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Add attended transfer support for chan_sip.c

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This now uses the core API for performing attended transfers.

Review https://reviewboard.asterisk.org/r/2513

(Closes issue ASTERISK-21520)
reported by Matt Jordan



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389869 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Mark Michelson
2013-05-28 15:26:15 +00:00
parent fac3839e68
commit cfe32ec1da
2 changed files with 129 additions and 431 deletions
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540
channels/chan_sip.c
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@@ -1200,7 +1200,8 @@ static int build_path(struct sip_pvt *p, struct sip_peer *peer, struct sip_reque
static int copy_route(struct sip_route **dst, const struct sip_route *src);
static enum check_auth_result register_verify(struct sip_pvt *p, struct ast_sockaddr *addr,
struct sip_request *req, const char *uri);
static struct sip_pvt *get_sip_pvt_byid_locked(const char *callid, const char *totag, const char *fromtag);
static int get_sip_pvt_from_replaces(const char *callid, const char *totag, const char *fromtag,
struct sip_pvt **out_pvt, struct ast_channel **out_chan);
static void check_pendings(struct sip_pvt *p);
static void *sip_pickup_thread(void *stuff);
@@ -1271,8 +1272,6 @@ static struct ast_channel *sip_pvt_lock_full(struct sip_pvt *pvt);
/* static int sip_addrcmp(char *name, struct sockaddr_in *sin); Support for peer matching */
static int sip_refer_alloc(struct sip_pvt *p);
static int sip_notify_alloc(struct sip_pvt *p);
static void ast_quiet_chan(struct ast_channel *chan);
static int attempt_transfer(struct sip_dual *transferer, struct sip_dual *target);
static int do_magic_pickup(struct ast_channel *channel, const char *extension, const char *context);
static void set_peer_nat(const struct sip_pvt *p, struct sip_peer *peer);
static void check_for_nat(const struct ast_sockaddr *them, struct sip_pvt *p);
@@ -1475,9 +1474,10 @@ static int handle_request_message(struct sip_pvt *p, struct sip_request *req, st
static int handle_request_subscribe(struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, uint32_t seqno, const char *e);
static void handle_request_info(struct sip_pvt *p, struct sip_request *req);
static int handle_request_options(struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, const char *e);
static int handle_invite_replaces(struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, uint32_t seqno, int *nounlock);
static int handle_invite_replaces(struct sip_pvt *p, struct sip_request *req,
int *nounlock, struct sip_pvt *replaces_pvt, struct ast_channel *replaces_chan);
static int handle_request_notify(struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, uint32_t seqno, const char *e);
static int local_attended_transfer(struct sip_pvt *transferer, struct sip_dual *current, struct sip_request *req, uint32_t seqno, int *nounlock);
static int local_attended_transfer(struct sip_pvt *transferer, struct ast_channel *transferer_chan, uint32_t seqno, int *nounlock);
/*------Response handling functions */
static void handle_response_publish(struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, uint32_t seqno);
@@ -6654,9 +6654,6 @@ void __sip_destroy(struct sip_pvt *p, int lockowner, int lockdialoglist)
p->udptl = NULL;
}
if (p->refer) {
if (p->refer->refer_call) {
p->refer->refer_call = dialog_unref(p->refer->refer_call, "unref dialog p->refer->refer_call");
}
ast_string_field_free_memory(p->refer);
ast_free(p->refer);
p->refer = NULL;
@@ -17811,10 +17808,25 @@ static enum sip_get_dest_result get_destination(struct sip_pvt *p, struct sip_re
return SIP_GET_DEST_EXTEN_NOT_FOUND;
}
/*! \brief Lock dialog lock and find matching pvt lock
\return a reference, remember to release it when done
*/
static struct sip_pvt *get_sip_pvt_byid_locked(const char *callid, const char *totag, const char *fromtag)
/*! \brief Find a companion dialog based on Replaces information
*
* This information may come from a Refer-To header in a REFER or from
* a Replaces header in an INVITE.
*
* This function will find the appropriate sip_pvt and increment the refcount
* of both the sip_pvt and its owner channel. These two references are returned
* in the out parameters
*
* \param callid Callid to search for
* \param totag to-tag parameter from Replaces
* \param fromtag from-tag parameter from Replaces
* \param[out] out_pvt The found sip_pvt.
* \param[out] out_chan The found sip_pvt's owner channel.
* \retval 0 Success
* \retval non-zero Failure
*/
static int get_sip_pvt_from_replaces(const char *callid, const char *totag,
const char *fromtag, struct sip_pvt **out_pvt, struct ast_channel **out_chan)
{
struct sip_pvt *sip_pvt_ptr;
struct sip_pvt tmp_dialog = {
@@ -17830,22 +17842,20 @@ static struct sip_pvt *get_sip_pvt_byid_locked(const char *callid, const char *t
sip_pvt_ptr = ao2_t_find(dialogs, &tmp_dialog, OBJ_POINTER, "ao2_find of dialog in dialogs table");
if (sip_pvt_ptr) {
/* Go ahead and lock it (and its owner) before returning */
sip_pvt_lock(sip_pvt_ptr);
SCOPED_LOCK(lock, sip_pvt_ptr, sip_pvt_lock, sip_pvt_unlock);
if (sip_cfg.pedanticsipchecking) {
unsigned char frommismatch = 0, tomismatch = 0;
if (ast_strlen_zero(fromtag)) {
sip_pvt_unlock(sip_pvt_ptr);
ast_debug(4, "Matched %s call for callid=%s - no from tag specified, pedantic check fails\n",
sip_pvt_ptr->outgoing_call == TRUE ? "OUTGOING": "INCOMING", sip_pvt_ptr->callid);
return NULL;
return -1;
}
if (ast_strlen_zero(totag)) {
sip_pvt_unlock(sip_pvt_ptr);
ast_debug(4, "Matched %s call for callid=%s - no to tag specified, pedantic check fails\n",
sip_pvt_ptr->outgoing_call == TRUE ? "OUTGOING": "INCOMING", sip_pvt_ptr->callid);
return NULL;
return -1;
}
/* RFC 3891
* > 3. User Agent Server Behavior: Receiving a Replaces Header
@@ -17868,7 +17878,6 @@ static struct sip_pvt *get_sip_pvt_byid_locked(const char *callid, const char *t
* can change when the call is not answered yet.
*/
if ((frommismatch && ast_test_flag(&sip_pvt_ptr->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED)) || tomismatch) {
sip_pvt_unlock(sip_pvt_ptr);
if (frommismatch) {
ast_debug(4, "Matched %s call for callid=%s - pedantic from tag check fails; their tag is %s our tag is %s\n",
sip_pvt_ptr->outgoing_call == TRUE ? "OUTGOING": "INCOMING", sip_pvt_ptr->callid,
@@ -17879,7 +17888,7 @@ static struct sip_pvt *get_sip_pvt_byid_locked(const char *callid, const char *t
sip_pvt_ptr->outgoing_call == TRUE ? "OUTGOING": "INCOMING", sip_pvt_ptr->callid,
totag, sip_pvt_ptr->tag);
}
return NULL;
return -1;
}
}
@@ -17888,15 +17897,13 @@ static struct sip_pvt *get_sip_pvt_byid_locked(const char *callid, const char *t
sip_pvt_ptr->outgoing_call == TRUE ? "OUTGOING": "INCOMING",
sip_pvt_ptr->theirtag, sip_pvt_ptr->tag);
/* deadlock avoidance... */
while (sip_pvt_ptr->owner && ast_channel_trylock(sip_pvt_ptr->owner)) {
sip_pvt_unlock(sip_pvt_ptr);
usleep(1);
sip_pvt_lock(sip_pvt_ptr);
*out_pvt = sip_pvt_ptr;
if (out_chan) {
*out_chan = sip_pvt_ptr->owner ? ast_channel_ref(sip_pvt_ptr->owner) : NULL;
}
}
return sip_pvt_ptr;
return 0;
}
/*! \brief Call transfer support (the REFER method)
@@ -24451,90 +24458,6 @@ static int sip_pickup(struct ast_channel *chan)
return 0;
}
/*! \brief Turn off generator data
XXX Does this function belong in the SIP channel?
*/
static void ast_quiet_chan(struct ast_channel *chan)
{
if (chan && ast_channel_state(chan) == AST_STATE_UP) {
if (ast_test_flag(ast_channel_flags(chan), AST_FLAG_MOH))
ast_moh_stop(chan);
else if (ast_channel_generatordata(chan))
ast_deactivate_generator(chan);
}
}
/*! \brief Attempt transfer of SIP call
This fix for attended transfers on a local PBX */
static int attempt_transfer(struct sip_dual *transferer, struct sip_dual *target)
{
int res = 0;
struct ast_channel *peera = NULL,
*peerb = NULL,
*peerc = NULL,
*peerd = NULL;
/* We will try to connect the transferee with the target and hangup
all channels to the transferer */
ast_debug(4, "Sip transfer:--------------------\n");
if (transferer->chan1)
ast_debug(4, "-- Transferer to PBX channel: %s State %s\n", ast_channel_name(transferer->chan1), ast_state2str(ast_channel_state(transferer->chan1)));
else
ast_debug(4, "-- No transferer first channel - odd??? \n");
if (target->chan1)
ast_debug(4, "-- Transferer to PBX second channel (target): %s State %s\n", ast_channel_name(target->chan1), ast_state2str(ast_channel_state(target->chan1)));
else
ast_debug(4, "-- No target first channel ---\n");
if (transferer->chan2)
ast_debug(4, "-- Bridged call to transferee: %s State %s\n", ast_channel_name(transferer->chan2), ast_state2str(ast_channel_state(transferer->chan2)));
else
ast_debug(4, "-- No bridged call to transferee\n");
if (target->chan2)
ast_debug(4, "-- Bridged call to transfer target: %s State %s\n", target->chan2 ? ast_channel_name(target->chan2) : "<none>", target->chan2 ? ast_state2str(ast_channel_state(target->chan2)) : "(none)");
else
ast_debug(4, "-- No target second channel ---\n");
ast_debug(4, "-- END Sip transfer:--------------------\n");
if (transferer->chan2) { /* We have a bridge on the transferer's channel */
peera = transferer->chan1; /* Transferer - PBX -> transferee channel * the one we hangup */
peerb = target->chan1; /* Transferer - PBX -> target channel - This will get lost in masq */
peerc = transferer->chan2; /* Asterisk to Transferee */
peerd = target->chan2; /* Asterisk to Target */
ast_debug(3, "SIP transfer: Four channels to handle\n");
} else if (target->chan2) { /* Transferer has no bridge (IVR), but transferee */
peera = target->chan1; /* Transferer to PBX -> target channel */
peerb = transferer->chan1; /* Transferer to IVR*/
peerc = target->chan2; /* Asterisk to Target */
peerd = transferer->chan2; /* Nothing */
ast_debug(3, "SIP transfer: Three channels to handle\n");
}
if (peera && peerb && peerc && (peerb != peerc)) {
ast_quiet_chan(peera); /* Stop generators */
ast_quiet_chan(peerb);
ast_quiet_chan(peerc);
if (peerd)
ast_quiet_chan(peerd);
ast_debug(4, "SIP transfer: trying to masquerade %s into %s\n", ast_channel_name(peerc), ast_channel_name(peerb));
if (ast_channel_masquerade(peerb, peerc)) {
ast_log(LOG_WARNING, "Failed to masquerade %s into %s\n", ast_channel_name(peerb), ast_channel_name(peerc));
res = -1;
} else
ast_debug(4, "SIP transfer: Succeeded to masquerade channels.\n");
return res;
} else {
ast_log(LOG_NOTICE, "SIP Transfer attempted with no appropriate bridged calls to transfer\n");
if (transferer->chan1)
ast_softhangup_nolock(transferer->chan1, AST_SOFTHANGUP_DEV);
if (target->chan1)
ast_softhangup_nolock(target->chan1, AST_SOFTHANGUP_DEV);
return -1;
}
return 0;
}
/*! \brief Get tag from packet
*
* \return Returns the pointer to the provided tag buffer,
@@ -24850,132 +24773,68 @@ static int handle_request_options(struct sip_pvt *p, struct sip_request *req, st
}
/*! \brief Handle the transfer part of INVITE with a replaces: header,
meaning a target pickup or an attended transfer.
Used only once.
XXX 'ignore' is unused.
\note this function is called by handle_request_invite(). Four locks
held at the beginning of this function, p, p->owner, p->refer->refer_call and
p->refere->refer_call->owner. only p's lock should remain at the end of this
function. p's lock as well as the channel p->owner's lock are held by
handle_request_do(), we unlock p->owner before the masq. By setting nounlock
we are indicating to handle_request_do() that we have already unlocked the owner.
*
* This is used for call-pickup and for attended transfers initiated on
* remote endpoints (i.e. a REFER received on a remote server).
*
* \note p and p->owner are locked upon entering this function. If the
* call pickup or attended transfer is successful, then p->owner will
* be unlocked upon exiting this function. This is communicated to the
* caller through the nounlock parameter.
*
* \param p The sip_pvt where the INVITE with Replaces was received
* \param req The incoming INVITE
* \param[out] nounlock Indicator if p->owner should remained locked. On successful transfer, this will be set true.
* \param replaces_pvt sip_pvt referenced by Replaces header
* \param replaces_chan replaces_pvt's owner channel
* \retval 0 Success
* \retval non-zero Failure
*/
static int handle_invite_replaces(struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, uint32_t seqno, int *nounlock)
static int handle_invite_replaces(struct sip_pvt *p, struct sip_request *req,
int *nounlock, struct sip_pvt *replaces_pvt, struct ast_channel *replaces_chan)
{
int earlyreplace = 0;
int oneleggedreplace = 0; /* Call with no bridge, propably IVR or voice message */
struct ast_channel *c = p->owner; /* Our incoming call */
struct ast_channel *replacecall = p->refer->refer_call->owner; /* The channel we're about to take over */
struct ast_channel *targetcall; /* The bridge to the take-over target */
/* Check if we're in ring state */
if (ast_channel_state(replacecall) == AST_STATE_RING)
earlyreplace = 1;
/* Check if we have a bridge */
if (!(targetcall = ast_bridged_channel(replacecall))) {
/* We have no bridge */
if (!earlyreplace) {
ast_debug(2, " Attended transfer attempted to replace call with no bridge (maybe ringing). Channel %s!\n", ast_channel_name(replacecall));
oneleggedreplace = 1;
}
}
if (targetcall && ast_channel_state(targetcall) == AST_STATE_RINGING)
ast_debug(4, "SIP transfer: Target channel is in ringing state\n");
if (targetcall)
ast_debug(4, "SIP transfer: Invite Replace incoming channel should bridge to channel %s while hanging up channel %s\n", ast_channel_name(targetcall), ast_channel_name(replacecall));
else
ast_debug(4, "SIP transfer: Invite Replace incoming channel should replace and hang up channel %s (one call leg)\n", ast_channel_name(replacecall));
RAII_VAR(struct ast_bridge *, bridge, NULL, ao2_cleanup);
RAII_VAR(struct ast_channel *, c, NULL, ao2_cleanup);
if (req->ignore) {
ast_log(LOG_NOTICE, "Ignoring this INVITE with replaces in a stupid way.\n");
/* We should answer something here. If we are here, the
call we are replacing exists, so an accepted
can't harm */
transmit_response_with_sdp(p, "200 OK", req, XMIT_RELIABLE, FALSE, FALSE);
/* Do something more clever here */
if (c) {
*nounlock = 1;
ast_channel_unlock(c);
return 0;
}
ast_channel_unlock(replacecall);
sip_pvt_unlock(p->refer->refer_call);
return 1;
}
if (!c) {
if (!p->owner) {
/* What to do if no channel ??? */
ast_log(LOG_ERROR, "Unable to create new channel. Invite/replace failed.\n");
transmit_response_reliable(p, "503 Service Unavailable", req);
append_history(p, "Xfer", "INVITE/Replace Failed. No new channel.");
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
ast_channel_unlock(replacecall);
sip_pvt_unlock(p->refer->refer_call);
return 1;
}
append_history(p, "Xfer", "INVITE/Replace received");
/* We have three channels to play with
channel c: New incoming call
targetcall: Call from PBX to target
p->refer->refer_call: SIP pvt dialog from transferer to pbx.
replacecall: The owner of the previous
We need to masq C into refer_call to connect to
targetcall;
If we are talking to internal audio stream, target call is null.
*/
c = ast_channel_ref(p->owner);
/* Fake call progress */
transmit_response(p, "100 Trying", req);
ast_setstate(c, AST_STATE_RING);
/* Masquerade the new call into the referred call to connect to target call
Targetcall is not touched by the masq */
ast_debug(4, "Invite/Replaces: preparing to replace %s with %s\n", ast_channel_name(replaces_chan), ast_channel_name(c));
/* Answer the incoming call and set channel to UP state */
transmit_response_with_sdp(p, "200 OK", req, XMIT_RELIABLE, FALSE, FALSE);
ast_setstate(c, AST_STATE_UP);
/* Stop music on hold and other generators */
ast_quiet_chan(replacecall);
ast_quiet_chan(targetcall);
ast_debug(4, "Invite/Replaces: preparing to masquerade %s into %s\n", ast_channel_name(c), ast_channel_name(replacecall));
/* Make sure that the masq does not free our PVT for the old call */
if (! earlyreplace && ! oneleggedreplace )
ast_set_flag(&p->refer->refer_call->flags[0], SIP_DEFER_BYE_ON_TRANSFER); /* Delay hangup */
/* Prepare the masquerade - if this does not happen, we will be gone */
if(ast_channel_masquerade(replacecall, c))
ast_log(LOG_ERROR, "Failed to masquerade C into Replacecall\n");
else
ast_debug(4, "Invite/Replaces: Going to masquerade %s into %s\n", ast_channel_name(c), ast_channel_name(replacecall));
/* C should now be in place of replacecall. all channel locks and pvt locks should be removed
* before issuing the masq. Since we are unlocking both the pvt (p) and its owner channel (c)
* it is possible for channel c to be destroyed on us. To prevent this, we must give c a reference
* before any unlocking takes place and remove it only once we are completely done with it */
ast_channel_ref(c);
ast_channel_unlock(replacecall);
ast_channel_unlock(c);
sip_pvt_unlock(p->refer->refer_call);
sip_pvt_unlock(p);
ast_do_masquerade(replacecall);
ast_channel_lock(c);
if (earlyreplace || oneleggedreplace ) {
ast_channel_hangupcause_set(c, AST_CAUSE_SWITCH_CONGESTION);
}
ast_setstate(c, AST_STATE_DOWN);
ast_channel_unlock(c);
/* c and c's tech pvt must be unlocked at this point for ast_hangup */
ast_hangup(c);
/* this indicates to handle_request_do that the owner channel has already been unlocked */
*nounlock = 1;
/* lock PVT structure again after hangup */
ast_channel_unlock(c);
sip_pvt_unlock(p);
ast_raw_answer(c, 1);
ast_channel_lock(replaces_chan);
bridge = ast_channel_get_bridge(replaces_chan);
ast_channel_unlock(replaces_chan);
if (bridge) {
ast_bridge_impart(bridge, c, replaces_chan, NULL, 1);
} else {
ast_channel_move(replaces_chan, c);
ast_hangup(c);
}
sip_pvt_lock(p);
ast_channel_unref(c);
return 0;
}
@@ -25085,7 +24944,6 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, str
int gotdest;
const char *p_replaces;
char *replace_id = NULL;
int refer_locked = 0;
const char *required;
unsigned int required_profile = 0;
struct ast_channel *c = NULL; /* New channel */
@@ -25110,6 +24968,8 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, str
} pickup = {
.exten = "",
};
RAII_VAR(struct sip_pvt *, replaces_pvt, NULL, ao2_cleanup);
RAII_VAR(struct ast_channel *, replaces_chan, NULL, ao2_cleanup);
/* Find out what they support */
if (!p->sipoptions) {
@@ -25287,45 +25147,41 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, str
First we cheat a little and look for a magic call-id from phones that support
dialog-info+xml so we can do technology independent pickup... */
if (strncmp(replace_id, "pickup-", 7) == 0) {
struct sip_pvt *subscription = NULL;
RAII_VAR(struct sip_pvt *, subscription, NULL, ao2_cleanup);
RAII_VAR(struct ast_channel *, subscription_chan, NULL, ao2_cleanup);
replace_id += 7; /* Worst case we are looking at \0 */
if ((subscription = get_sip_pvt_byid_locked(replace_id, totag, fromtag)) == NULL) {
if (get_sip_pvt_from_replaces(replace_id, totag, fromtag, &subscription, &subscription_chan)) {
ast_log(LOG_NOTICE, "Unable to find subscription with call-id: %s\n", replace_id);
transmit_response_reliable(p, "481 Call Leg Does Not Exist (Replaces)", req);
error = 1;
} else {
SCOPED_LOCK(lock, subscription, sip_pvt_lock, sip_pvt_unlock);
ast_log(LOG_NOTICE, "Trying to pick up %s@%s\n", subscription->exten, subscription->context);
ast_copy_string(pickup.exten, subscription->exten, sizeof(pickup.exten));
ast_copy_string(pickup.context, subscription->context, sizeof(pickup.context));
sip_pvt_unlock(subscription);
if (subscription->owner) {
ast_channel_unlock(subscription->owner);
}
subscription = dialog_unref(subscription, "unref dialog subscription");
}
}
/* This locks both refer_call pvt and refer_call pvt's owner!!!*/
if (!error && ast_strlen_zero(pickup.exten) && (p->refer->refer_call = get_sip_pvt_byid_locked(replace_id, totag, fromtag)) == NULL) {
if (!error && ast_strlen_zero(pickup.exten) && get_sip_pvt_from_replaces(replace_id,
totag, fromtag, &replaces_pvt, &replaces_chan)) {
ast_log(LOG_NOTICE, "Supervised transfer attempted to replace non-existent call id (%s)!\n", replace_id);
transmit_response_reliable(p, "481 Call Leg Does Not Exist (Replaces)", req);
error = 1;
} else {
refer_locked = 1;
}
/* The matched call is the call from the transferer to Asterisk .
We want to bridge the bridged part of the call to the
incoming invite, thus taking over the refered call */
if (p->refer->refer_call == p) {
if (replaces_pvt == p) {
ast_log(LOG_NOTICE, "INVITE with replaces into it's own call id (%s == %s)!\n", replace_id, p->callid);
transmit_response_reliable(p, "400 Bad request", req); /* The best way to not not accept the transfer */
error = 1;
}
if (!error && ast_strlen_zero(pickup.exten) && !p->refer->refer_call->owner) {
if (!error && ast_strlen_zero(pickup.exten) && !replaces_chan) {
/* Oops, someting wrong anyway, no owner, no call */
ast_log(LOG_NOTICE, "Supervised transfer attempted to replace non-existing call id (%s)!\n", replace_id);
/* Check for better return code */
@@ -25333,7 +25189,10 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, str
error = 1;
}
if (!error && ast_strlen_zero(pickup.exten) && ast_channel_state(p->refer->refer_call->owner) != AST_STATE_RINGING && ast_channel_state(p->refer->refer_call->owner) != AST_STATE_RING && ast_channel_state(p->refer->refer_call->owner) != AST_STATE_UP) {
if (!error && ast_strlen_zero(pickup.exten) &&
ast_channel_state(replaces_chan) != AST_STATE_RINGING &&
ast_channel_state(replaces_chan) != AST_STATE_RING &&
ast_channel_state(replaces_chan) != AST_STATE_UP) {
ast_log(LOG_NOTICE, "Supervised transfer attempted to replace non-ringing or active call id (%s)!\n", replace_id);
transmit_response_reliable(p, "603 Declined (Replaces)", req);
error = 1;
@@ -25342,15 +25201,6 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, str
if (error) { /* Give up this dialog */
append_history(p, "Xfer", "INVITE/Replace Failed.");
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
sip_pvt_unlock(p);
if (p->refer->refer_call) {
sip_pvt_unlock(p->refer->refer_call);
if (p->refer->refer_call->owner) {
ast_channel_unlock(p->refer->refer_call->owner);
}
p->refer->refer_call = dialog_unref(p->refer->refer_call, "unref dialog p->refer->refer_call");
}
refer_locked = 0;
p->invitestate = INV_COMPLETED;
res = INV_REQ_ERROR;
check_via(p, req);
@@ -25791,9 +25641,8 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, str
} else {
/* Go and take over the target call */
if (sipdebug)
ast_debug(4, "Sending this call to the invite/replcaes handler %s\n", p->callid);
res = handle_invite_replaces(p, req, addr, seqno, nounlock);
refer_locked = 0;
ast_debug(4, "Sending this call to the invite/replaces handler %s\n", p->callid);
res = handle_invite_replaces(p, req, nounlock, replaces_pvt, replaces_chan);
goto request_invite_cleanup;
}
}
@@ -25932,13 +25781,6 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, str
request_invite_cleanup:
if (refer_locked && p->refer && p->refer->refer_call) {
sip_pvt_unlock(p->refer->refer_call);
if (p->refer->refer_call->owner) {
ast_channel_unlock(p->refer->refer_call->owner);
}
p->refer->refer_call = dialog_unref(p->refer->refer_call, "unref dialog p->refer->refer_call");
}
if (authpeer) {
authpeer = sip_unref_peer(authpeer, "sip_unref_peer, from handle_request_invite authpeer");
}
@@ -26004,24 +25846,17 @@ static void parse_oli(struct sip_request *req, struct ast_channel *chan)
* If this function is successful, only the transferer pvt lock will remain on return. Setting nounlock indicates
* to handle_request_do() that the pvt's owner it locked does not require an unlock.
*/
/* XXX XXX XXX XXX XXX XXX
* This function is COMPLETELY broken at the moment. It *will* crash if called
*/
static int local_attended_transfer(struct sip_pvt *transferer, struct sip_dual *current, struct sip_request *req, uint32_t seqno, int *nounlock)
static int local_attended_transfer(struct sip_pvt *transferer, struct ast_channel *transferer_chan, uint32_t seqno, int *nounlock)
{
struct sip_dual target; /* Chan 1: Call from tranferer to Asterisk */
/* Chan 2: Call from Asterisk to target */
int res = 0;
struct sip_pvt *targetcall_pvt;
struct ast_party_connected_line connected_to_transferee;
struct ast_party_connected_line connected_to_target;
char transferer_linkedid[32];
struct ast_channel *chans[2];
RAII_VAR(struct sip_pvt *, targetcall_pvt, NULL, ao2_cleanup);
RAII_VAR(struct ast_channel *, targetcall_chan, NULL, ao2_cleanup);
enum ast_transfer_result transfer_res;
/* Check if the call ID of the replaces header does exist locally */
if (!(targetcall_pvt = get_sip_pvt_byid_locked(transferer->refer->replaces_callid, transferer->refer->replaces_callid_totag,
transferer->refer->replaces_callid_fromtag))) {
if (get_sip_pvt_from_replaces(transferer->refer->replaces_callid,
transferer->refer->replaces_callid_totag,
transferer->refer->replaces_callid_fromtag,
&targetcall_pvt, &targetcall_chan)) {
if (transferer->refer->localtransfer) {
/* We did not find the refered call. Sorry, can't accept then */
/* Let's fake a response from someone else in order
@@ -26037,174 +25872,51 @@ static int local_attended_transfer(struct sip_pvt *transferer, struct sip_dual *
return 0;
}
/* Ok, we can accept this transfer */
append_history(transferer, "Xfer", "Refer accepted");
if (!targetcall_pvt->owner) { /* No active channel */
if (!targetcall_chan) { /* No active channel */
ast_debug(4, "SIP attended transfer: Error: No owner of target call\n");
/* Cancel transfer */
transmit_notify_with_sipfrag(transferer, seqno, "503 Service Unavailable", TRUE);
append_history(transferer, "Xfer", "Refer failed");
ast_clear_flag(&transferer->flags[0], SIP_GOTREFER);
transferer->refer->status = REFER_FAILED;
sip_pvt_unlock(targetcall_pvt);
if (targetcall_pvt)
ao2_t_ref(targetcall_pvt, -1, "Drop targetcall_pvt pointer");
return -1;
}
/* We have a channel, find the bridge */
target.chan1 = ast_channel_ref(targetcall_pvt->owner); /* Transferer to Asterisk */
target.chan2 = ast_bridged_channel(targetcall_pvt->owner); /* Asterisk to target */
if (target.chan2) {
ast_channel_ref(target.chan2);
}
if (!target.chan2 || !(ast_channel_state(target.chan2) == AST_STATE_UP || ast_channel_state(target.chan2) == AST_STATE_RINGING) ) {
/* Wrong state of new channel */
if (target.chan2)
ast_debug(4, "SIP attended transfer: Error: Wrong state of target call: %s\n", ast_state2str(ast_channel_state(target.chan2)));
else if (ast_channel_state(target.chan1) != AST_STATE_RING)
ast_debug(4, "SIP attended transfer: Error: No target channel\n");
else
ast_debug(4, "SIP attended transfer: Attempting transfer in ringing state\n");
}
/* Transfer */
if (sipdebug) {
if (current->chan2) /* We have two bridges */
ast_debug(4, "SIP attended transfer: trying to bridge %s and %s\n", ast_channel_name(target.chan1), ast_channel_name(current->chan2));
else /* One bridge, propably transfer of IVR/voicemail etc */
ast_debug(4, "SIP attended transfer: trying to make %s take over (masq) %s\n", ast_channel_name(target.chan1), ast_channel_name(current->chan1));
}
ast_set_flag(&transferer->flags[0], SIP_DEFER_BYE_ON_TRANSFER); /* Delay hangup */
ast_copy_string(transferer_linkedid, ast_channel_linkedid(transferer->owner), sizeof(transferer_linkedid));
sip_pvt_unlock(transferer);
ast_channel_unlock(transferer_chan);
*nounlock = 1;
/* Perform the transfer */
chans[0] = transferer->owner;
chans[1] = target.chan1;
ast_manager_event_multichan(EVENT_FLAG_CALL, "Transfer", 2, chans,
"TransferMethod: SIP\r\n"
"TransferType: Attended\r\n"
"Channel: %s\r\n"
"Uniqueid: %s\r\n"
"SIP-Callid: %s\r\n"
"TargetChannel: %s\r\n"
"TargetUniqueid: %s\r\n",
ast_channel_name(transferer->owner),
ast_channel_uniqueid(transferer->owner),
transferer->callid,
ast_channel_name(target.chan1),
ast_channel_uniqueid(target.chan1));
ast_party_connected_line_init(&connected_to_transferee);
ast_party_connected_line_init(&connected_to_target);
/* No need to lock current->chan1 here since it was locked in sipsock_read */
ast_party_connected_line_copy(&connected_to_transferee, ast_channel_connected(current->chan1));
/* No need to lock target.chan1 here since it was locked in get_sip_pvt_byid_locked */
ast_party_connected_line_copy(&connected_to_target, ast_channel_connected(target.chan1));
/* Reset any earlier private connected id representation */
ast_party_id_reset(&connected_to_transferee.priv);
ast_party_id_reset(&connected_to_target.priv);
connected_to_target.source = connected_to_transferee.source = AST_CONNECTED_LINE_UPDATE_SOURCE_TRANSFER;
res = attempt_transfer(current, &target);
if (res) {
/* Failed transfer */
transmit_notify_with_sipfrag(transferer, seqno, "486 Busy Here", TRUE);
append_history(transferer, "Xfer", "Refer failed");
ast_clear_flag(&transferer->flags[0], SIP_DEFER_BYE_ON_TRANSFER);
/* if transfer failed, go ahead and unlock targetcall_pvt and it's owner channel */
sip_pvt_unlock(targetcall_pvt);
ast_channel_unlock(target.chan1);
} else {
/* Transfer succeeded! */
const char *xfersound = pbx_builtin_getvar_helper(target.chan1, "ATTENDED_TRANSFER_COMPLETE_SOUND");
transfer_res = ast_bridge_transfer_attended(transferer_chan, targetcall_chan);
/* target.chan1 was locked in get_sip_pvt_byid_locked, do not unlock target.chan1 before this */
ast_cel_report_event(target.chan1, AST_CEL_ATTENDEDTRANSFER, NULL, transferer_linkedid, target.chan2);
sip_pvt_lock(transferer);
/* Tell transferer that we're done. */
switch (transfer_res) {
case AST_BRIDGE_TRANSFER_SUCCESS:
transferer->refer->status = REFER_200OK;
transmit_notify_with_sipfrag(transferer, seqno, "200 OK", TRUE);
append_history(transferer, "Xfer", "Refer succeeded");
transferer->refer->status = REFER_200OK;
if (target.chan2 && !ast_strlen_zero(xfersound) && ast_streamfile(target.chan2, xfersound, ast_channel_language(target.chan2)) >= 0) {
ast_waitstream(target.chan2, "");
return 1;
case AST_BRIDGE_TRANSFER_FAIL:
transferer->refer->status = REFER_FAILED;
transmit_notify_with_sipfrag(transferer, seqno, "500 Internal Server Error", TRUE);
append_history(transferer, "Xfer", "Refer failed (internal error)");
return -1;
case AST_BRIDGE_TRANSFER_INVALID:
transferer->refer->status = REFER_FAILED;
transmit_notify_with_sipfrag(transferer, seqno, "503 Service Unavailable", TRUE);
append_history(transferer, "Xfer", "Refer failed (invalid bridge state)");
return -1;
case AST_BRIDGE_TRANSFER_NOT_PERMITTED:
transferer->refer->status = REFER_FAILED;
transmit_notify_with_sipfrag(transferer, seqno, "403 Forbidden", TRUE);
append_history(transferer, "Xfer", "Refer failed (operation not permitted)");
return -1;
default:
break;
}
/* By forcing the masquerade, we know that target.chan1 and target.chan2 are bridged. We then
* can queue connected line updates where they need to go.
*
* before a masquerade, all channel and pvt locks must be unlocked. Any recursive
* channel locks held before this function invalidates channel container locking order.
* Since we are unlocking both the pvt (transferer) and its owner channel (current.chan1)
* it is possible for current.chan1 to be destroyed in the pbx thread. To prevent this
* we must give c a reference before any unlocking takes place.
*/
ast_channel_ref(current->chan1);
ast_channel_unlock(current->chan1); /* current.chan1 is p->owner before the masq, it was locked by socket_read()*/
ast_channel_unlock(target.chan1);
*nounlock = 1; /* we just unlocked the dialog's channel and have no plans of locking it again. */
sip_pvt_unlock(targetcall_pvt);
sip_pvt_unlock(transferer);
ast_do_masquerade(target.chan1);
ast_indicate(target.chan1, AST_CONTROL_UNHOLD);
if (target.chan2) {
ast_indicate(target.chan2, AST_CONTROL_UNHOLD);
}
if (current->chan2 && ast_channel_state(current->chan2) == AST_STATE_RING) {
ast_indicate(target.chan1, AST_CONTROL_RINGING);
}
if (target.chan2) {
ast_channel_queue_connected_line_update(target.chan1, &connected_to_transferee, NULL);
ast_channel_queue_connected_line_update(target.chan2, &connected_to_target, NULL);
} else {
/* Since target.chan1 isn't actually connected to another channel, there is no way for us
* to queue a frame so that its connected line status will be updated.
*
* Instead, we use the somewhat hackish approach of using a special control frame type that
* instructs ast_read to perform a specific action. In this case, the frame we queue tells
* ast_read to call the connected line interception macro configured for target.chan1.
*/
struct ast_control_read_action_payload *frame_payload;
int payload_size;
int frame_size;
unsigned char connected_line_data[1024];
payload_size = ast_connected_line_build_data(connected_line_data,
sizeof(connected_line_data), &connected_to_target, NULL);
frame_size = payload_size + sizeof(*frame_payload);
if (payload_size != -1) {
frame_payload = ast_alloca(frame_size);
frame_payload->payload_size = payload_size;
memcpy(frame_payload->payload, connected_line_data, payload_size);
frame_payload->action = AST_FRAME_READ_ACTION_CONNECTED_LINE_MACRO;
ast_queue_control_data(target.chan1, AST_CONTROL_READ_ACTION, frame_payload, frame_size);
}
/* In addition to queueing the read action frame so that target.chan1's connected line info
* will be updated, we also are going to queue a plain old connected line update on target.chan1. This
* way, either Dial or Queue can apply this connected line update to the outgoing ringing channel.
*/
ast_channel_queue_connected_line_update(target.chan1, &connected_to_transferee, NULL);
}
sip_pvt_lock(transferer); /* the transferer pvt is expected to remain locked on return */
ast_channel_unref(current->chan1);
}
/* at this point if the transfer is successful only the transferer pvt should be locked. */
ast_party_connected_line_free(&connected_to_target);
ast_party_connected_line_free(&connected_to_transferee);
ast_channel_unref(target.chan1);
if (target.chan2) {
ast_channel_unref(target.chan2);
}
if (targetcall_pvt)
ao2_t_ref(targetcall_pvt, -1, "drop targetcall_pvt");
return 1;
}
@@ -26438,7 +26150,7 @@ static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, uint
/* Attended transfer: Find all call legs and bridge transferee with target*/
if (p->refer->attendedtransfer) {
/* both p and p->owner _MUST_ be locked while calling local_attended_transfer */
if ((res = local_attended_transfer(p, NULL, req, seqno, nounlock))) {
if ((res = local_attended_transfer(p, transferer, seqno, nounlock))) {
ast_clear_flag(&p->flags[0], SIP_GOTREFER);
return res;
}

14
channels/sip/include/sip.h
View File

@@ -834,16 +834,6 @@ struct sip_request {
*/
#define REQ_OFFSET_TO_STR(req,offset) (ast_str_buffer((req)->data) + ((req)->offset))
/*! \brief structure used in transfers */
struct sip_dual {
struct ast_channel *chan1; /*!< First channel involved */
struct ast_channel *chan2; /*!< Second channel involved */
struct sip_request req; /*!< Request that caused the transfer (REFER) */
uint32_t seqno; /*!< Sequence number */
char *park_exten;
char *park_context;
};
/*! \brief Parameters to the transmit_invite function */
struct sip_invite_param {
int addsipheaders; /*!< Add extra SIP headers */
@@ -935,10 +925,6 @@ struct sip_refer {
AST_STRING_FIELD(replaces_callid_totag); /*!< Replace info: to-tag */
AST_STRING_FIELD(replaces_callid_fromtag); /*!< Replace info: from-tag */
);
struct sip_pvt *refer_call; /*!< Call we are referring. This is just a reference to a
* dialog owned by someone else, so we should not destroy
* it when the sip_refer object goes.
*/
int attendedtransfer; /*!< Attended or blind transfer? */
int localtransfer; /*!< Transfer to local domain? */
enum referstatus status; /*!< REFER status */