
+		if (p->t38.state == T38_DISABLED)
+			ast_copy_string(buf, "0", sizeof("0"));
+		else    /* T38 is offered or enabled in this call */
+			ast_copy_string(buf, "1", sizeof("1"));
 	} else {
 		ast_channel_unlock(chan);
 		return -1;
@@ -10498,6 +11045,7 @@ static struct ast_custom_function sipchaninfo_function = {
 	"- uri                   The URI from the Contact: header.\n"
 	"- useragent             The useragent.\n"
 	"- peername              The name of the peer.\n"
+	"- t38passthrough        1 if T38 is offered or enabled in this channel, otherwise 0\n"
 };
 
 /*! \brief Parse 302 Moved temporalily response */
@@ -10672,6 +11220,51 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
 			build_route(p, req, 1);
 		}
 		
+		if (p->owner && (p->owner->_state == AST_STATE_UP)) { /* if this is a re-invite */
+			struct ast_channel *bridgepeer = NULL;
+			struct sip_pvt *bridgepvt = NULL;
+
+			bridgepeer = ast_bridged_channel(p->owner);
+			if (!bridgepeer->tech) {
+				ast_log(LOG_WARNING, "Ooooh.. no tech!  That's REALLY bad\n");
+				break;
+			}
+			if (!strcasecmp(bridgepeer->tech->type,"SIP")) {
+				bridgepvt = (struct sip_pvt*)(bridgepeer->tech_pvt);
+				if (bridgepvt->udptl) {
+					if (p->t38.state == T38_PEER_REINVITE) {
+						sip_handle_t38_reinvite(bridgepeer, p, 0);
+					} else if (p->t38.state == T38_DISABLED && bridgepeer && (bridgepvt->t38.state == T38_ENABLED)) {
+						ast_log(LOG_WARNING, "RTP re-inivte after T38 session not handled yet !\n");
+						/* Insted of this we should somehow re-invite the other side of the bridge to RTP */
+						ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
+					}
+				} else {
+					ast_log(LOG_WARNING, "Strange... The other side of the bridge don't have udptl struct\n");
+					ast_mutex_lock(&bridgepvt->lock);
+					bridgepvt->t38.state = T38_DISABLED;
+					ast_mutex_unlock(&bridgepvt->lock);
+					if (option_debug)
+						ast_log(LOG_DEBUG,"T38 state changed to %d on channel %s\n", bridgepvt->t38.state, bridgepeer->tech->type);
+					p->t38.state = T38_DISABLED;
+					if (option_debug > 1)
+						ast_log(LOG_DEBUG,"T38 state changed to %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>");
+				}
+			} else {
+				/* Other side is not a SIP channel */
+				ast_log(LOG_WARNING, "Strange... The other side of the bridge is not a SIP channel\n");
+				p->t38.state = T38_DISABLED;
+				if (option_debug > 1)
+					ast_log(LOG_DEBUG,"T38 state changed to %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>");
+			}
+		}
+		if ((p->t38.state == T38_LOCAL_REINVITE) || (p->t38.state == T38_LOCAL_DIRECT)) {
+			/* If there was T38 reinvite and we are supposed to answer with 200 OK than this should set us to T38 negotiated mode */
+			p->t38.state = T38_ENABLED;
+			if (option_debug)
+				ast_log(LOG_DEBUG, "T38 changed state to %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>");
+		}
+
 		if (!ast_test_flag(req, SIP_PKT_IGNORE) && p->owner) {
 			if (!reinvite) {
 				ast_queue_control(p->owner, AST_CONTROL_ANSWER);
@@ -11164,6 +11757,10 @@ static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_
 					/* Immediately stop VRTP */
 					ast_rtp_stop(p->vrtp);
 				}
+				if (p->udptl) {
+					/* Immediately stop UDPTL */
+					ast_udptl_stop(p->udptl);
+				}
 				/* XXX Locking issues?? XXX */
 				switch(resp) {
 				case 300: /* Multiple Choices */
@@ -12280,8 +12877,74 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
 		case AST_STATE_UP:
 			if (option_debug > 1)
 				ast_log(LOG_DEBUG, "%s: New call is UP.... \n", c->name);
-			transmit_response_with_sdp(p, "200 OK", req, XMIT_CRITICAL);
-
+			if (p->t38.state == T38_PEER_REINVITE) {
+				struct ast_channel *bridgepeer = NULL;
+				struct sip_pvt *bridgepvt = NULL;
+				
+				if ((bridgepeer = ast_bridged_channel(p->owner))) {
+					/* We have a bridge, and this is re-invite to switchover to T38 so we send re-invite with T38 SDP, to other side of bridge*/
+					/*! XXX: we should also check here does the other side supports t38 at all !!! XXX */
+					if (!strcasecmp(bridgepeer->tech->type, "SIP")) { /* If we are bridged to SIP channel */
+						bridgepvt = (struct sip_pvt*)bridgepeer->tech_pvt;
+						if (bridgepvt->t38.state == T38_DISABLED) {
+							if (bridgepvt->udptl) { /* If everything is OK with other side's udptl struct */
+								/* Send re-invite to the bridged channel */
+								sip_handle_t38_reinvite(bridgepeer, p, 1);
+							} else { /* Something is wrong with peers udptl struct */
+								ast_log(LOG_WARNING, "Strange... The other side of the bridge don't have udptl struct\n");
+								ast_mutex_lock(&bridgepvt->lock);
+								bridgepvt->t38.state = T38_DISABLED;
+								ast_mutex_unlock(&bridgepvt->lock);
+								if (option_debug > 1)
+									ast_log(LOG_DEBUG,"T38 state changed to %d on channel %s\n", bridgepvt->t38.state, bridgepeer->name);
+								if (ast_test_flag(req, SIP_PKT_IGNORE))
+									transmit_response(p, "488 Not acceptable here", req);
+								else
+									transmit_response_reliable(p, "488 Not acceptable here", req);
+								ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
+							}
+						}
+					} else {
+						/* Other side is not a SIP channel */
+						if (ast_test_flag(req, SIP_PKT_IGNORE))
+							transmit_response(p, "488 Not acceptable here", req);
+						else
+							transmit_response_reliable(p, "488 Not acceptable here", req);
+						p->t38.state = T38_DISABLED;
+						if (option_debug > 1)
+							ast_log(LOG_DEBUG,"T38 state changed to %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>");
+						ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
+					}
+				} else {
+					/* we are not bridged in a call */
+					transmit_response_with_t38_sdp(p, "200 OK", req, XMIT_CRITICAL);
+					p->t38.state = T38_ENABLED;
+					if (option_debug)
+						ast_log(LOG_DEBUG,"T38 state changed to %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>");
+				}
+			} else if (p->t38.state == T38_DISABLED) { /* Channel doesn't have T38 offered or enabled */
+				/* If we are bridged to a channel that has T38 enabled than this is a case of RTP re-invite after T38 session */
+				/* so handle it here (re-invite other party to RTP) */
+				struct ast_channel *bridgepeer = NULL;
+				struct sip_pvt *bridgepvt = NULL;
+				if ((bridgepeer = ast_bridged_channel(p->owner))) {
+					if (!strcasecmp(bridgepeer->tech->type, sip_tech.type)) {
+						bridgepvt = (struct sip_pvt*)bridgepeer->tech_pvt;
+						if (bridgepvt->t38.state == T38_ENABLED) {
+							ast_log(LOG_WARNING, "RTP re-invite after T38 session not handled yet !\n");
+							/* Insted of this we should somehow re-invite the other side of the bridge to RTP */
+							if (ast_test_flag(req, SIP_PKT_IGNORE))
+								transmit_response(p, "488 Not Acceptable Here (unsupported)", req);
+							else
+								transmit_response_reliable(p, "488 Not Acceptable Here (unsupported)", req);
+							ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
+						} else {
+							/* No bridged peer with T38 enabled*/
+							transmit_response_with_sdp(p, "200 OK", req, XMIT_CRITICAL);
+						}
+					}
+				}
+			}
 			break;
 		default:
 			ast_log(LOG_WARNING, "Don't know how to handle INVITE in state %d\n", c->_state);
@@ -12780,6 +13443,10 @@ static int handle_request_cancel(struct sip_pvt *p, struct sip_request *req)
 		/* Immediately stop VRTP */
 		ast_rtp_stop(p->vrtp);
 	}
+	if (p->udptl) {
+		/* Immediately stop UDPTL */
+		ast_udptl_stop(p->udptl);
+	}
 	if (p->owner)
 		ast_queue_hangup(p->owner);
 	else
@@ -12835,6 +13502,10 @@ static int handle_request_bye(struct sip_pvt *p, struct sip_request *req)
 		/* Immediately stop VRTP */
 		ast_rtp_stop(p->vrtp);
 	}
+	if (p->udptl) {
+		/* Immediately stop UDPTL */
+		ast_udptl_stop(p->udptl);
+	}
 	if (!ast_strlen_zero(get_header(req, "Also"))) {
 		ast_log(LOG_NOTICE, "Client '%s' using deprecated BYE/Also transfer method.  Ask vendor to support REFER instead\n",
 			ast_inet_ntoa(iabuf, sizeof(iabuf), p->recv.sin_addr));
@@ -14063,6 +14734,15 @@ static int handle_common_options(struct ast_flags *flags, struct ast_flags *mask
 	} else if (!strcasecmp(v->name, "allowsubscribe")) {
 		ast_set_flag(&mask[1], SIP_PAGE2_ALLOWSUBSCRIBE);
 		ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_ALLOWSUBSCRIBE);
+	} else if (!strcasecmp(v->name, "t38pt_udptl")) {
+		ast_set_flag(&mask[1], SIP_PAGE2_T38SUPPORT_UDPTL);
+		ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_T38SUPPORT_UDPTL);
+	} else if (!strcasecmp(v->name, "t38pt_rtp")) {
+		ast_set_flag(&mask[1], SIP_PAGE2_T38SUPPORT_RTP);
+		ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_T38SUPPORT_RTP);
+	} else if (!strcasecmp(v->name, "t38pt_tcp")) {
+		ast_set_flag(&mask[1], SIP_PAGE2_T38SUPPORT_TCP);
+		ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_T38SUPPORT_TCP);
 	}
 
 	return res;
@@ -14309,7 +14989,22 @@ static struct sip_user *build_user(const char *name, struct ast_variable *v, int
 			user->maxcallbitrate = atoi(v->value);
 			if (user->maxcallbitrate < 0)
 				user->maxcallbitrate = default_maxcallbitrate;
- 		}
+ 		} else if (!strcasecmp(v->name, "t38pt_udptl")) {
+			if (ast_true(v->value)) {
+				ast_set_flag(&user->flags[1], SIP_PAGE2_T38SUPPORT_UDPTL);
+			} else
+				ast_clear_flag(&user->flags[1], SIP_PAGE2_T38SUPPORT_UDPTL);
+		} else if (!strcasecmp(v->name, "t38pt_rtp")) {
+			if (ast_true(v->value)) {
+				ast_set_flag(&user->flags[1], SIP_PAGE2_T38SUPPORT_RTP);
+			} else
+				ast_clear_flag(&user->flags[1], SIP_PAGE2_T38SUPPORT_RTP);
+		} else if (!strcasecmp(v->name, "t38pt_tcp")) {
+			if (ast_true(v->value)) {
+				ast_set_flag(&user->flags[1], SIP_PAGE2_T38SUPPORT_TCP);
+			} else
+				ast_clear_flag(&user->flags[1], SIP_PAGE2_T38SUPPORT_TCP);
+		}
 	}
 	ast_copy_flags(&user->flags[0], &userflags[0], mask[0].flags);
 	ast_copy_flags(&user->flags[1], &userflags[1], mask[1].flags);
@@ -14591,6 +15286,21 @@ static struct sip_peer *build_peer(const char *name, struct ast_variable *v, int
 			peer->maxcallbitrate = atoi(v->value);
 			if (peer->maxcallbitrate < 0)
 				peer->maxcallbitrate = default_maxcallbitrate;
+		} else if (!strcasecmp(v->name, "t38pt_udptl")) {
+			if (ast_true(v->value)) {
+				ast_set_flag(&peer->flags[1], SIP_PAGE2_T38SUPPORT_UDPTL);
+			} else
+				ast_clear_flag(&peer->flags[1], SIP_PAGE2_T38SUPPORT_UDPTL);
+		} else if (!strcasecmp(v->name, "t38pt_rtp")) {
+			if (ast_true(v->value)) {
+				ast_set_flag(&peer->flags[1], SIP_PAGE2_T38SUPPORT_RTP);
+			} else
+				ast_clear_flag(&peer->flags[1], SIP_PAGE2_T38SUPPORT_RTP);
+		} else if (!strcasecmp(v->name, "t38pt_tcp")) {
+			if (ast_true(v->value)) {
+				ast_set_flag(&peer->flags[1], SIP_PAGE2_T38SUPPORT_TCP);
+			} else
+				ast_clear_flag(&peer->flags[1], SIP_PAGE2_T38SUPPORT_TCP);
 		}
 	}
 	if (!ast_test_flag(&global_flags[1], SIP_PAGE2_IGNOREREGEXPIRE) && ast_test_flag(&peer->flags[1], SIP_PAGE2_DYNAMIC) && realtime) {
@@ -14952,6 +15662,18 @@ static int reload_config(enum channelreloadreason reason)
 			default_maxcallbitrate = atoi(v->value);
 			if (default_maxcallbitrate < 0)
 				default_maxcallbitrate = DEFAULT_MAX_CALL_BITRATE;
+		} else if (!strcasecmp(v->name, "t38pt_udptl")) {
+			if (ast_true(v->value)) {
+				ast_set_flag(&global_flags[1], SIP_PAGE2_T38SUPPORT_UDPTL);
+			}
+		} else if (!strcasecmp(v->name, "t38pt_rtp")) {
+			if (ast_true(v->value)) {
+				ast_set_flag(&global_flags[1], SIP_PAGE2_T38SUPPORT_RTP);
+			}
+		} else if (!strcasecmp(v->name, "t38pt_tcp")) {
+			if (ast_true(v->value)) {
+				ast_set_flag(&global_flags[1], SIP_PAGE2_T38SUPPORT_TCP);
+			}
 		}
 	}
 
@@ -15093,6 +15815,131 @@ static int reload_config(enum channelreloadreason reason)
 	return 0;
 }
 
+static struct ast_udptl *sip_get_udptl_peer(struct ast_channel *chan)
+{
+	struct sip_pvt *p;
+	struct ast_udptl *udptl = NULL;
+	
+	p = chan->tech_pvt;
+	if (!p)
+		return NULL;
+	
+	ast_mutex_lock(&p->lock);
+	if (p->udptl && ast_test_flag(&p->flags[0], SIP_CAN_REINVITE))
+		udptl = p->udptl;
+	ast_mutex_unlock(&p->lock);
+	return udptl;
+}
+
+static int sip_set_udptl_peer(struct ast_channel *chan, struct ast_udptl *udptl)
+{
+	struct sip_pvt *p;
+	
+	p = chan->tech_pvt;
+	if (!p)
+		return -1;
+	ast_mutex_lock(&p->lock);
+	if (udptl)
+		ast_udptl_get_peer(udptl, &p->udptlredirip);
+	else
+		memset(&p->udptlredirip, 0, sizeof(p->udptlredirip));
+	if (!ast_test_flag(&p->flags[0], SIP_GOTREFER)) {
+		if (!p->pendinginvite) {
+			if (option_debug > 2) {
+				char iabuf[INET_ADDRSTRLEN];
+				ast_log(LOG_DEBUG, "Sending reinvite on SIP '%s' - It's UDPTL soon redirected to IP %s:%d\n", p->callid, ast_inet_ntoa(iabuf, sizeof(iabuf), udptl ? p->udptlredirip.sin_addr : p->ourip), udptl ? ntohs(p->udptlredirip.sin_port) : 0);
+			}
+			transmit_reinvite_with_t38_sdp(p);
+		} else if (!ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
+			if (option_debug > 2) {
+				char iabuf[INET_ADDRSTRLEN];
+				ast_log(LOG_DEBUG, "Deferring reinvite on SIP '%s' - It's UDPTL will be redirected to IP %s:%d\n", p->callid, ast_inet_ntoa(iabuf, sizeof(iabuf),udptl ? p->udptlredirip.sin_addr : p->ourip), udptl ? ntohs(p->udptlredirip.sin_port) : 0);
+			}
+			ast_set_flag(&p->flags[0], SIP_NEEDREINVITE);
+		}
+	}
+	/* Reset lastrtprx timer */
+	p->lastrtprx = p->lastrtptx = time(NULL);
+	ast_mutex_unlock(&p->lock);
+	return 0;
+}
+
+static int sip_handle_t38_reinvite(struct ast_channel *chan, struct sip_pvt *pvt, int reinvite)
+{
+	struct sip_pvt *p;
+	int flag = 0;
+	
+	p = chan->tech_pvt;
+	if (!p || !pvt->udptl)
+		return -1;
+	
+	/* Setup everything on the other side like offered/responded from first side */
+	ast_mutex_lock(&p->lock);
+	p->t38.jointcapability = p->t38.peercapability = pvt->t38.jointcapability;
+	ast_udptl_set_far_max_datagram(p->udptl, ast_udptl_get_local_max_datagram(pvt->udptl));
+	ast_udptl_set_local_max_datagram(p->udptl, ast_udptl_get_local_max_datagram(pvt->udptl));
+	ast_udptl_set_error_correction_scheme(p->udptl, ast_udptl_get_error_correction_scheme(pvt->udptl));
+	
+	if (reinvite) {		/* If we are handling sending re-invite to the other side of the bridge */
+		if (ast_test_flag(&p->flags[0], SIP_CAN_REINVITE) && ast_test_flag(&pvt->flags[0], SIP_CAN_REINVITE)) {
+			ast_udptl_get_peer(pvt->udptl, &p->udptlredirip);
+			flag =1;
+		} else {
+			memset(&p->udptlredirip, 0, sizeof(p->udptlredirip));
+		}
+		if (!ast_test_flag(&p->flags[0], SIP_GOTREFER)) {
+			if (!p->pendinginvite) {
+				if (option_debug > 2) {
+					char iabuf[INET_ADDRSTRLEN];
+					if (flag)
+						ast_log(LOG_DEBUG, "Sending reinvite on SIP '%s' - It's UDPTL soon redirected to IP %s:%d\n", p->callid, ast_inet_ntoa(iabuf, sizeof(iabuf), p->udptlredirip.sin_addr), ntohs(p->udptlredirip.sin_port));
+					else
+						ast_log(LOG_DEBUG, "Sending reinvite on SIP '%s' - It's UDPTL soon redirected to us (IP %s)\n", p->callid, ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip));
+				}
+				transmit_reinvite_with_t38_sdp(p);
+			} else if (!ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
+				if (option_debug > 2) {
+					char iabuf[INET_ADDRSTRLEN];
+					if (flag)
+						ast_log(LOG_DEBUG, "Deferring reinvite on SIP '%s' - It's UDPTL will be redirected to IP %s:%d\n", p->callid, ast_inet_ntoa(iabuf, sizeof(iabuf), p->udptlredirip.sin_addr), ntohs(p->udptlredirip.sin_port));
+					else
+						ast_log(LOG_DEBUG, "Deferring reinvite on SIP '%s' - It's UDPTL will be redirected to us (IP %s)\n", p->callid, ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip));
+				}
+				ast_set_flag(&p->flags[0], SIP_NEEDREINVITE);
+			}
+		}
+		/* Reset lastrtprx timer */
+		p->lastrtprx = p->lastrtptx = time(NULL);
+		ast_mutex_unlock(&p->lock);
+		return 0;
+	} else {	/* If we are handling sending 200 OK to the other side of the bridge */
+		if (ast_test_flag(&p->flags[0], SIP_CAN_REINVITE) && ast_test_flag(&pvt->flags[0], SIP_CAN_REINVITE)) {
+			ast_udptl_get_peer(pvt->udptl, &p->udptlredirip);
+			flag = 1;
+		} else {
+			memset(&p->udptlredirip, 0, sizeof(p->udptlredirip));
+		}
+		if (option_debug > 2) {
+			char iabuf[INET_ADDRSTRLEN];
+			if (flag)
+				ast_log(LOG_DEBUG, "Responding 200 OK on SIP '%s' - It's UDPTL soon redirected to IP %s:%d\n", p->callid, ast_inet_ntoa(iabuf, sizeof(iabuf), p->udptlredirip.sin_addr), ntohs(p->udptlredirip.sin_port));
+			else
+				ast_log(LOG_DEBUG, "Responding 200 OK on SIP '%s' - It's UDPTL soon redirected to us (IP %s)\n", p->callid, ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip));
+		}
+		pvt->t38.state = T38_ENABLED;
+		p->t38.state = T38_ENABLED;
+		if (option_debug > 1) {
+			ast_log(LOG_DEBUG, "T38 changed state to %d on channel %s\n", pvt->t38.state, pvt->owner ? pvt->owner->name : "<none>");
+			ast_log(LOG_DEBUG, "T38 changed state to %d on channel %s\n", p->t38.state, chan ? chan->name : "<none>");
+		}
+		transmit_response_with_t38_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL);
+		p->lastrtprx = p->lastrtptx = time(NULL);
+		ast_mutex_unlock(&p->lock);
+		return 0;
+	}
+}
+
+
 /*! \brief Returns null if we can't reinvite audio (part of RTP interface) */
 static struct ast_rtp *sip_get_rtp_peer(struct ast_channel *chan)
 {
@@ -15522,6 +16369,9 @@ static int load_module(void *mod)
 	/* Tell the RTP subdriver that we're here */
 	ast_rtp_proto_register(&sip_rtp);
 
+	/* Tell the UDPTL subdriver that we're here */
+	ast_udptl_proto_register(&sip_udptl);
+
 	/* Register dialplan applications */
 	ast_register_application(app_dtmfmode, sip_dtmfmode, synopsis_dtmfmode, descrip_dtmfmode);
 	ast_register_application(app_sipaddheader, sip_addheader, synopsis_sipaddheader, descrip_sipaddheader);
@@ -15566,6 +16416,8 @@ static int unload_module(void *mod)
 
 	ast_rtp_proto_unregister(&sip_rtp);
 
+	ast_udptl_proto_unregister(&sip_udptl);
+
 	ast_manager_unregister("SIPpeers");
 	ast_manager_unregister("SIPshowpeer");