Home page | Mailing list | Docs

Last updated: Sat Feb 3 05:01:52 2007

Asterisk developer's documentation :: Codename Pineapple

---

sip3_refer.c File Reference

---

Detailed Description

Various SIP transfer/refer functions Version 3 of chan_sip.

Author:

Mark Spencer <markster@digium.com>

Olle E. Johansson <oej@edvina.net> (all the chan_sip3 changes)

See Also:

• CREDITS

Definition in file sip3_refer.c.

#include "asterisk.h"
#include <stdio.h>
#include <ctype.h>
#include <string.h>
#include <unistd.h>
#include <sys/socket.h>
#include <sys/ioctl.h>
#include <net/if.h>
#include <errno.h>
#include <stdlib.h>
#include <fcntl.h>
#include <netdb.h>
#include <signal.h>
#include <sys/signal.h>
#include <netinet/in.h>
#include <netinet/in_systm.h>
#include <arpa/inet.h>
#include <netinet/ip.h>
#include <regex.h>
#include "asterisk/lock.h"
#include "asterisk/channel.h"
#include "asterisk/config.h"
#include "asterisk/logger.h"
#include "asterisk/module.h"
#include "asterisk/pbx.h"
#include "asterisk/options.h"
#include "asterisk/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"
#include "asterisk/cli.h"
#include "asterisk/app.h"
#include "asterisk/musiconhold.h"
#include "asterisk/dsp.h"
#include "asterisk/features.h"
#include "asterisk/srv.h"
#include "asterisk/astdb.h"
#include "asterisk/causes.h"
#include "asterisk/utils.h"
#include "asterisk/file.h"
#include "asterisk/astobj.h"
#include "asterisk/dnsmgr.h"
#include "asterisk/devicestate.h"
#include "asterisk/linkedlists.h"
#include "asterisk/stringfields.h"
#include "asterisk/monitor.h"
#include "asterisk/localtime.h"
#include "asterisk/abstract_jb.h"
#include "asterisk/compiler.h"
#include "sip3.h"
#include "sip3funcs.h"

Include dependency graph for sip3_refer.c:

Go to the source code of this file.

Data Structures
struct	c_referstatusstring
	Table to convert from REFER status variable to string. More...
Functions
static int	attempt_transfer (struct sip_dual *transferer, struct sip_dual *target)
	Attempt transfer of SIP call This fix for attended transfers on a local PBX.
static int	get_refer_info (struct sip_dialog *transferer, struct sip_request *outgoing_req)
	Call transfer support (the REFER method) Extracts Refer headers into pvt dialog structure.
int	handle_request_refer (struct sip_dialog *p, struct sip_request *req, int debug, int seqno, int *nounlock)
static int	local_attended_transfer (struct sip_dialog *transferer, struct sip_dual *current, struct sip_request *req, int seqno)
	Find all call legs and bridge transferee with target called from handle_request_refer.
const char *	referstatus2str (enum referstatus rstatus)
	Convert transfer status to string.
static int	sip_park (struct ast_channel *chan1, struct ast_channel *chan2, struct sip_request *req, int seqno)
	Park a call using the subsystem in res_features.c This is executed in a separate thread.
static void *	sip_park_thread (void *stuff)
	Park SIP call support function Starts in a new thread, then parks the call XXX Should we add a wait period after streaming audio and before hangup?? Sometimes the audio can't be heard before hangup.
int	sip_refer_allocate (struct sip_dialog *p)
	Allocate SIP refer structure.
GNURK int	transmit_notify_with_sipfrag (struct sip_dialog *p, int cseq, char *message, int terminate)
	Notify a transferring party of the status of transfer.
Variables
static const struct c_referstatusstring	referstatusstrings []
	Table to convert from REFER status variable to string.

---

Function Documentation

static int attempt_transfer (  struct sip_dual *  transferer, struct sip_dual *  target )  [static]

Attempt transfer of SIP call This fix for attended transfers on a local PBX. Definition at line 451 of file sip3_refer.c. References ast_channel::_state, ast_cdr_append(), ast_channel_masquerade(), ast_log(), ast_quiet_chan(), AST_SOFTHANGUP_DEV, ast_softhangup_nolock(), ast_state2str(), sip_dual::chan1, sip_dual::chan2, LOG_DEBUG, LOG_NOTICE, LOG_WARNING, and option_debug. { 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 */ if (option_debug > 3) { ast_log(LOG_DEBUG, "Sip transfer:--------------------\n"); if (transferer->chan1) ast_log(LOG_DEBUG, "-- Transferer to PBX channel: %s State %s\n", transferer->chan1->name, ast_state2str(transferer->chan1->_state)); else ast_log(LOG_DEBUG, "-- No transferer first channel - odd??? \n"); if (target->chan1) ast_log(LOG_DEBUG, "-- Transferer to PBX second channel (target): %s State %s\n", target->chan1->name, ast_state2str(target->chan1->_state)); else ast_log(LOG_DEBUG, "-- No target first channel ---\n"); if (transferer->chan2) ast_log(LOG_DEBUG, "-- Bridged call to transferee: %s State %s\n", transferer->chan2->name, ast_state2str(transferer->chan2->_state)); else ast_log(LOG_DEBUG, "-- No bridged call to transferee\n"); if (target->chan2) ast_log(LOG_DEBUG, "-- Bridged call to transfer target: %s State %s\n", target->chan2 ? target->chan2->name : "<none>", target->chan2 ? ast_state2str(target->chan2->_state) : "(none)"); else ast_log(LOG_DEBUG, "-- No target second channel ---\n"); ast_log(LOG_DEBUG, "-- 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 */ if (option_debug > 2) ast_log(LOG_DEBUG, "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 */ if (option_debug > 2) ast_log(LOG_DEBUG, "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); /* Fix CDRs so they're attached to the remaining channel */ if (peera->cdr && peerb->cdr) peerb->cdr = ast_cdr_append(peerb->cdr, peera->cdr); else if (peera->cdr) peerb->cdr = peera->cdr; peera->cdr = NULL; if (peerb->cdr && peerc->cdr) peerb->cdr = ast_cdr_append(peerb->cdr, peerc->cdr); else if (peerc->cdr) peerb->cdr = peerc->cdr; peerc->cdr = NULL; if (option_debug > 3) ast_log(LOG_DEBUG, "SIP transfer: trying to masquerade %s into %s\n", peerc->name, peerb->name); if (ast_channel_masquerade(peerb, peerc)) { ast_log(LOG_WARNING, "Failed to masquerade %s into %s\n", peerb->name, peerc->name); res = -1; } else ast_log(LOG_DEBUG, "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; }

static int get_refer_info (  struct sip_dialog *  transferer, struct sip_request *  outgoing_req )  [static]

Call transfer support (the REFER method) Extracts Refer headers into pvt dialog structure. Definition at line 203 of file sip3_refer.c. References ast_exists_extension(), ast_log(), ast_strdupa, ast_strlen_zero(), ast_uri_decode(), ast_verbose(), sip_refer::attendedtransfer, sip_globals::default_context, get_header(), get_in_brackets(), global, sip_dialog::initreq, LOG_DEBUG, LOG_WARNING, ast_channel::macrocontext, option_debug, sip_dialog::owner, pbx_builtin_getvar_helper(), sip_dialog::refer, sip_refer::refer_to, sip_refer::refer_to_context, sip_refer::refer_to_domain, sip_refer::refer_to_urioption, sip_refer::referred_by, sip_refer::referred_by_name, sip_refer::replaces_callid, sip_refer::replaces_callid_fromtag, sip_refer::replaces_callid_totag, S_OR, sip_debug_test_pvt(), and strcasestr(). { const char *p_referred_by = NULL; char *h_refer_to = NULL; char *h_referred_by = NULL; char *refer_to; const char *p_refer_to; char *referred_by_uri = NULL; char *ptr; struct sip_request *req = NULL; const char *transfer_context = NULL; struct sip_refer *referdata; req = outgoing_req; referdata = transferer->refer; if (!req) req = transferer->initreq; p_refer_to = get_header(req, "Refer-To"); if (ast_strlen_zero(p_refer_to)) { ast_log(LOG_WARNING, "Refer-To Header missing. Skipping transfer.\n"); return -2; /* Syntax error */ } h_refer_to = ast_strdupa(p_refer_to); refer_to = get_in_brackets(h_refer_to); ast_uri_decode(refer_to); if (strncasecmp(refer_to, "sip:", 4)) { ast_log(LOG_WARNING, "Can't transfer to non-sip: URI. (Refer-to: %s)?\n", refer_to); return -3; } refer_to += 4; /* Skip sip: */ /* Get referred by header if it exists */ p_referred_by = get_header(req, "Referred-By"); if (!ast_strlen_zero(p_referred_by)) { char *lessthan; h_referred_by = ast_strdupa(p_referred_by); ast_uri_decode(h_referred_by); /* Store referrer's caller ID name */ ast_copy_string(referdata->referred_by_name, h_referred_by, sizeof(referdata->referred_by_name)); if ((lessthan = strchr(referdata->referred_by_name, '<'))) { *(lessthan - 1) = '\0'; /* Space */ } referred_by_uri = get_in_brackets(h_referred_by); if(strncasecmp(referred_by_uri, "sip:", 4)) { ast_log(LOG_WARNING, "Huh? Not a sip: header (Referred-by: %s). Skipping.\n", referred_by_uri); referred_by_uri = (char *) NULL; } else { referred_by_uri += 4; /* Skip sip: */ } } /* Check for arguments in the refer_to header */ if ((ptr = strchr(refer_to, '?'))) { /* Search for arguments */ *ptr++ = '\0'; if (!strncasecmp(ptr, "REPLACES=", 9)) { char *to = NULL, *from = NULL; /* This is an attended transfer */ referdata->attendedtransfer = 1; strncpy(referdata->replaces_callid, ptr+9, sizeof(referdata->replaces_callid)); ast_uri_decode(referdata->replaces_callid); if ((ptr = strchr(referdata->replaces_callid, ';'))) /* Find options */ { *ptr++ = '\0'; } if (ptr) { /* Find the different tags before we destroy the string */ to = strcasestr(ptr, "to-tag="); from = strcasestr(ptr, "from-tag="); } /* Grab the to header */ if (to) { ptr = to + 7; if ((to = strchr(ptr, '&'))) *to = '\0'; if ((to = strchr(ptr, ';'))) *to = '\0'; ast_copy_string(referdata->replaces_callid_totag, ptr, sizeof(referdata->replaces_callid_totag)); } if (from) { ptr = from + 9; if ((to = strchr(ptr, '&'))) *to = '\0'; if ((to = strchr(ptr, ';'))) *to = '\0'; ast_copy_string(referdata->replaces_callid_fromtag, ptr, sizeof(referdata->replaces_callid_fromtag)); } if (option_debug > 1) ast_log(LOG_DEBUG,"Attended transfer: Will use Replace-Call-ID : %s F-tag: %s T-tag: %s\n", referdata->replaces_callid, referdata->replaces_callid_fromtag ? referdata->replaces_callid_fromtag : "<none>", referdata->replaces_callid_totag ? referdata->replaces_callid_totag : "<none>" ); } } if ((ptr = strchr(refer_to, '@'))) { /* Separate domain */ char *urioption; *ptr++ = '\0'; if ((urioption = strchr(ptr, ';'))) *urioption++ = '\0'; /* Save the domain for the dial plan */ strncpy(referdata->refer_to_domain, ptr, sizeof(referdata->refer_to_domain)); if (urioption) strncpy(referdata->refer_to_urioption, urioption, sizeof(referdata->refer_to_urioption)); } if ((ptr = strchr(refer_to, ';'))) /* Remove options */ *ptr = '\0'; ast_copy_string(referdata->refer_to, refer_to, sizeof(referdata->refer_to)); if (referred_by_uri) { if ((ptr = strchr(referred_by_uri, ';'))) /* Remove options */ *ptr = '\0'; ast_copy_string(referdata->referred_by, referred_by_uri, sizeof(referdata->referred_by)); } else { referdata->referred_by[0] = '\0'; } /* Determine transfer context */ if (transferer->owner) /* Mimic behaviour in res_features.c */ transfer_context = pbx_builtin_getvar_helper(transferer->owner, "TRANSFER_CONTEXT"); /* By default, use the context in the channel sending the REFER */ if (ast_strlen_zero(transfer_context)) { transfer_context = S_OR(transferer->owner->macrocontext, S_OR(transferer->context, global.default_context)); } strncpy(referdata->refer_to_context, transfer_context, sizeof(referdata->refer_to_context)); /* Either an existing extension or the parking extension */ if (ast_exists_extension(NULL, transfer_context, refer_to, 1, NULL) ) { if (sip_debug_test_pvt(transferer)) { ast_verbose("SIP transfer to extension %s@%s by %s\n", refer_to, transfer_context, referred_by_uri); } /* We are ready to transfer to the extension */ return 0; } if (sip_debug_test_pvt(transferer)) ast_verbose("Failed SIP Transfer to non-existing extension %s in context %s\n n", refer_to, transfer_context); /* Failure, we can't find this extension */ return -1; }

int handle_request_refer (  struct sip_dialog *  p, struct sip_request *  req, int  debug, int  seqno, int *  nounlock )

Definition at line 708 of file sip3_refer.c. References sip_globals::allow_external_domains, sip_dialog::allowtransfer, append_history, ast_async_goto(), ast_bridged_channel(), AST_CAUSE_NORMAL_CLEARING, ast_channel_unlock, ast_clear_flag, AST_CONTROL_UNHOLD, ast_log(), ast_parking_ext(), ast_queue_control(), ast_set_flag, ast_string_field_set, ast_strlen_zero(), ast_test_flag, ast_uri_encode(), ast_verbose(), sip_refer::attendedtransfer, sip_dual::chan1, sip_dual::chan2, check_sip_domain(), context, sip_globals::default_context, domains_configured(), FALSE, sip_dialog::flags, get_refer_info(), global, ast_channel::hangupcause, local_attended_transfer(), sip_refer::localtransfer, LOG_DEBUG, option_debug, sip_dialog::owner, pbx_builtin_setvar_helper(), sip_dialog::refer, REFER_200OK, REFER_FAILED, REFER_SENT, sip_refer::refer_to, sip_refer::refer_to_context, sip_refer::refer_to_domain, sip_refer::referred_by, sip_refer::replaces_callid, sip_refer::replaces_callid_fromtag, sip_refer::replaces_callid_totag, sip_dual::req, SIP_ALREADYGONE, SIP_DEFER_BYE_ON_TRANSFER, SIP_GOTREFER, SIP_NEEDDESTROY, SIP_OUTGOING, sip_park(), SIP_PKT_DEBUG, SIP_PKT_IGNORE, sip_refer_allocate(), sipdebug, sip_refer::status, TRANSFER_CLOSED, transmit_notify_with_sipfrag(), transmit_response(), and TRUE. { struct sip_dual current; /* Chan1: Call between asterisk and transferer */ /* Chan2: Call between asterisk and transferee */ int res = 0; if (ast_test_flag(req, SIP_PKT_DEBUG)) ast_verbose("Call %s got a SIP call transfer from %s: (REFER)!\n", p->callid, ast_test_flag(&p->flags[0], SIP_OUTGOING) ? "callee" : "caller"); if (!p->owner) { /* This is a REFER outside of an existing SIP dialog */ /* We can't handle that, so decline it */ if (option_debug > 2) ast_log(LOG_DEBUG, "Call %s: Declined REFER, outside of dialog...\n", p->callid); transmit_response(p, "603 Declined (No dialog)", req); if (!ast_test_flag(req, SIP_PKT_IGNORE)) { append_history(p, "Xfer", "Refer failed. Outside of dialog."); ast_set_flag(&p->flags[0], SIP_ALREADYGONE); ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); } return 0; } /* Check if transfer is allowed from this device */ if (p->allowtransfer == TRANSFER_CLOSED ) { /* Transfer not allowed, decline */ transmit_response(p, "603 Declined (policy)", req); append_history(p, "Xfer", "Refer failed. Allowtransfer == closed."); /* Do not destroy SIP session */ return 0; } if(!ast_test_flag(req, SIP_PKT_IGNORE) && ast_test_flag(&p->flags[0], SIP_GOTREFER)) { /* Already have a pending REFER */ transmit_response(p, "491 Request pending", req); append_history(p, "Xfer", "Refer failed. Request pending."); return 0; } /* Allocate memory for call transfer data */ if (!p->refer && !sip_refer_allocate(p)) { transmit_response(p, "500 Internal Server Error", req); append_history(p, "Xfer", "Refer failed. Memory allocation error."); return -3; } res = get_refer_info(p, req); /* Extract headers */ p->refer->status = REFER_SENT; if (res != 0) { switch (res) { case -2: /* Syntax error */ transmit_response(p, "400 Bad Request (Refer-to missing)", req); append_history(p, "Xfer", "Refer failed. Refer-to missing."); if (ast_test_flag(req, SIP_PKT_DEBUG) && option_debug) ast_log(LOG_DEBUG, "SIP transfer to black hole can't be handled (no refer-to: )\n"); break; case -3: transmit_response(p, "603 Declined (Non sip: uri)", req); append_history(p, "Xfer", "Refer failed. Non SIP uri"); if (ast_test_flag(req, SIP_PKT_DEBUG) && option_debug) ast_log(LOG_DEBUG, "SIP transfer to non-SIP uri denied\n"); break; default: /* Refer-to extension not found, fake a failed transfer */ transmit_response(p, "202 Accepted", req); append_history(p, "Xfer", "Refer failed. Bad extension."); transmit_notify_with_sipfrag(p, seqno, "404 Not found", TRUE); ast_clear_flag(&p->flags[0], SIP_GOTREFER); if (ast_test_flag(req, SIP_PKT_DEBUG) && option_debug) ast_log(LOG_DEBUG, "SIP transfer to bad extension: %s\n", p->refer->refer_to); break; } return 0; } if (ast_strlen_zero(p->context)) ast_string_field_set(p, context, global.default_context); /* If we do not support SIP domains, all transfers are local */ if (global.allow_external_domains && check_sip_domain(p->refer->refer_to_domain, NULL, 0)) { p->refer->localtransfer = 1; if (sipdebug && option_debug > 2) ast_log(LOG_DEBUG, "This SIP transfer is local : %s\n", p->refer->refer_to_domain); } else if (domains_configured()) { /* This PBX don't bother with SIP domains, so all transfers are local */ p->refer->localtransfer = 1; } else if (sipdebug && option_debug > 2) ast_log(LOG_DEBUG, "This SIP transfer is to a remote SIP extension (remote domain %s)\n", p->refer->refer_to_domain); /* Is this a repeat of a current request? Ignore it */ /* Don't know what else to do right now. */ if (ast_test_flag(req, SIP_PKT_IGNORE)) return res; /* If this is a blind transfer, we have the following channels to work with: - chan1, chan2: The current call between transferer and transferee (2 channels) - target_channel: A new call from the transferee to the target (1 channel) We need to stay tuned to what happens in order to be able to bring back the call to the transferer */ /* If this is a attended transfer, we should have all call legs within reach: - chan1, chan2: The call between the transferer and transferee (2 channels) - target_channel, targetcall_pvt: The call between the transferer and the target (2 channels) We want to bridge chan2 with targetcall_pvt! The replaces call id in the refer message points to the call leg between Asterisk and the transferer. So we need to connect the target and the transferee channel and hangup the two other channels silently If the target is non-local, the call ID could be on a remote machine and we need to send an INVITE with replaces to the target. We basically handle this as a blind transfer and let the sip_call function catch that we need replaces header in the INVITE. */ /* Get the transferer's channel */ current.chan1 = p->owner; /* Find the other part of the bridge (2) - transferee */ current.chan2 = ast_bridged_channel(current.chan1); if (sipdebug && option_debug > 2) ast_log(LOG_DEBUG, "SIP %s transfer: Transferer channel %s, transferee channel %s\n", p->refer->attendedtransfer ? "attended" : "blind", current.chan1->name, current.chan2 ? current.chan2->name : "<none>"); if (!current.chan2 && !p->refer->attendedtransfer) { /* No bridged channel, propably IVR or echo or similar... */ /* Guess we should masquerade or something here */ /* Until we figure it out, refuse transfer of such calls */ if (sipdebug && option_debug > 2) ast_log(LOG_DEBUG,"Refused SIP transfer on non-bridged channel.\n"); p->refer->status = REFER_FAILED; append_history(p, "Xfer", "Refer failed. Non-bridged channel."); transmit_response(p, "603 Declined", req); return -1; } if (current.chan2) { if (sipdebug && option_debug > 3) ast_log(LOG_DEBUG, "Got SIP transfer, applying to bridged peer '%s'\n", current.chan2->name); ast_queue_control(current.chan1, AST_CONTROL_UNHOLD); } ast_set_flag(&p->flags[0], SIP_GOTREFER); /* Attended transfer: Find all call legs and bridge transferee with target*/ if (p->refer->attendedtransfer) { if ((res = local_attended_transfer(p, &current, req, seqno))) return res; /* We're done with the transfer */ /* Fall through for remote transfers that we did not find locally */ if (sipdebug && option_debug > 3) ast_log(LOG_DEBUG, "SIP attended transfer: Still not our call - generating INVITE with replaces\n"); /* Fallthrough if we can't find the call leg internally */ } /* Parking a call */ if (p->refer->localtransfer && !strcmp(p->refer->refer_to, ast_parking_ext())) { /* Must release c's lock now, because it will not longer be accessible after the transfer! */ *nounlock = 1; ast_channel_unlock(current.chan1); current.req = req; ast_clear_flag(&p->flags[0], SIP_GOTREFER); p->refer->status = REFER_200OK; append_history(p, "Xfer", "REFER to call parking."); if (sipdebug && option_debug > 3) ast_log(LOG_DEBUG, "SIP transfer to parking: trying to park %s. Parked by %s\n", current.chan2->name, current.chan1->name); sip_park(current.chan2, current.chan1, req, seqno); return res; } /* Blind transfers and remote attended xfers */ transmit_response(p, "202 Accepted", req); if (current.chan1 && current.chan2) { if (option_debug > 2) ast_log(LOG_DEBUG, "chan1->name: %s\n", current.chan1->name); pbx_builtin_setvar_helper(current.chan1, "BLINDTRANSFER", current.chan2->name); } if (current.chan2) { pbx_builtin_setvar_helper(current.chan2, "BLINDTRANSFER", current.chan1->name); pbx_builtin_setvar_helper(current.chan2, "SIPDOMAIN", p->refer->refer_to_domain); pbx_builtin_setvar_helper(current.chan2, "SIPTRANSFER", "yes"); /* One for the new channel */ pbx_builtin_setvar_helper(current.chan2, "_SIPTRANSFER", "yes"); if (p->refer->referred_by) pbx_builtin_setvar_helper(current.chan2, "_SIPTRANSFER_REFERER", p->refer->referred_by); if (p->refer->referred_by) /* Attended transfer to remote host, prepare headers for the INVITE */ pbx_builtin_setvar_helper(current.chan2, "_SIPTRANSFER_REFERER", p->refer->referred_by); } /* Generate an URI-encoded string */ if (p->refer->replaces_callid && !ast_strlen_zero(p->refer->replaces_callid)) { char tempheader[BUFSIZ]; char tempheader2[BUFSIZ]; snprintf(tempheader, sizeof(tempheader), "%s%s%s%s%s", p->refer->replaces_callid, p->refer->replaces_callid_totag ? ";to-tag=" : "", p->refer->replaces_callid_totag, p->refer->replaces_callid_fromtag ? ";from-tag=" : "", p->refer->replaces_callid_fromtag); /* Convert it to URL encoding, also convert reserved strings */ ast_uri_encode(tempheader, tempheader2, sizeof(tempheader2), 1); if (current.chan2) pbx_builtin_setvar_helper(current.chan2, "_SIPTRANSFER_REPLACES", tempheader2); } /* Must release lock now, because it will not longer be accessible after the transfer! */ *nounlock = 1; ast_channel_unlock(current.chan1); ast_channel_unlock(current.chan2); /* Connect the call */ /* FAKE ringing if not attended transfer */ if (!p->refer->attendedtransfer) transmit_notify_with_sipfrag(p, seqno, "183 Ringing", FALSE); /* For blind transfer, this will lead to a new call */ /* For attended transfer to remote host, this will lead to a new SIP call with a replaces header, if the dial plan allows it */ if (!current.chan2) { /* We have no bridge, so we're talking with Asterisk somehow */ /* We need to masquerade this call */ /* What to do to fix this situation: * Set up the new call in a new channel * Let the new channel masq into this channel Please add that code here :-) */ transmit_response(p, "202 Accepted", req); p->refer->status = REFER_FAILED; transmit_notify_with_sipfrag(p, seqno, "503 Service Unavailable (can't handle one-legged xfers)", TRUE); ast_clear_flag(&p->flags[0], SIP_GOTREFER); append_history(p, "Xfer", "Refer failed (only bridged calls)."); return -1; } ast_set_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER); /* Delay hangup */ /* For blind transfers, move the call to the new extensions. For attended transfers on multiple servers - generate an INVITE with Replaces. Either way, let the dial plan decided */ res = ast_async_goto(current.chan2, p->refer->refer_to_context, p->refer->refer_to, 1); if (!res) { /* Success - we have a new channel */ if (option_debug > 2) ast_log(LOG_DEBUG, "%s transfer succeeded. Telling transferer.\n", p->refer->attendedtransfer? "Attended" : "Blind"); transmit_notify_with_sipfrag(p, seqno, "200 Ok", TRUE); if (p->refer->localtransfer) p->refer->status = REFER_200OK; if (p->owner) p->owner->hangupcause = AST_CAUSE_NORMAL_CLEARING; append_history(p, "Xfer", "Refer succeeded."); ast_clear_flag(&p->flags[0], SIP_GOTREFER); /* Do not hangup call, the other side do that when we say 200 OK */ /* We could possibly implement a timer here, auto congestion */ res = 0; } else { ast_clear_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER); /* Don't delay hangup */ if (option_debug > 2) ast_log(LOG_DEBUG, "%s transfer failed. Resuming original call.\n", p->refer->attendedtransfer? "Attended" : "Blind"); append_history(p, "Xfer", "Refer failed."); /* Failure of some kind */ p->refer->status = REFER_FAILED; transmit_notify_with_sipfrag(p, seqno, "503 Service Unavailable", TRUE); ast_clear_flag(&p->flags[0], SIP_GOTREFER); res = -1; } return res; }

static int local_attended_transfer (  struct sip_dialog *  transferer, struct sip_dual *  current, struct sip_request *  req, int  seqno )  [static]

Find all call legs and bridge transferee with target called from handle_request_refer. Definition at line 539 of file sip3_refer.c. References ast_channel::_state, append_history, ast_bridged_channel(), ast_channel_unlock, ast_clear_flag, ast_hangup(), ast_log(), ast_set_flag, ast_state2str(), AST_STATE_RING, AST_STATE_RINGING, AST_STATE_UP, attempt_transfer(), sip_dual::chan1, sip_dual::chan2, dialog_lock(), error(), FALSE, sip_dialog::flags, get_sip_dialog_byid_locked(), sip_refer::localtransfer, LOG_DEBUG, option_debug, sip_dialog::owner, sip_dialog::refer, REFER_200OK, REFER_FAILED, sip_refer::replaces_callid, sip_refer::replaces_callid_fromtag, sip_refer::replaces_callid_totag, SIP_DEFER_BYE_ON_TRANSFER, SIP_GOTREFER, sipdebug, sip_refer::status, transmit_notify_with_sipfrag(), transmit_response(), and TRUE. { struct sip_dual target; /* Chan 1: Call from tranferer to Asterisk */ /* Chan 2: Call from Asterisk to target */ int res = 0; struct sip_dialog *targetcall_pvt; int error = 0; /* Check if the call ID of the replaces header does exist locally */ if (!(targetcall_pvt = get_sip_dialog_byid_locked(transferer->refer->replaces_callid, transferer->refer->replaces_callid_totag, transferer->refer->replaces_callid_fromtag))) { if (transferer->refer->localtransfer) { /* We did not find the refered call. Sorry, can't accept then */ transmit_response(transferer, "202 Accepted", req); /* Let's fake a response from someone else in order to follow the standard */ transmit_notify_with_sipfrag(transferer, seqno, "481 Call leg/transaction does not exist", TRUE); append_history(transferer, "Xfer", "Refer failed"); ast_clear_flag(&transferer->flags[0], SIP_GOTREFER); transferer->refer->status = REFER_FAILED; return -1; } /* Fall through for remote transfers that we did not find locally */ if (option_debug > 2) ast_log(LOG_DEBUG, "SIP attended transfer: Not our call - generating INVITE with replaces\n"); return 0; } /* Ok, we can accept this transfer */ transmit_response(transferer, "202 Accepted", req); append_history(transferer, "Xfer", "Refer accepted"); if (!targetcall_pvt->owner) { /* No active channel */ if (option_debug > 3) ast_log(LOG_DEBUG, "SIP attended transfer: Error: No owner of target call\n"); error = 1; } /* We have a channel, find the bridge */ target.chan1 = targetcall_pvt->owner; /* Transferer to Asterisk */ if (!error) { target.chan2 = ast_bridged_channel(targetcall_pvt->owner); /* Asterisk to target */ if (!target.chan2 || !(target.chan2->_state == AST_STATE_UP || target.chan2->_state == AST_STATE_RINGING) ) { /* Wrong state of new channel */ if (option_debug > 3) { if (target.chan2) ast_log(LOG_DEBUG, "SIP attended transfer: Error: Wrong state of target call: %s\n", ast_state2str(target.chan2->_state)); else if (target.chan1->_state != AST_STATE_RING) ast_log(LOG_DEBUG, "SIP attended transfer: Error: No target channel\n"); else ast_log(LOG_DEBUG, "SIP attended transfer: Attempting transfer in ringing state\n"); } if (target.chan1->_state != AST_STATE_RING) error = 1; } } if (error) { /* 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; dialog_lock(targetcall_pvt, FALSE); ast_channel_unlock(current->chan1); ast_channel_unlock(target.chan1); return -1; } /* Transfer */ if (option_debug > 3 && sipdebug) { if (current->chan2) /* We have two bridges */ ast_log(LOG_DEBUG, "SIP attended transfer: trying to bridge %s and %s\n", target.chan1->name, current->chan2->name); else /* One bridge, propably transfer of IVR/voicemail etc */ ast_log(LOG_DEBUG, "SIP attended transfer: trying to make %s take over (masq) %s\n", target.chan1->name, current->chan1->name); } ast_set_flag(&transferer->flags[0], SIP_DEFER_BYE_ON_TRANSFER); /* Delay hangup */ /* Perform the transfer */ res = attempt_transfer(current, &target); dialog_lock(targetcall_pvt, FALSE); if (res) { /* Failed transfer */ /* Could find better message, but they will get the point */ transmit_notify_with_sipfrag(transferer, seqno, "486 Busy", TRUE); append_history(transferer, "Xfer", "Refer failed"); if (targetcall_pvt->owner) ast_channel_unlock(targetcall_pvt->owner); /* Right now, we have to hangup, sorry. Bridge is destroyed */ ast_hangup(transferer->owner); } else { /* Transfer succeeded! */ /* Tell transferer that we're done. */ transmit_notify_with_sipfrag(transferer, seqno, "200 OK", TRUE); append_history(transferer, "Xfer", "Refer succeeded"); transferer->refer->status = REFER_200OK; if (targetcall_pvt->owner) { if (option_debug) ast_log(LOG_DEBUG, "SIP attended transfer: Unlocking channel %s\n", targetcall_pvt->owner->name); ast_channel_unlock(targetcall_pvt->owner); } } return 1; }

const char* referstatus2str (  enum referstatus  rstatus  )

Convert transfer status to string. sip3_refer.c Definition at line 109 of file sip3_refer.c. References referstatusstrings. { int i = (sizeof(referstatusstrings) / sizeof(referstatusstrings[0])); int x; for (x = 0; x < i; x++) { if (referstatusstrings[x].status == rstatus) return (char *) referstatusstrings[x].text; } return ""; }

static int sip_park (  struct ast_channel *  chan1, struct ast_channel *  chan2, struct sip_request *  req, int  seqno )  [static]

Park a call using the subsystem in res_features.c This is executed in a separate thread. Definition at line 360 of file sip3_refer.c. References ast_calloc, AST_CAUSE_SWITCH_CONGESTION, ast_channel_alloc(), ast_channel_lock, ast_channel_masquerade(), ast_channel_unlock, ast_do_masquerade(), ast_hangup(), ast_log(), ast_pthread_create_background, AST_STATE_DOWN, ast_channel::context, ast_channel::exten, free, ast_channel::hangupcause, LOG_DEBUG, LOG_WARNING, option_debug, ast_channel::priority, ast_channel::readformat, sip_park_thread(), and ast_channel::writeformat. { struct sip_dual *d; struct ast_channel *transferee, *transferer; /* Chan2m: The transferer, chan1m: The transferee */ pthread_t th; transferee = ast_channel_alloc(0, AST_STATE_DOWN, 0, 0, "Parking/%s", chan1->name); transferer = ast_channel_alloc(0, AST_STATE_DOWN, 0, 0, "SIPPeer/%s", chan2->name); if ((!transferer) || (!transferee)) { if (transferee) { transferee->hangupcause = AST_CAUSE_SWITCH_CONGESTION; ast_hangup(transferee); } if (transferer) { transferer->hangupcause = AST_CAUSE_SWITCH_CONGESTION; ast_hangup(transferer); } return -1; } /* Make formats okay */ transferee->readformat = chan1->readformat; transferee->writeformat = chan1->writeformat; /* Prepare for taking over the channel */ ast_channel_masquerade(transferee, chan1); /* Setup the extensions and such */ ast_copy_string(transferee->context, chan1->context, sizeof(transferee->context)); ast_copy_string(transferee->exten, chan1->exten, sizeof(transferee->exten)); transferee->priority = chan1->priority; /* We make a clone of the peer channel too, so we can play back the announcement */ /* Make formats okay */ transferer->readformat = chan2->readformat; transferer->writeformat = chan2->writeformat; /* Prepare for taking over the channel */ ast_channel_masquerade(transferer, chan2); /* Setup the extensions and such */ ast_copy_string(transferer->context, chan2->context, sizeof(transferer->context)); ast_copy_string(transferer->exten, chan2->exten, sizeof(transferer->exten)); transferer->priority = chan2->priority; ast_channel_lock(transferer); if (ast_do_masquerade(transferer)) { ast_log(LOG_WARNING, "Masquerade failed :(\n"); ast_channel_unlock(transferer); transferer->hangupcause = AST_CAUSE_SWITCH_CONGESTION; ast_hangup(transferer); return -1; } ast_channel_unlock(transferer); if (!transferer || !transferee) { if (!transferer) { if (option_debug) ast_log(LOG_DEBUG, "No transferer channel, giving up parking\n"); } if (!transferee) { if (option_debug) ast_log(LOG_DEBUG, "No transferee channel, giving up parking\n"); } return -1; } if ((d = ast_calloc(1, sizeof(*d)))) { pthread_attr_t attr; pthread_attr_init(&attr); pthread_attr_setdetachstate(&attr, PTHREAD_CREATE_DETACHED); /* Save original request for followup */ d->req = req; d->chan1 = transferee; /* Transferee */ d->chan2 = transferer; /* Transferer */ d->seqno = seqno; if (ast_pthread_create_background(&th, &attr, sip_park_thread, d) < 0) { /* Could not start thread */ free(d); /* We don't need it anymore. If thread is created, d will be free'd by sip_park_thread() */ return 0; } } return -1; }

static void* sip_park_thread (  void *  stuff  )  [static]

Park SIP call support function Starts in a new thread, then parks the call XXX Should we add a wait period after streaming audio and before hangup?? Sometimes the audio can't be heard before hangup. Definition at line 133 of file sip3_refer.c. References append_history, AST_CAUSE_NORMAL_CLEARING, ast_channel_lock, ast_channel_unlock, ast_do_masquerade(), ast_hangup(), ast_log(), ast_park_call(), sip_dual::chan1, sip_dual::chan2, ext, free, ast_channel::hangupcause, LOG_DEBUG, LOG_ERROR, LOG_WARNING, option_debug, sip_dual::req, sip_dual::seqno, siprequest_free(), ast_channel::tech_pvt, transmit_message_with_text(), transmit_notify_with_sipfrag(), transmit_response(), and TRUE. { struct ast_channel *transferee, *transferer; /* Chan1: The transferee, Chan2: The transferer */ struct sip_dual *d; struct sip_request *req; int ext; int res; d = stuff; transferee = d->chan1; transferer = d->chan2; req = d->req; free(d); if (!transferee || !transferer) { ast_log(LOG_ERROR, "Missing channels for parking! Transferer %s Transferee %s\n", transferer ? "<available>" : "<missing>", transferee ? "<available>" : "<missing>" ); return NULL; } if (option_debug > 3) ast_log(LOG_DEBUG, "SIP Park: Transferer channel %s, Transferee %s\n", transferer->name, transferee->name); ast_channel_lock(transferee); if (ast_do_masquerade(transferee)) { ast_log(LOG_WARNING, "Masquerade failed.\n"); transmit_response(transferer->tech_pvt, "503 Internal error", req); ast_channel_unlock(transferee); return NULL; } ast_channel_unlock(transferee); res = ast_park_call(transferee, transferer, 0, &ext); #ifdef WHEN_WE_KNOW_THAT_THE_CLIENT_SUPPORTS_MESSAGE if (!res) { transmit_message_with_text(transferer->tech_pvt, "Unable to park call.\n"); } else { /* Then tell the transferer what happened */ sprintf(buf, "Call parked on extension '%d'", ext); transmit_message_with_text(transferer->tech_pvt, buf); } #endif /* Any way back to the current call??? */ /* Transmit response to the REFER request */ transmit_response(transferer->tech_pvt, "202 Accepted", req); if (!res) { /* Transfer succeeded */ append_history(transferer->tech_pvt, "SIPpark","Parked call on %d", ext); transmit_notify_with_sipfrag(transferer->tech_pvt, d->seqno, "200 OK", TRUE); transferer->hangupcause = AST_CAUSE_NORMAL_CLEARING; ast_hangup(transferer); /* This will cause a BYE */ if (option_debug) ast_log(LOG_DEBUG, "SIP Call parked on extension '%d'\n", ext); } else { transmit_notify_with_sipfrag(transferer->tech_pvt, d->seqno, "503 Service Unavailable", TRUE); append_history(transferer->tech_pvt, "SIPpark","Parking failed\n"); if (option_debug) ast_log(LOG_DEBUG, "SIP Call parked failed \n"); /* Do not hangup call */ } siprequest_free(req); return NULL; }

int sip_refer_allocate (  struct sip_dialog *  p  )

Allocate SIP refer structure. Definition at line 122 of file sip3_refer.c. References ast_calloc, and sip_dialog::refer. { p->refer = ast_calloc(1, sizeof(struct sip_refer)); return p->refer ? 1 : 0; }

GNURK int transmit_notify_with_sipfrag (  struct sip_dialog *  p, int  cseq, char *  message, int  terminate )

Notify a transferring party of the status of transfer. Definition at line 989 of file sip3_refer.c. References add_header(), add_header_contentLength(), add_line(), ALLOWED_METHODS, initialize_initreq(), sip_dialog::initreq, reqprep(), send_request(), SIP_MAX_PACKET, SIP_NOTIFY, sipnet, siprequest_alloc(), SUPPORTED_EXTENSIONS, TRUE, and XMIT_RELIABLE. { struct sip_request *req; char tmp[BUFSIZ/2]; req = siprequest_alloc(SIP_MAX_PACKET, &sipnet); reqprep(req, p, SIP_NOTIFY, 0, TRUE); snprintf(tmp, sizeof(tmp), "refer;id=%d", cseq); add_header(req, "Event", tmp); add_header(req, "Subscription-state", terminate ? "terminated;reason=noresource" : "active"); add_header(req, "Content-Type", "message/sipfrag;version=2.0"); add_header(req, "Allow", ALLOWED_METHODS); add_header(req, "Supported", SUPPORTED_EXTENSIONS); snprintf(tmp, sizeof(tmp), "SIP/2.0 %s\r\n", message); add_header_contentLength(req, strlen(tmp)); add_line(req, tmp); if (!p->initreq) initialize_initreq(p, req); return send_request(p, req, XMIT_RELIABLE); }

---

Variable Documentation

const struct c_referstatusstring referstatusstrings[] [static]

Table to convert from REFER status variable to string.

---

Asterisk is a trademark for Digium, inc.. | Edvina.net | Asterisk.org | This documentation was generated with Doxygen