chan_sip.c

	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) {
		free_old_route(p->route);
		p->route = NULL;
	}
	if (p->registry) {
		if (p->registry->call == p)
			p->registry->call = NULL;
		unref_registry(p->registry);
	}

	/* Unlink us from the owner if we have one */
	if (p->owner) {
		if (lockowner)
			ast_channel_lock(p->owner);
		if (option_debug)
			ast_log(LOG_DEBUG, "Detaching from %s\n", p->owner->name);
		p->owner->tech_pvt = NULL;
		if (lockowner)
			ast_channel_unlock(p->owner);
	}
	/* Clear history */
	if (p->history) {
		struct sip_history *hist;
		while( (hist = AST_LIST_REMOVE_HEAD(p->history, list)) )
			free(hist);
		free(p->history);
		p->history = NULL;
	}

	/* Lock dialog list before removing ourselves from the list */
	if (lockdialoglist)
		dialoglist_lock();
	for (prev = NULL, cur = dialoglist; cur; prev = cur, cur = cur->next) {
		if (cur == p) {
			UNLINK(cur, dialoglist, prev);
			break;
		}
	}
	if (lockdialoglist)
		dialoglist_unlock();
	if (!cur) {
		ast_log(LOG_WARNING, "Trying to destroy \"%s\", not found in dialog list?!?! \n", p->callid);
		return;
	} 

	/* remove all current packets in this dialog */
	while((cp = p->packets)) {
		p->packets = p->packets->next;
		if (cp->retransid > -1)
			ast_sched_del(sched, cp->retransid);
		free(cp);
	}
	if (p->chanvars) {
		ast_variables_destroy(p->chanvars);
		p->chanvars = NULL;
	}
	ast_mutex_destroy(&p->pvt_lock);

	ast_string_field_free_pools(p);

	free(p);
}

/*! \brief  update_call_counter: Handle call_limit for SIP users 
 * Setting a call-limit will cause calls above the limit not to be accepted.
 *
 * Remember that for a type=friend, there's one limit for the user and
 * another for the peer, not a combined call limit.
 * This will cause unexpected behaviour in subscriptions, since a "friend"
 * is *two* devices in Asterisk, not one.
 *
 * Thought: For realtime, we should probably update storage with inuse counter... 
 *
 * \return 0 if call is ok (no call limit, below threshold)
 *	-1 on rejection of call
 *		
 */
static int update_call_counter(struct sip_pvt *fup, int event)
{
	char name[256];
	int *inuse = NULL, *call_limit = NULL, *inringing = NULL;
	int outgoing = ast_test_flag(&fup->flags[0], SIP_OUTGOING);
	struct sip_user *u = NULL;
	struct sip_peer *p = NULL;

	if (option_debug > 2)
		ast_log(LOG_DEBUG, "Updating call counter for %s call\n", outgoing ? "outgoing" : "incoming");
	/* Test if we need to check call limits, in order to avoid 
	   realtime lookups if we do not need it */
	if (!ast_test_flag(&fup->flags[0], SIP_CALL_LIMIT))
		return 0;

	ast_copy_string(name, fup->username, sizeof(name));

	/* Check the list of users only for incoming calls */
	if (global_limitonpeers == FALSE && !outgoing && (u = find_user(name, 1)))  {
		inuse = &u->inUse;
		call_limit = &u->call_limit;
		inringing = NULL;
	} else if ( (p = find_peer(ast_strlen_zero(fup->peername) ? name : fup->peername, NULL, 1) ) ) { /* Try to find peer */
		inuse = &p->inUse;
		call_limit = &p->call_limit;
		inringing = &p->inRinging;
		ast_copy_string(name, fup->peername, sizeof(name));
	} 
	if (!p && !u) {
		if (option_debug > 1)
			ast_log(LOG_DEBUG, "%s is not a local device, no call limit\n", name);
		return 0;
	}

	switch(event) {
	/* incoming and outgoing affects the inUse counter */
	case DEC_CALL_LIMIT:
		if ( *inuse > 0 ) {
			if (ast_test_flag(&fup->flags[0], SIP_INC_COUNT))
				(*inuse)--;
		} else {
			*inuse = 0;
		}
		if (inringing) {
			if (ast_test_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING)) {
				if (*inringing > 0)
					(*inringing)--;
				else
					ast_log(LOG_WARNING, "Inringing for peer '%s' < 0?\n", fup->peername);
				ast_clear_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING);
			}
		}
		if (option_debug > 1 || sipdebug) {
			ast_log(LOG_DEBUG, "Call %s %s '%s' removed from call limit %d\n", outgoing ? "to" : "from", u ? "user":"peer", name, *call_limit);
		}
		break;

	case INC_CALL_RINGING:
	case INC_CALL_LIMIT:
		if (*call_limit > 0 ) {
			if (*inuse >= *call_limit) {
				ast_log(LOG_ERROR, "Call %s %s '%s' rejected due to usage limit of %d\n", outgoing ? "to" : "from", u ? "user":"peer", name, *call_limit);
				if (u)
					unref_user(u);
				else
					unref_peer(p);
				return -1; 
			}
		}
		if (inringing && (event == INC_CALL_RINGING)) {
			if (!ast_test_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING)) {
				(*inringing)++;
				ast_set_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING);
			}
		}
		/* Continue */
		(*inuse)++;
		ast_set_flag(&fup->flags[0], SIP_INC_COUNT);
		if (option_debug > 1 || sipdebug) {
			ast_log(LOG_DEBUG, "Call %s %s '%s' is %d out of %d\n", outgoing ? "to" : "from", u ? "user":"peer", name, *inuse, *call_limit);
		}
		break;

	case DEC_CALL_RINGING:
		if (inringing) {
			if (ast_test_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING)) {
				if (*inringing > 0)
					(*inringing)--;
				else
					ast_log(LOG_WARNING, "Inringing for peer '%s' < 0?\n", p->name);
				ast_clear_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING);
			}
		}
		break;

	default:
		ast_log(LOG_ERROR, "update_call_counter(%s, %d) called with no event!\n", name, event);
	}
	if (p) {
		ast_device_state_changed("SIP/%s", p->name);
		unref_peer(p);
	} else /* u must be set */
		unref_user(u);
	return 0;
}

/*! \brief Destroy SIP call structure */
static void sip_destroy(struct sip_pvt *p)
{
	if (option_debug > 2)
		ast_log(LOG_DEBUG, "Destroying SIP dialog %s\n", p->callid);
	__sip_destroy(p, TRUE, TRUE);
}

/*! \brief Convert SIP hangup causes to Asterisk hangup causes */
static int hangup_sip2cause(int cause)
{
	/* Possible values taken from causes.h */

	switch(cause) {
		case 401:	/* Unauthorized */
			return AST_CAUSE_CALL_REJECTED;
		case 403:	/* Not found */
			return AST_CAUSE_CALL_REJECTED;
		case 404:	/* Not found */
			return AST_CAUSE_UNALLOCATED;
		case 405:	/* Method not allowed */
			return AST_CAUSE_INTERWORKING;
		case 407:	/* Proxy authentication required */
			return AST_CAUSE_CALL_REJECTED;
		case 408:	/* No reaction */
			return AST_CAUSE_NO_USER_RESPONSE;
		case 409:	/* Conflict */
			return AST_CAUSE_NORMAL_TEMPORARY_FAILURE;
		case 410:	/* Gone */
			return AST_CAUSE_UNALLOCATED;
		case 411:	/* Length required */
			return AST_CAUSE_INTERWORKING;
		case 413:	/* Request entity too large */
			return AST_CAUSE_INTERWORKING;
		case 414:	/* Request URI too large */
			return AST_CAUSE_INTERWORKING;
		case 415:	/* Unsupported media type */
			return AST_CAUSE_INTERWORKING;
		case 420:	/* Bad extension */
			return AST_CAUSE_NO_ROUTE_DESTINATION;
		case 480:	/* No answer */
			return AST_CAUSE_NO_ANSWER;
		case 481:	/* No answer */
			return AST_CAUSE_INTERWORKING;
		case 482:	/* Loop detected */
			return AST_CAUSE_INTERWORKING;
		case 483:	/* Too many hops */
			return AST_CAUSE_NO_ANSWER;
		case 484:	/* Address incomplete */
			return AST_CAUSE_INVALID_NUMBER_FORMAT;
		case 485:	/* Ambiguous */
			return AST_CAUSE_UNALLOCATED;
		case 486:	/* Busy everywhere */
			return AST_CAUSE_BUSY;
		case 487:	/* Request terminated */
			return AST_CAUSE_INTERWORKING;
		case 488:	/* No codecs approved */
			return AST_CAUSE_BEARERCAPABILITY_NOTAVAIL;
		case 491:	/* Request pending */
			return AST_CAUSE_INTERWORKING;
		case 493:	/* Undecipherable */
			return AST_CAUSE_INTERWORKING;
		case 500:	/* Server internal failure */
			return AST_CAUSE_FAILURE;
		case 501:	/* Call rejected */
			return AST_CAUSE_FACILITY_REJECTED;
		case 502:	
			return AST_CAUSE_DESTINATION_OUT_OF_ORDER;
		case 503:	/* Service unavailable */
			return AST_CAUSE_CONGESTION;
		case 504:	/* Gateway timeout */
			return AST_CAUSE_RECOVERY_ON_TIMER_EXPIRE;
		case 505:	/* SIP version not supported */
			return AST_CAUSE_INTERWORKING;
		case 600:	/* Busy everywhere */
			return AST_CAUSE_USER_BUSY;
		case 603:	/* Decline */
			return AST_CAUSE_CALL_REJECTED;
		case 604:	/* Does not exist anywhere */
			return AST_CAUSE_UNALLOCATED;
		case 606:	/* Not acceptable */
			return AST_CAUSE_BEARERCAPABILITY_NOTAVAIL;
		default:
			return AST_CAUSE_NORMAL;
	}
	/* Never reached */
	return 0;
}

/*! \brief Convert Asterisk hangup causes to SIP codes 
\verbatim
 Possible values from causes.h
        AST_CAUSE_NOTDEFINED    AST_CAUSE_NORMAL        AST_CAUSE_BUSY
        AST_CAUSE_FAILURE       AST_CAUSE_CONGESTION    AST_CAUSE_UNALLOCATED

	In addition to these, a lot of PRI codes is defined in causes.h 
	...should we take care of them too ?
	
	Quote RFC 3398

   ISUP Cause value                        SIP response
   ----------------                        ------------
   1  unallocated number                   404 Not Found
   2  no route to network                  404 Not found
   3  no route to destination              404 Not found
   16 normal call clearing                 --- (*)
   17 user busy                            486 Busy here
   18 no user responding                   408 Request Timeout
   19 no answer from the user              480 Temporarily unavailable
   20 subscriber absent                    480 Temporarily unavailable
   21 call rejected                        403 Forbidden (+)
   22 number changed (w/o diagnostic)      410 Gone
   22 number changed (w/ diagnostic)       301 Moved Permanently
   23 redirection to new destination       410 Gone
   26 non-selected user clearing           404 Not Found (=)
   27 destination out of order             502 Bad Gateway
   28 address incomplete                   484 Address incomplete
   29 facility rejected                    501 Not implemented
   31 normal unspecified                   480 Temporarily unavailable
\endverbatim
*/
static const char *hangup_cause2sip(int cause)
{
	switch (cause) {
		case AST_CAUSE_UNALLOCATED:		/* 1 */
		case AST_CAUSE_NO_ROUTE_DESTINATION:	/* 3 IAX2: Can't find extension in context */
		case AST_CAUSE_NO_ROUTE_TRANSIT_NET:	/* 2 */
			return "404 Not Found";
		case AST_CAUSE_CONGESTION:		/* 34 */
		case AST_CAUSE_SWITCH_CONGESTION:	/* 42 */
			return "503 Service Unavailable";
		case AST_CAUSE_NO_USER_RESPONSE:	/* 18 */
			return "408 Request Timeout";
		case AST_CAUSE_NO_ANSWER:		/* 19 */
			return "480 Temporarily unavailable";
		case AST_CAUSE_CALL_REJECTED:		/* 21 */
			return "403 Forbidden";
		case AST_CAUSE_NUMBER_CHANGED:		/* 22 */
			return "410 Gone";
		case AST_CAUSE_NORMAL_UNSPECIFIED:	/* 31 */
			return "480 Temporarily unavailable";
		case AST_CAUSE_INVALID_NUMBER_FORMAT:
			return "484 Address incomplete";
		case AST_CAUSE_USER_BUSY:
			return "486 Busy here";
		case AST_CAUSE_FAILURE:
			return "500 Server internal failure";
		case AST_CAUSE_FACILITY_REJECTED:	/* 29 */
			return "501 Not Implemented";
		case AST_CAUSE_CHAN_NOT_IMPLEMENTED:
			return "503 Service Unavailable";
		/* Used in chan_iax2 */
		case AST_CAUSE_DESTINATION_OUT_OF_ORDER:
			return "502 Bad Gateway";
		case AST_CAUSE_BEARERCAPABILITY_NOTAVAIL:	/* Can't find codec to connect to host */
			return "488 Not Acceptable Here";
			
		case AST_CAUSE_NOTDEFINED:
		default:
			if (option_debug)
				ast_log(LOG_DEBUG, "AST hangup cause %d (no match found in SIP)\n", cause);
			return NULL;
	}

	/* Never reached */
	return 0;
}


/*! \brief  sip_hangup: Hangup SIP call
 * Part of PBX interface, called from ast_hangup */
static int sip_hangup(struct ast_channel *ast)
{
	struct sip_pvt *p = ast->tech_pvt;
	int needcancel = FALSE;
	int needdestroy = 0;
	struct ast_channel *oldowner = ast;

	if (!p) {
		if (option_debug)
			ast_log(LOG_DEBUG, "Asked to hangup channel that was not connected\n");
		return 0;
	}

	if (ast_test_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER)) {
		if (option_debug >3)
			ast_log(LOG_DEBUG, "SIP Transfer: Not hanging up right now... Rescheduling hangup for %s.\n", p->callid);
		if (p->autokillid > -1)
			sip_cancel_destroy(p);
		sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
		ast_clear_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER);	/* Really hang up next time */
		ast_clear_flag(&p->flags[0], SIP_NEEDDESTROY);
		p->owner->tech_pvt = NULL;
		p->owner = NULL;  /* Owner will be gone after we return, so take it away */
		return 0;
	}
	if (option_debug) {
		if (ast_test_flag(ast, AST_FLAG_ZOMBIE) && p->refer && option_debug)
         		ast_log(LOG_DEBUG, "SIP Transfer: Hanging up Zombie channel %s after transfer ... Call-ID: %s\n", ast->name, p->callid);
		else  {
			if (option_debug)
				ast_log(LOG_DEBUG, "Hangup call %s, SIP callid %s)\n", ast->name, p->callid);
		}
	}
	if (option_debug && ast_test_flag(ast, AST_FLAG_ZOMBIE)) 
		ast_log(LOG_DEBUG, "Hanging up zombie call. Be scared.\n");

	sip_pvt_lock(p);
	if (option_debug && sipdebug)
		ast_log(LOG_DEBUG, "update_call_counter(%s) - decrement call limit counter on hangup\n", p->username);
	update_call_counter(p, DEC_CALL_LIMIT);

	/* Determine how to disconnect */
	if (p->owner != ast) {
		ast_log(LOG_WARNING, "Huh?  We aren't the owner? Can't hangup call.\n");
		sip_pvt_unlock(p);
		return 0;
	}
	/* If the call is not UP, we need to send CANCEL instead of BYE */
	/* In case of re-invites, the call might be UP even though we have an incomplete invite transaction */
	if (p->invitestate < INV_COMPLETED && p->owner->_state != AST_STATE_UP) {
		needcancel = TRUE;
		if (option_debug > 3)
			ast_log(LOG_DEBUG, "Hanging up channel in state %s (not UP)\n", ast_state2str(ast->_state));
	}

	/* Disconnect */
	if (p->vad)
		ast_dsp_free(p->vad);

	p->owner = NULL;
	ast->tech_pvt = NULL;

	/* Do not destroy this pvt until we have timeout or
	   get an answer to the BYE or INVITE/CANCEL 
	   If we get no answer during retransmit period, drop the call anyway.
	   (Sorry, mother-in-law, you can't deny a hangup by sending
	   603 declined to BYE...)
	*/
	if (ast_test_flag(&p->flags[0], SIP_ALREADYGONE))
		needdestroy = 1;	/* Set destroy flag at end of this function */
	else if (p->invitestate != INV_CALLING)
		sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);

	/* Start the process if it's not already started */
	if (!ast_test_flag(&p->flags[0], SIP_ALREADYGONE) && !ast_strlen_zero(p->initreq.data)) {
		if (needcancel) {	/* Outgoing call, not up */
			if (ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
				/* stop retransmitting an INVITE that has not received a response */
				__sip_pretend_ack(p);

				/* if we can't send right now, mark it pending */
				if (p->invitestate == INV_CALLING) {
					/* We can't send anything in CALLING state */
					ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
					/* Do we need a timer here if we don't hear from them at all? */
				} else {
					/* Send a new request: CANCEL */
					transmit_request(p, SIP_CANCEL, p->ocseq, XMIT_RELIABLE, FALSE);
					/* Actually don't destroy us yet, wait for the 487 on our original 
					   INVITE, but do set an autodestruct just in case we never get it. */
					needdestroy = 0;
					sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
				}
				if ( p->initid != -1 ) {
					/* channel still up - reverse dec of inUse counter
					   only if the channel is not auto-congested */
					update_call_counter(p, INC_CALL_LIMIT);
				}
			} else {	/* Incoming call, not up */
				const char *res;
				if (ast->hangupcause && (res = hangup_cause2sip(ast->hangupcause)))
					transmit_response_reliable(p, res, &p->initreq);
				else 
					transmit_response_reliable(p, "603 Declined", &p->initreq);
			}
		} else {	/* Call is in UP state, send BYE */
			if (!p->pendinginvite) {
				char *audioqos = "";
				char *videoqos = "";
				if (p->rtp)
					audioqos = ast_rtp_get_quality(p->rtp);
				if (p->vrtp)
					videoqos = ast_rtp_get_quality(p->vrtp);
				/* Send a hangup */
				transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, 1);

				/* Get RTCP quality before end of call */
				if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY)) {
					if (p->rtp)
						append_history(p, "RTCPaudio", "Quality:%s", audioqos);
					if (p->vrtp)
						append_history(p, "RTCPvideo", "Quality:%s", videoqos);
				}
				if (p->rtp && oldowner)
					pbx_builtin_setvar_helper(oldowner, "RTPAUDIOQOS", audioqos);
				if (p->vrtp && oldowner)
					pbx_builtin_setvar_helper(oldowner, "RTPVIDEOQOS", videoqos);
			} else {
				/* Note we will need a BYE when this all settles out
				   but we can't send one while we have "INVITE" outstanding. */
				ast_set_flag(&p->flags[0], SIP_PENDINGBYE);	
				ast_clear_flag(&p->flags[0], SIP_NEEDREINVITE);	
				sip_cancel_destroy(p);
			}
		}
	}
	if (needdestroy)
		ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
	sip_pvt_unlock(p);
	return 0;
}

/*! \brief Try setting codec suggested by the SIP_CODEC channel variable */
static void try_suggested_sip_codec(struct sip_pvt *p)
{
	int fmt;
	const char *codec;

	codec = pbx_builtin_getvar_helper(p->owner, "SIP_CODEC");
	if (!codec) 
		return;

	fmt = ast_getformatbyname(codec);
	if (fmt) {
		ast_log(LOG_NOTICE, "Changing codec to '%s' for this call because of ${SIP_CODEC} variable\n", codec);
		if (p->jointcapability & fmt) {
			p->jointcapability &= fmt;
			p->capability &= fmt;
		} else
			ast_log(LOG_NOTICE, "Ignoring ${SIP_CODEC} variable because it is not shared by both ends.\n");
	} else
		ast_log(LOG_NOTICE, "Ignoring ${SIP_CODEC} variable because of unrecognized/not configured codec (check allow/disallow in sip.conf): %s\n", codec);
	return;	
}

/*! \brief  sip_answer: Answer SIP call , send 200 OK on Invite 
 * Part of PBX interface */
static int sip_answer(struct ast_channel *ast)
{
	int res = 0;
	struct sip_pvt *p = ast->tech_pvt;

	sip_pvt_lock(p);
	if (ast->_state != AST_STATE_UP) {
		try_suggested_sip_codec(p);	

		ast_setstate(ast, AST_STATE_UP);
		if (option_debug)
			ast_log(LOG_DEBUG, "SIP answering channel: %s\n", ast->name);
		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);
	}
	sip_pvt_unlock(p);
	return res;
}

/*! \brief Send frame to media channel (rtp) */
static int sip_write(struct ast_channel *ast, struct ast_frame *frame)
{
	struct sip_pvt *p = ast->tech_pvt;
	int res = 0;

	switch (frame->frametype) {
	case AST_FRAME_VOICE:
		if (!(frame->subclass & ast->nativeformats)) {
			char s1[512], s2[512], s3[512];
			ast_log(LOG_WARNING, "Asked to transmit frame type %d, while native formats is %s(%d) read/write = %s(%d)/%s(%d)\n",
				frame->subclass, 
				ast_getformatname_multiple(s1, sizeof(s1) - 1, ast->nativeformats & AST_FORMAT_AUDIO_MASK),
				ast->nativeformats & AST_FORMAT_AUDIO_MASK,
				ast_getformatname_multiple(s2, sizeof(s2) - 1, ast->readformat),
				ast->readformat,
				ast_getformatname_multiple(s3, sizeof(s3) - 1, ast->writeformat),
				ast->writeformat);
			return 0;
		}
		if (p) {
			sip_pvt_lock(p);
			if (p->rtp) {
				/* If channel is not up, activate early media session */
				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_sdp(p, "183 Session Progress", &p->initreq, XMIT_UNRELIABLE);
					ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);	
				}
				p->lastrtptx = time(NULL);
				res = ast_rtp_write(p->rtp, frame);
			}
			sip_pvt_unlock(p);
		}
		break;
	case AST_FRAME_VIDEO:
		if (p) {
			sip_pvt_lock(p);
			if (p->vrtp) {
				/* Activate video early media */
				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_sdp(p, "183 Session Progress", &p->initreq, XMIT_UNRELIABLE);
					ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);	
				}
				p->lastrtptx = time(NULL);
				res = ast_rtp_write(p->vrtp, frame);
			}
			sip_pvt_unlock(p);
		}
		break;
	case AST_FRAME_IMAGE:
		return 0;
		break;
	case AST_FRAME_MODEM:
		if (p) {
			sip_pvt_lock(p);
			/* UDPTL requires two-way communication, so early media is not needed here.
				we simply forget the frames if we get modem frames before the bridge is up.
				Fax will re-transmit.
			*/
			if (p->udptl && ast->_state != AST_STATE_UP) 
				res = ast_udptl_write(p->udptl, frame);
			sip_pvt_unlock(p);
		}
		break;
	default: 
		ast_log(LOG_WARNING, "Can't send %d type frames with SIP write\n", frame->frametype);
		return 0;
	}

	return res;
}

/*! \brief  sip_fixup: Fix up a channel:  If a channel is consumed, this is called.
        Basically update any ->owner links */
static int sip_fixup(struct ast_channel *oldchan, struct ast_channel *newchan)
{
	int ret = -1;
	struct sip_pvt *p;

	if (newchan && ast_test_flag(newchan, AST_FLAG_ZOMBIE) && option_debug)
		ast_log(LOG_DEBUG, "New channel is zombie\n");
	if (oldchan && ast_test_flag(oldchan, AST_FLAG_ZOMBIE) && option_debug)
		ast_log(LOG_DEBUG, "Old channel is zombie\n");

	if (!newchan || !newchan->tech_pvt) {
		if (!newchan)
			ast_log(LOG_WARNING, "No new channel! Fixup of %s failed.\n", oldchan->name);
		else
			ast_log(LOG_WARNING, "No SIP tech_pvt! Fixup of %s failed.\n", oldchan->name);
		return -1;
	}
	p = newchan->tech_pvt;

	sip_pvt_lock(p);
	append_history(p, "Masq", "Old channel: %s\n", oldchan->name);
	append_history(p, "Masq (cont)", "...new owner: %s\n", newchan->name);
	if (p->owner != oldchan)
		ast_log(LOG_WARNING, "old channel wasn't %p but was %p\n", oldchan, p->owner);
	else {
		p->owner = newchan;
		ret = 0;
	}
	if (option_debug > 2)
		ast_log(LOG_DEBUG, "SIP Fixup: New owner for dialogue %s: %s (Old parent: %s)\n", p->callid, p->owner->name, oldchan->name);

	sip_pvt_unlock(p);
	return ret;
}

static int sip_senddigit_begin(struct ast_channel *ast, char digit)
{
	struct sip_pvt *p = ast->tech_pvt;
	int res = 0;

	sip_pvt_lock(p);
	switch (ast_test_flag(&p->flags[0], SIP_DTMF)) {
	case SIP_DTMF_INBAND:
		res = -1; /* Tell Asterisk to generate inband indications */
		break;
	case SIP_DTMF_RFC2833:
		if (p->rtp)
			ast_rtp_senddigit_begin(p->rtp, digit);
		break;
	default:
		break;
	}
	sip_pvt_unlock(p);

	return res;
}

/*! \brief Send DTMF character on SIP channel
	within one call, we're able to transmit in many methods simultaneously */
static int sip_senddigit_end(struct ast_channel *ast, char digit)
{
	struct sip_pvt *p = ast->tech_pvt;
	int res = 0;

	sip_pvt_lock(p);
	switch (ast_test_flag(&p->flags[0], SIP_DTMF)) {
	case SIP_DTMF_INFO:
		transmit_info_with_digit(p, digit);
		break;
	case SIP_DTMF_RFC2833:
		if (p->rtp)
			ast_rtp_senddigit_end(p->rtp, digit);
		break;
	case SIP_DTMF_INBAND:
		res = -1; /* Tell Asterisk to stop inband indications */
		break;
	}
	sip_pvt_unlock(p);

	return res;
}

/*! \brief Transfer SIP call */
static int sip_transfer(struct ast_channel *ast, const char *dest)
{
	struct sip_pvt *p = ast->tech_pvt;
	int res;

	if (dest == NULL)	/* functions below do not take a NULL */
		dest = "";
	sip_pvt_lock(p);
	if (ast->_state == AST_STATE_RING)
		res = sip_sipredirect(p, dest);
	else
		res = transmit_refer(p, dest);
	sip_pvt_unlock(p);
	return res;
}

/*! \brief Play indication to user 
 * With SIP a lot of indications is sent as messages, letting the device play
   the indication - busy signal, congestion etc 
   \return -1 to force ast_indicate to send indication in audio, 0 if SIP can handle the indication by sending a message
*/
static int sip_indicate(struct ast_channel *ast, int condition, const void *data, size_t datalen)
{
	struct sip_pvt *p = ast->tech_pvt;
	int res = 0;

	sip_pvt_lock(p);
	switch(condition) {
	case AST_CONTROL_RINGING:
		if (ast->_state == AST_STATE_RING) {
			p->invitestate = INV_EARLY_MEDIA;
			if (!ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) ||
			    (ast_test_flag(&p->flags[0], SIP_PROG_INBAND) == SIP_PROG_INBAND_NEVER)) {				
				/* Send 180 ringing if out-of-band seems reasonable */
				transmit_response(p, "180 Ringing", &p->initreq);
				ast_set_flag(&p->flags[0], SIP_RINGING);
				if (ast_test_flag(&p->flags[0], SIP_PROG_INBAND) != SIP_PROG_INBAND_YES)
					break;
			} else {
				/* Well, if it's not reasonable, just send in-band */
			}
		}
		res = -1;
		break;
	case AST_CONTROL_BUSY:
		if (ast->_state != AST_STATE_UP) {
			transmit_response(p, "486 Busy Here", &p->initreq);
			p->invitestate = INV_COMPLETED;
			sip_alreadygone(p);
			ast_softhangup_nolock(ast, AST_SOFTHANGUP_DEV);
			break;
		}
		res = -1;
		break;
	case AST_CONTROL_CONGESTION:
		if (ast->_state != AST_STATE_UP) {
			transmit_response(p, "503 Service Unavailable", &p->initreq);
			p->invitestate = INV_COMPLETED;
			sip_alreadygone(p);
			ast_softhangup_nolock(ast, AST_SOFTHANGUP_DEV);
			break;
		}
		res = -1;
		break;
	case AST_CONTROL_PROCEEDING:
		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(p, "100 Trying", &p->initreq);
			p->invitestate = INV_PROCEEDING;  
			break;
		}
		res = -1;
		break;
	case AST_CONTROL_PROGRESS:
		if ((ast->_state != AST_STATE_UP) &&
		    !ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) &&
		    !ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
			p->invitestate = INV_EARLY_MEDIA;
			transmit_response_with_sdp(p, "183 Session Progress", &p->initreq, XMIT_UNRELIABLE);
			ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);	
			break;
		}
		res = -1;
		break;
	case AST_CONTROL_HOLD:
		ast_moh_start(ast, data, p->mohinterpret);
		break;
	case AST_CONTROL_UNHOLD:
		ast_moh_stop(ast);
		break;
	case AST_CONTROL_VIDUPDATE:	/* Request a video frame update */
		if (p->vrtp && !ast_test_flag(&p->flags[0], SIP_NOVIDEO)) {
			transmit_info_with_vidupdate(p);
			/* ast_rtcp_send_h261fur(p->vrtp); */
		} else
			res = -1;
		break;
	case -1:
		res = -1;
		break;
	default:
		ast_log(LOG_WARNING, "Don't know how to indicate condition %d\n", condition);
		res = -1;
		break;
	}
	sip_pvt_unlock(p);
	return res;
}



/*! \brief Initiate a call in the SIP channel
	called from sip_request_call (calls from the pbx ) for outbound channels
	and from handle_request_invite for inbound channels
	
*/
static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *title)
{
	struct ast_channel *tmp;
	struct ast_variable *v = NULL;
	int fmt;
	int what;
	int needvideo = 0;
	{
		const char *my_name;	/* pick a good name */
	
		if (title)
			my_name = title;
		else if ( (my_name = strchr(i->fromdomain,':')) )
			my_name++;	/* skip ':' */
		else
			my_name = i->fromdomain;

		sip_pvt_unlock(i);
		/* Don't hold a sip pvt lock while we allocate a channel */

		tmp = ast_channel_alloc(1, state, i->cid_num, i->cid_name, "SIP/%s-%08x", my_name, (int)(long) i);

	}
	sip_pvt_lock(i);
	if (!tmp) {
		ast_log(LOG_WARNING, "Unable to allocate AST channel structure for SIP channel\n");
		return NULL;
	}
	tmp->tech = &sip_tech;

	/* Select our native format based on codec preference until we receive
	   something from another device to the contrary. */
	if (i->jointcapability)	 	/* The joint capabilities of us and peer */
		what = i->jointcapability;
	else if (i->capability)		/* Our configured capability for this peer */
		what = i->capability;
	else
		what = global_capability;	/* Global codec support */

	/* Set the native formats for audio  and merge in video */
	tmp->nativeformats = ast_codec_choose(&i->prefs, what, 1) | (i->jointcapability & AST_FORMAT_VIDEO_MASK);
	if (option_debug > 2) {
		char buf[BUFSIZ];
		ast_log(LOG_DEBUG, "*** Our native formats are %s \n", ast_getformatname_multiple(buf, BUFSIZ, tmp->nativeformats));
		ast_log(LOG_DEBUG, "*** Joint capabilities are %s \n", ast_getformatname_multiple(buf, BUFSIZ, i->jointcapability));
		ast_log(LOG_DEBUG, "*** Our capabilities are %s \n", ast_getformatname_multiple(buf, BUFSIZ, i->capability));
		ast_log(LOG_DEBUG, "*** AST_CODEC_CHOOSE formats are %s \n", ast_getformatname_multiple(buf, BUFSIZ, ast_codec_choose(&i->prefs, what, 1)));
		if (i->prefcodec)
			ast_log(LOG_DEBUG, "*** Our preferred formats from the incoming channel are %s \n", ast_getformatname_multiple(buf, BUFSIZ, i->prefcodec));
	}

	/* XXX Why are we choosing a codec from the native formats?? */
	fmt = ast_best_codec(tmp->nativeformats);

	/* If we have a prefcodec setting, we have an inbound channel that set a 
	   preferred format for this call. Otherwise, we check the jointcapability
	   We also check for vrtp. If it's not there, we are not allowed do any video anyway.
	 */
	if (i->vrtp) {
		if (i->prefcodec)
			needvideo = i->prefcodec & AST_FORMAT_VIDEO_MASK;	/* Outbound call */
 		else
			needvideo = i->jointcapability & AST_FORMAT_VIDEO_MASK;	/* Inbound call */
	}

	if (option_debug > 2) {
		if (needvideo) 
			ast_log(LOG_DEBUG, "This channel can handle video! HOLLYWOOD next!\n");
		else
			ast_log(LOG_DEBUG, "This channel will not be able to handle video.\n");
	}



	if (ast_test_flag(&i->flags[0], SIP_DTMF) ==  SIP_DTMF_INBAND) {
		i->vad = ast_dsp_new();
		ast_dsp_set_features(i->vad, DSP_FEATURE_DTMF_DETECT);
		if (global_relaxdtmf)
			ast_dsp_digitmode(i->vad, DSP_DIGITMODE_DTMF | DSP_DIGITMODE_RELAXDTMF);
	}
	if (i->rtp) {
		tmp->fds[0] = ast_rtp_fd(i->rtp);
		tmp->fds[1] = ast_rtcp_fd(i->rtp);
	}
	if (needvideo && i->vrtp) {
		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;
	tmp->writeformat = fmt;
	tmp->rawwriteformat = fmt;
	tmp->readformat = fmt;
	tmp->rawreadformat = fmt;
	tmp->tech_pvt = i;

	tmp->callgroup = i->callgroup;
	tmp->pickupgroup = i->pickupgroup;
	tmp->cid.cid_pres = i->callingpres;
	if (!ast_strlen_zero(i->accountcode))
		ast_string_field_set(tmp, accountcode, i->accountcode);
	if (i->amaflags)
		tmp->amaflags = i->amaflags;
	if (!ast_strlen_zero(i->language))
		ast_string_field_set(tmp, language, i->language);
	i->owner = tmp;
	ast_copy_string(tmp->context, i->context, sizeof(tmp->context));
	ast_copy_string(tmp->exten, i->exten, sizeof(tmp->exten));

	/* Don't use ast_set_callerid() here because it will
	 * generate an unnecessary NewCallerID event  */
	tmp->cid.cid_num = ast_strdup(i->cid_num);
	tmp->cid.cid_ani = ast_strdup(i->cid_num);
	tmp->cid.cid_name = ast_strdup(i->cid_name);
	if (!ast_strlen_zero(i->rdnis))
		tmp->cid.cid_rdnis = ast_strdup(i->rdnis);
	
	if (!ast_strlen_zero(i->exten) && strcmp(i->exten, "s"))
		tmp->cid.cid_dnid = ast_strdup(i->exten);

	tmp->priority = 1;
	if (!ast_strlen_zero(i->uri))
		pbx_builtin_setvar_helper(tmp, "SIPURI", i->uri);
	if (!ast_strlen_zero(i->domain))
		pbx_builtin_setvar_helper(tmp, "SIPDOMAIN", i->domain);
	if (!ast_strlen_zero(i->callid))
		pbx_builtin_setvar_helper(tmp, "SIPCALLID", i->callid);
	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;
	}
	/* Set channel variables for this call from configuration */
	for (v = i->chanvars ; v ; v = v->next)
		pbx_builtin_setvar_helper(tmp,v->name,v->value);

	if (!ast_test_flag(&i->flags[0], SIP_NO_HISTORY))
		append_history(i, "NewChan", "Channel %s - from %s", tmp->name, i->callid);

	return tmp;
}

/*! \brief Reads one line of SIP message body */
static char *get_body_by_line(const char *line, const char *name, int nameLen)
{
	if (strncasecmp(line, name, nameLen) == 0 && line[nameLen] == '=')
		return ast_skip_blanks(line + nameLen + 1);

	return "";
}

/*! \brief Lookup 'name' in the SDP starting
 * at the 'start' line. Returns the matching line, and 'start'
 * is updated with the next line number.
 */
static const char *get_sdp_iterate(int *start, struct sip_request *req, const char *name)
{
	int len = strlen(name);

	while (*start < req->sdp_end) {
		const char *r = get_body_by_line(req->line[(*start)++], name, len);
		if (r[0] != '\0')
			return r;
	}