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asterisk1.6.0.21_fcc_001.patch
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--- channels/chan_sip.c.orig 2010-08-12 15:40:52.000000000 +0900
+++ channels/chan_sip.c 2010-08-12 16:39:35.000000000 +0900
@@ -233,6 +233,7 @@
#define DEFAULT_TRANS_TIMEOUT -1 /* Use default SIP transaction timeout */
#define PROVIS_KEEPALIVE_TIMEOUT 60000 /*!< How long to wait before retransmitting a provisional response (rfc 3261 13.3.1.1) */
#define MAX_AUTHTRIES 3 /*!< Try authentication three times, then fail */
+#define MAX_AUTHTRIES_F 1
#define SIP_MAX_HEADERS 64 /*!< Max amount of SIP headers to read */
#define SIP_MAX_LINES 64 /*!< Max amount of lines in SIP attachment (like SDP) */
@@ -586,6 +587,7 @@
allowsubscribe and allowrefer on in sip.conf.
*/
#define ALLOWED_METHODS "INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO"
+#define ALLOWED_METHODS_F "INVITE, ACK, CANCEL, BYE"
/*! \brief SIP Extensions we support */
#define SUPPORTED_EXTENSIONS "replaces, timer"
@@ -636,6 +638,7 @@
#define DEFAULT_REGEXTENONQUALIFY FALSE
#define DEFAULT_T1MIN 100 /*!< 100 MS for minimal roundtrip time */
#define DEFAULT_MAX_CALL_BITRATE (384) /*!< Max bitrate for video */
+#define DEFAULT_FUSIONCOM TRUE /*!< Fusion Com. use */
#ifndef DEFAULT_USERAGENT
#define DEFAULT_USERAGENT "Asterisk PBX" /*!< Default Useragent: header unless re-defined in sip.conf */
#define DEFAULT_SDPSESSION "Asterisk PBX" /*!< Default SDP session name, (s=) header unless re-defined in sip.conf */
@@ -660,6 +663,7 @@
static char default_mohsuggest[MAX_MUSICCLASS]; /*!< Global setting for moh class to suggest when putting
* a bridged channel on hold */
static int default_maxcallbitrate; /*!< Maximum bitrate for call */
+static int default_fusioncom; /*!< Fusion Com. use */
static struct ast_codec_pref default_prefs; /*!< Default codec prefs */
/*! \brief a place to store all global settings for the sip channel driver */
@@ -1225,6 +1229,7 @@
int jointnoncodeccapability; /*!< Joint Non codec capability */
int redircodecs; /*!< Redirect codecs */
int maxcallbitrate; /*!< Maximum Call Bitrate for Video Calls */
+ int fusioncom; /*!< Fusion Com. use */
int t38_maxdatagram; /*!< T.38 FaxMaxDatagram override */
struct sip_proxy *outboundproxy; /*!< Outbound proxy for this dialog */
struct t38properties t38; /*!< T38 settings */
@@ -1429,6 +1434,7 @@
struct ast_ha *ha; /*!< ACL setting */
struct ast_variable *chanvars; /*!< Variables to set for channel created by user */
int maxcallbitrate; /*!< Maximum Bitrate for a video call */
+ int fusioncom; /*!< Fusion Com. use */
int autoframing;
struct sip_st_cfg stimer; /*!< SIP Session-Timers */
};
@@ -1508,6 +1514,7 @@
struct ast_dnsmgr_entry *dnsmgr;/*!< DNS refresh manager for peer */
struct sockaddr_in addr; /*!< IP address of peer */
int maxcallbitrate; /*!< Maximum Bitrate for a video call */
+ int fusioncom; /*!< Fusion Com. use */
unsigned int portinuri:1; /*!< Whether the port should be included in the URI */
/* Qualification */
@@ -3136,7 +3143,10 @@
ast_queue_hangup(p->owner);
} else if (p->refer && !p->alreadygone) {
ast_debug(3, "Finally hanging up channel after transfer: %s\n", p->callid);
- transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, 1);
+ if (!p->fusioncom)
+ transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, 1);
+ else
+ transmit_request(p, SIP_BYE, 0, XMIT_RELIABLE, 1);
append_history(p, "ReferBYE", "Sending BYE on transferer call leg %s", p->callid);
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
} else {
@@ -4364,6 +4374,7 @@
dialog->rtptimeout = peer->rtptimeout;
dialog->peerauth = peer->auth;
dialog->maxcallbitrate = peer->maxcallbitrate;
+ dialog->fusioncom = peer->fusioncom;
if (!dialog->portinuri)
dialog->portinuri = peer->portinuri;
if (ast_strlen_zero(dialog->tohost))
@@ -5239,7 +5250,10 @@
textqos = ast_rtp_get_quality(p->trtp, NULL);
/* Send a hangup */
if (oldowner->_state == AST_STATE_UP) {
- transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, 1);
+ if (!p->fusioncom)
+ transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, 1);
+ else
+ transmit_request(p, SIP_BYE, 0, XMIT_RELIABLE, 1);
}
/* Get RTCP quality before end of call */
@@ -6190,7 +6204,10 @@
const char *host = S_OR(pvt->fromdomain, ast_inet_ntoa(pvt->ourip.sin_addr));
- ast_string_field_build(pvt, callid, "%s@%s", generate_random_string(buf, sizeof(buf)), host);
+ if (!pvt->fusioncom)
+ ast_string_field_build(pvt, callid, "%s@%s", generate_random_string(buf, sizeof(buf)), host);
+ else
+ ast_string_field_build(pvt, callid, "%s@%s", generate_random_string(buf, sizeof(buf)), ast_inet_ntoa(pvt->ourip.sin_addr));
}
@@ -6333,6 +6350,7 @@
if (p->udptl)
ast_udptl_setqos(p->udptl, global_tos_audio, global_cos_audio);
p->maxcallbitrate = default_maxcallbitrate;
+ p->fusioncom = default_fusioncom;
p->autoframing = global_autoframing;
ast_rtp_codec_setpref(p->rtp, &p->prefs);
}
@@ -7952,17 +7970,22 @@
}
/* Add rport to first VIA header if requested */
- snprintf(new, sizeof(new), "%s;received=%s;rport=%d%s%s",
- leftmost, ast_inet_ntoa(p->recv.sin_addr),
- ntohs(p->recv.sin_port),
- others ? "," : "", others ? others : "");
+ if (!p->fusioncom) {
+ snprintf(new, sizeof(new), "%s;received=%s;rport=%d%s%s",
+ leftmost, ast_inet_ntoa(p->recv.sin_addr),
+ ntohs(p->recv.sin_port),
+ others ? "," : "", others ? others : "");
+ }
} else {
/* We should *always* add a received to the topmost via */
- snprintf(new, sizeof(new), "%s;received=%s%s%s",
- leftmost, ast_inet_ntoa(p->recv.sin_addr),
- others ? "," : "", others ? others : "");
+ if (!p->fusioncom) {
+ snprintf(new, sizeof(new), "%s;received=%s%s%s",
+ leftmost, ast_inet_ntoa(p->recv.sin_addr),
+ others ? "," : "", others ? others : "");
+ }
}
- oh = new; /* the header to copy */
+ if (!p->fusioncom)
+ oh = new; /* the header to copy */
} /* else add the following via headers untouched */
add_header(req, field, oh);
copied++;
@@ -8157,6 +8180,7 @@
{
char newto[256];
const char *ot;
+ const char *allowed_methods = p->fusioncom ? ALLOWED_METHODS_F : ALLOWED_METHODS;
init_resp(resp, msg);
copy_via_headers(p, resp, req, "Via");
@@ -8180,8 +8204,9 @@
copy_header(resp, req, "CSeq");
if (!ast_strlen_zero(global_useragent))
add_header(resp, "User-Agent", global_useragent);
- add_header(resp, "Allow", ALLOWED_METHODS);
- add_header(resp, "Supported", SUPPORTED_EXTENSIONS);
+ add_header(resp, "Allow", allowed_methods);
+ if (!p->fusioncom)
+ add_header(resp, "Supported", SUPPORTED_EXTENSIONS);
/* If this is an invite, add Session-Timers related headers if the feature is active for this session */
if (p->method == SIP_INVITE && p->stimer && p->stimer->st_active == TRUE && p->stimer->st_active_peer_ua == TRUE) {
@@ -8207,7 +8232,12 @@
add_header(resp, "Contact", contact); /* Not when we unregister */
}
} else if (!ast_strlen_zero(p->our_contact) && resp_needs_contact(msg, p->method)) {
- add_header(resp, "Contact", p->our_contact);
+ if (!p->fusioncom) {
+ add_header(resp, "Contact", p->our_contact);
+ } else {
+ if (strncmp(msg, "100", 3))
+ add_header(resp, "Contact", p->our_contact);
+ }
}
if (!ast_strlen_zero(p->url)) {
@@ -8285,9 +8315,11 @@
snprintf(tmp, sizeof(tmp), "%d %s", seqno, sip_methods[sipmethod].text);
add_header(req, "Via", p->via);
- if (p->route) {
- set_destination(p, p->route->hop);
- add_route(req, is_strict ? p->route->next : p->route);
+ if (!p->fusioncom) {
+ if (p->route) {
+ set_destination(p, p->route->hop);
+ add_route(req, is_strict ? p->route->next : p->route);
+ }
}
add_header(req, "Max-Forwards", DEFAULT_MAX_FORWARDS);
@@ -8325,8 +8357,10 @@
if (!ast_strlen_zero(global_useragent))
add_header(req, "User-Agent", global_useragent);
- if (!ast_strlen_zero(p->rpid))
- add_header(req, "Remote-Party-ID", p->rpid);
+ if (!p->fusioncom) {
+ if (!ast_strlen_zero(p->rpid))
+ add_header(req, "Remote-Party-ID", p->rpid);
+ }
if (!ast_strlen_zero(p->url)) {
add_header(req, "Access-URL", p->url);
@@ -8366,14 +8400,16 @@
}
respprep(&resp, p, msg, req);
add_header_contentLength(&resp, 0);
- /* If we are cancelling an incoming invite for some reason, add information
+ if (!p->fusioncom) {
+ /* If we are cancelling an incoming invite for some reason, add information
about the reason why we are doing this in clear text */
- if (p->method == SIP_INVITE && msg[0] != '1' && p->owner && p->owner->hangupcause) {
- char buf[10];
+ if (p->method == SIP_INVITE && msg[0] != '1' && p->owner && p->owner->hangupcause) {
+ char buf[10];
- add_header(&resp, "X-Asterisk-HangupCause", ast_cause2str(p->owner->hangupcause));
- snprintf(buf, sizeof(buf), "%d", p->owner->hangupcause);
- add_header(&resp, "X-Asterisk-HangupCauseCode", buf);
+ add_header(&resp, "X-Asterisk-HangupCause", ast_cause2str(p->owner->hangupcause));
+ snprintf(buf, sizeof(buf), "%d", p->owner->hangupcause);
+ add_header(&resp, "X-Asterisk-HangupCauseCode", buf);
+ }
}
return send_response(p, &resp, reliable, seqno);
}
@@ -9014,8 +9050,10 @@
ast_debug(3, "-- Done with adding codecs to SDP\n");
- if (!p->owner || !ast_internal_timing_enabled(p->owner))
- ast_str_append(&a_audio, 0, "a=silenceSupp:off - - - -\r\n");
+ if (!p->fusioncom) {
+ if (!p->owner || !ast_internal_timing_enabled(p->owner))
+ ast_str_append(&a_audio, 0, "a=silenceSupp:off - - - -\r\n");
+ }
if (min_audio_packet_size)
ast_str_append(&a_audio, 0, "a=ptime:%d\r\n", min_audio_packet_size);
@@ -9276,11 +9314,13 @@
static int transmit_reinvite_with_sdp(struct sip_pvt *p, int t38version, int oldsdp)
{
struct sip_request req;
+ const char *allowed_methods = p->fusioncom ? ALLOWED_METHODS_F : ALLOWED_METHODS;
reqprep(&req, p, ast_test_flag(&p->flags[0], SIP_REINVITE_UPDATE) ? SIP_UPDATE : SIP_INVITE, 0, 1);
- add_header(&req, "Allow", ALLOWED_METHODS);
- add_header(&req, "Supported", SUPPORTED_EXTENSIONS);
+ add_header(&req, "Allow", allowed_methods);
+ if (!p->fusioncom)
+ add_header(&req, "Supported", SUPPORTED_EXTENSIONS);
if (sipdebug) {
if (oldsdp == TRUE)
add_header(&req, "X-asterisk-Info", "SIP re-invite (Session-Timers)");
@@ -9566,7 +9606,8 @@
* NOTIFY messages will use this function for preparing the request and should
* have Route headers present.
*/
- add_route(req, p->route);
+ if (!p->fusioncom)
+ add_route(req, p->route);
/* Build Remote Party-ID and From */
if (ast_test_flag(&p->flags[0], SIP_SENDRPID) && (sipmethod == SIP_INVITE)) {
@@ -9582,8 +9623,10 @@
add_header(req, "CSeq", tmp_n);
if (!ast_strlen_zero(global_useragent))
add_header(req, "User-Agent", global_useragent);
- if (!ast_strlen_zero(p->rpid))
- add_header(req, "Remote-Party-ID", p->rpid);
+ if (!p->fusioncom) {
+ if (!ast_strlen_zero(p->rpid))
+ add_header(req, "Remote-Party-ID", p->rpid);
+ }
}
/*! \brief Build REFER/INVITE/OPTIONS message and transmit it
@@ -9596,7 +9639,8 @@
static int transmit_invite(struct sip_pvt *p, int sipmethod, int sdp, int init)
{
struct sip_request req;
-
+ const char *allowed_methods = p->fusioncom ? ALLOWED_METHODS_F : ALLOWED_METHODS;
+
req.method = sipmethod;
if (init) {/* Bump branch even on initial requests */
p->branch ^= ast_random();
@@ -9611,7 +9655,8 @@
if (p->options && p->options->auth)
add_header(&req, p->options->authheader, p->options->auth);
- append_date(&req);
+ if (!p->fusioncom)
+ append_date(&req);
if (sipmethod == SIP_REFER) { /* Call transfer */
if (p->refer) {
char buf[SIPBUFSIZE];
@@ -9645,8 +9690,9 @@
add_header(&req, "Min-SE", i2astr);
}
- add_header(&req, "Allow", ALLOWED_METHODS);
- add_header(&req, "Supported", SUPPORTED_EXTENSIONS);
+ add_header(&req, "Allow", allowed_methods);
+ if (!p->fusioncom)
+ add_header(&req, "Supported", SUPPORTED_EXTENSIONS);
if (p->options && p->options->addsipheaders && p->owner) {
struct ast_channel *chan = p->owner; /* The owner channel */
struct varshead *headp;
@@ -10097,6 +10143,8 @@
char addr[80];
struct sip_pvt *p;
char *fromdomain;
+ char callidcopy[MAXHOSTNAMELEN];
+ char *wk = NULL;
/* exit if we are already in process with this registrar ?*/
if ( r == NULL || ((auth==NULL) && (r->regstate==REG_STATE_REGSENT || r->regstate==REG_STATE_AUTHSENT))) {
@@ -10159,7 +10207,8 @@
return 0;
}
/* Copy back Call-ID in case create_addr changed it */
- ast_string_field_set(r, callid, p->callid);
+ if (!p->fusioncom)
+ ast_string_field_set(r, callid, p->callid);
if (r->portno) {
p->sa.sin_port = htons(r->portno);
p->recv.sin_port = htons(r->portno);
@@ -10169,7 +10218,7 @@
r->call = p; /* Save pointer to SIP dialog */
p->registry = registry_addref(r); /* Add pointer to registry in packet */
if (!ast_strlen_zero(r->secret)) /* Secret (password) */
- ast_string_field_set(p, peersecret, r->secret);
+ ast_string_field_set(p, peersecret, r->secret);
if (!ast_strlen_zero(r->md5secret))
ast_string_field_set(p, peermd5secret, r->md5secret);
/* User name in this realm
@@ -10200,6 +10249,16 @@
*/
ast_sip_ouraddrfor(&p->sa.sin_addr, &p->ourip, p);
build_contact(p);
+ if (p->fusioncom) {
+ ast_copy_string(callidcopy, r->callid, sizeof(callidcopy));
+ wk = strrchr(callidcopy, '@');
+ if (wk) {
+ *wk++ = '\0';
+ ast_string_field_build(p, callid, "%s@%s", callidcopy, ast_inet_ntoa(p->ourip.sin_addr));
+ }
+ /* Copy back Call-ID in case create_addr changed it */
+ ast_string_field_set(r, callid, p->callid);
+ }
}
/* set up a timeout */
@@ -10287,17 +10346,20 @@
p->noncecount = ++r->noncecount;
memset(digest, 0, sizeof(digest));
- if(!build_reply_digest(p, sipmethod, digest, sizeof(digest)))
- add_header(&req, "Authorization", digest);
- else
- ast_log(LOG_NOTICE, "No authorization available for authentication of registration to %s@%s\n", r->username, r->hostname);
+ if (!p->fusioncom) {
+ if(!build_reply_digest(p, sipmethod, digest, sizeof(digest)))
+ add_header(&req, "Authorization", digest);
+ else
+ ast_log(LOG_NOTICE, "No authorization available for authentication of registration to %s@%s\n", r->username, r->hostname);
+ }
}
snprintf(tmp, sizeof(tmp), "%d", r->expiry);
add_header(&req, "Expires", tmp);
add_header(&req, "Contact", p->our_contact);
- add_header(&req, "Event", "registration");
+ if (!p->fusioncom)
+ add_header(&req, "Event", "registration");
add_header_contentLength(&req, 0);
initialize_initreq(p, &req);
@@ -10344,6 +10406,7 @@
char referto[256];
char *ttag, *ftag;
char *theirtag = ast_strdupa(p->theirtag);
+ const char *allowed_methods = p->fusioncom ? ALLOWED_METHODS_F : ALLOWED_METHODS;
if (sipdebug)
ast_debug(1, "SIP transfer of %s to %s\n", p->callid, dest);
@@ -10386,11 +10449,13 @@
reqprep(&req, p, SIP_REFER, 0, 1);
- add_header(&req, "Refer-To", referto);
- add_header(&req, "Allow", ALLOWED_METHODS);
- add_header(&req, "Supported", SUPPORTED_EXTENSIONS);
- if (!ast_strlen_zero(p->our_contact))
- add_header(&req, "Referred-By", p->our_contact);
+ add_header(&req, "Allow", allowed_methods);
+ if (!p->fusioncom) {
+ add_header(&req, "Refer-To", referto);
+ add_header(&req, "Supported", SUPPORTED_EXTENSIONS);
+ if (!ast_strlen_zero(p->our_contact))
+ add_header(&req, "Referred-By", p->our_contact);
+ }
return send_request(p, &req, XMIT_RELIABLE, p->ocseq);
/* We should propably wait for a NOTIFY here until we ack the transfer */
@@ -12424,6 +12489,7 @@
p->noncodeccapability &= ~AST_RTP_DTMF;
p->jointnoncodeccapability = p->noncodeccapability;
p->maxcallbitrate = user->maxcallbitrate;
+ p->fusioncom = user->fusioncom;
/* If we do not support video, remove video from call structure */
if ((!ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) || !(p->capability & AST_FORMAT_VIDEO_MASK)) && p->vrtp) {
ast_rtp_destroy(p->vrtp);
@@ -12564,6 +12630,7 @@
if (p->peercapability)
p->jointcapability &= p->peercapability;
p->maxcallbitrate = peer->maxcallbitrate;
+ p->fusioncom = peer->fusioncom;
if ((!ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) || !(p->capability & AST_FORMAT_VIDEO_MASK)) && p->vrtp) {
ast_rtp_destroy(p->vrtp);
p->vrtp = NULL;
@@ -13741,6 +13808,7 @@
ast_cli(fd, " Dynamic : %s\n", cli_yesno(peer->host_dynamic));
ast_cli(fd, " Callerid : %s\n", ast_callerid_merge(cbuf, sizeof(cbuf), peer->cid_name, peer->cid_num, "<unspecified>"));
ast_cli(fd, " MaxCallBR : %d kbps\n", peer->maxcallbitrate);
+ ast_cli(fd, " Fusion Com. : %s\n", peer->fusioncom ? "Yes" : "No");
ast_cli(fd, " Expire : %ld\n", ast_sched_when(sched, peer->expire));
ast_cli(fd, " Insecure : %s\n", insecure2str(ast_test_flag(&peer->flags[0], SIP_INSECURE)));
ast_cli(fd, " Nat : %s\n", nat2str(ast_test_flag(&peer->flags[0], SIP_NAT)));
@@ -13843,6 +13911,7 @@
astman_append(s, "Call-limit: %d\r\n", peer->call_limit);
astman_append(s, "Busy-level: %d\r\n", peer->busy_level);
astman_append(s, "MaxCallBR: %d kbps\r\n", peer->maxcallbitrate);
+ astman_append(s, "Fusion Com.: %s\r\n", peer->fusioncom ? "Yes" : "No");
astman_append(s, "Dynamic: %s\r\n", peer->host_dynamic?"Y":"N");
astman_append(s, "Callerid: %s\r\n", ast_callerid_merge(cbuf, sizeof(cbuf), peer->cid_name, peer->cid_num, ""));
astman_append(s, "RegExpire: %ld seconds\r\n", ast_sched_when(sched, peer->expire));
@@ -13947,6 +14016,7 @@
ast_cli(a->fd, " AMA flags : %s\n", ast_cdr_flags2str(user->amaflags));
ast_cli(a->fd, " Transfer mode: %s\n", transfermode2str(user->allowtransfer));
ast_cli(a->fd, " MaxCallBR : %d kbps\n", user->maxcallbitrate);
+ ast_cli(a->fd, " Fusion Com. : %s\n", user->fusioncom ? "Yes" : "No");
ast_cli(a->fd, " CallingPres : %s\n", ast_describe_caller_presentation(user->callingpres));
ast_cli(a->fd, " Call limit : %d\n", user->call_limit);
ast_cli(a->fd, " Callgroup : ");
@@ -14219,6 +14289,7 @@
ast_cli(a->fd, " Notify hold state: %s\n", cli_yesno(global_notifyhold));
ast_cli(a->fd, " SIP Transfer mode: %s\n", transfermode2str(global_allowtransfer));
ast_cli(a->fd, " Max Call Bitrate: %d kbps\n", default_maxcallbitrate);
+ ast_cli(a->fd, " Fusion Com.: %s\n", default_fusioncom ? "Yes" : "No");
ast_cli(a->fd, " Auto-Framing: %s\n", cli_yesno(global_autoframing));
ast_cli(a->fd, " Outb. proxy: %s %s\n", ast_strlen_zero(global_outboundproxy.name) ? "<not set>" : global_outboundproxy.name,
global_outboundproxy.force ? "(forced)" : "");
@@ -14583,6 +14654,7 @@
ast_cli(a->fd, " T.38 support %s\n", cli_yesno(cur->udptl != NULL));
ast_cli(a->fd, " Video support %s\n", cli_yesno(cur->vrtp != NULL));
ast_cli(a->fd, " MaxCallBR: %d kbps\n", cur->maxcallbitrate);
+ ast_cli(a->fd, " Fusion Com.: %s\n", cur->fusioncom ? "Yes" : "No");
ast_cli(a->fd, " Theoretical Address: %s:%d\n", ast_inet_ntoa(cur->sa.sin_addr), ntohs(cur->sa.sin_port));
ast_cli(a->fd, " Received Address: %s:%d\n", ast_inet_ntoa(cur->recv.sin_addr), ntohs(cur->recv.sin_port));
ast_cli(a->fd, " SIP Transfer mode: %s\n", transfermode2str(cur->allowtransfer));
@@ -15708,7 +15780,10 @@
return;
/* Perhaps there is an SD change INVITE outstanding */
- transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, TRUE);
+ if (!p->fusioncom)
+ transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, TRUE);
+ else
+ transmit_request(p, SIP_BYE, 0, XMIT_RELIABLE, TRUE);
}
ast_clear_flag(&p->flags[0], SIP_PENDINGBYE);
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
@@ -15934,9 +16009,10 @@
/* Then we AUTH */
ast_string_field_set(p, theirtag, NULL); /* forget their old tag, so we don't match tags when getting response */
if (!req->ignore) {
- if (p->authtries < MAX_AUTHTRIES)
+ int max_authtries = p->fusioncom ? MAX_AUTHTRIES_F : MAX_AUTHTRIES;
+ if (p->authtries < max_authtries)
p->invitestate = INV_CALLING;
- if (p->authtries == MAX_AUTHTRIES || do_proxy_auth(p, req, resp, SIP_INVITE, 1)) {
+ if ((p->authtries == max_authtries) || do_proxy_auth(p, req, resp, SIP_INVITE, 1)) {
ast_log(LOG_NOTICE, "Failed to authenticate on INVITE to '%s'\n", get_header(&p->initreq, "From"));
p->needdestroy = 1;
sip_alreadygone(p);
@@ -16113,10 +16189,11 @@
int expires, expires_ms;
struct sip_registry *r;
r=p->registry;
+ int max_authtries = p->fusioncom ? MAX_AUTHTRIES_F : MAX_AUTHTRIES;
switch (resp) {
case 401: /* Unauthorized */
- if (p->authtries == MAX_AUTHTRIES || do_register_auth(p, req, resp)) {
+ if (p->authtries == max_authtries || do_register_auth(p, req, resp)) {
ast_log(LOG_NOTICE, "Failed to authenticate on REGISTER to '%s@%s' (Tries %d)\n", p->registry->username, p->registry->hostname, p->authtries);
p->needdestroy = 1;
}
@@ -16135,7 +16212,7 @@
AST_SCHED_DEL(sched, r->timeout);
break;
case 407: /* Proxy auth */
- if (p->authtries == MAX_AUTHTRIES || do_register_auth(p, req, resp)) {
+ if (p->authtries == max_authtries || do_register_auth(p, req, resp)) {
ast_log(LOG_NOTICE, "Failed to authenticate on REGISTER to '%s' (tries '%d')\n", get_header(&p->initreq, "From"), p->authtries);
p->needdestroy = 1;
}
@@ -16235,7 +16312,10 @@
if (expires <= EXPIRY_GUARD_LIMIT)
expires_ms -= MAX((expires_ms * EXPIRY_GUARD_PCT), EXPIRY_GUARD_MIN);
else
- expires_ms -= EXPIRY_GUARD_SECS * 1000;
+ if (!p->fusioncom)
+ expires_ms -= EXPIRY_GUARD_SECS * 1000;
+ else
+ expires_ms -= (expires * 0.5) * 1000;
if (sipdebug)
ast_log(LOG_NOTICE, "Outbound Registration: Expiry for %s is %d sec (Scheduling reregistration in %d s)\n", r->hostname, expires, expires_ms/1000);
@@ -16326,6 +16406,7 @@
char *c_copy = ast_strdupa(c);
/* Skip the Cseq and its subsequent spaces */
const char *msg = ast_skip_blanks(ast_skip_nonblanks(c_copy));
+ int max_authtries = p->fusioncom ? MAX_AUTHTRIES_F : MAX_AUTHTRIES;
if (!msg)
msg = "";
@@ -16461,7 +16542,7 @@
ast_log(LOG_WARNING, "Asked to authenticate %s, to %s:%d but we have no matching peer!\n",
msg, ast_inet_ntoa(p->recv.sin_addr), ntohs(p->recv.sin_port));
p->needdestroy = 1;
- } else if ((p->authtries == MAX_AUTHTRIES) || do_proxy_auth(p, req, resp, sipmethod, 0)) {
+ } else if ((p->authtries == max_authtries) || do_proxy_auth(p, req, resp, sipmethod, 0)) {
ast_log(LOG_NOTICE, "Failed to authenticate on %s to '%s'\n", msg, get_header(&p->initreq, "From"));
p->needdestroy = 1;
}
@@ -16699,7 +16780,7 @@
else if (sipmethod == SIP_INVITE)
handle_response_invite(p, resp, rest, req, seqno);
else if (sipmethod == SIP_BYE) {
- if (p->authtries == MAX_AUTHTRIES || do_proxy_auth(p, req, resp, sipmethod, 0)) {
+ if (p->authtries == max_authtries || do_proxy_auth(p, req, resp, sipmethod, 0)) {
ast_log(LOG_NOTICE, "Failed to authenticate on %s to '%s'\n", msg, get_header(&p->initreq, "From"));
p->needdestroy = 1;
}
@@ -21341,6 +21422,7 @@
user->capability = global_capability;
user->allowtransfer = global_allowtransfer;
user->maxcallbitrate = default_maxcallbitrate;
+ user->fusioncom = default_fusioncom;
user->autoframing = global_autoframing;
user->t38_maxdatagram = global_t38_maxdatagram;
if (global_callcounter)
@@ -21434,6 +21516,8 @@
user->maxcallbitrate = atoi(v->value);
if (user->maxcallbitrate < 0)
user->maxcallbitrate = default_maxcallbitrate;
+ } else if (!strcasecmp(v->name, "fusioncom")) {
+ user->fusioncom = ast_true(v->value);
} else if (!strcasecmp(v->name, "session-timers")) {
int i = (int) str2stmode(v->value);
if (i < 0) {
@@ -21501,6 +21585,7 @@
peer->defaddr.sin_family = AF_INET;
peer->capability = global_capability;
peer->maxcallbitrate = default_maxcallbitrate;
+ peer->fusioncom = default_fusioncom;
peer->rtptimeout = global_rtptimeout;
peer->rtpholdtimeout = global_rtpholdtimeout;
peer->rtpkeepalive = global_rtpkeepalive;
@@ -21895,6 +21980,8 @@
if (peer->maxcallbitrate < 0) {
peer->maxcallbitrate = default_maxcallbitrate;
}
+ } else if (!strcasecmp(v->name, "fusioncom")) {
+ peer->fusioncom = ast_true(v->value);
} else if (!strcasecmp(v->name, "session-timers")) {
int i = (int) str2stmode(v->value);
if (i < 0) {
@@ -22271,6 +22358,7 @@
default_fromdomain[0] = '\0';
default_qualify = DEFAULT_QUALIFY;
default_maxcallbitrate = DEFAULT_MAX_CALL_BITRATE;
+ default_fusioncom = DEFAULT_FUSIONCOM;
ast_copy_string(default_mohinterpret, DEFAULT_MOHINTERPRET, sizeof(default_mohinterpret));
ast_copy_string(default_mohsuggest, DEFAULT_MOHSUGGEST, sizeof(default_mohsuggest));
ast_copy_string(default_vmexten, DEFAULT_VMEXTEN, sizeof(default_vmexten));
@@ -22647,6 +22735,8 @@
default_maxcallbitrate = DEFAULT_MAX_CALL_BITRATE;
} else if (!strcasecmp(v->name, "matchexterniplocally")) {
global_matchexterniplocally = ast_true(v->value);
+ } else if (!strcasecmp(v->name, "fusioncom")) {
+ default_fusioncom = ast_true(v->value);
} else if (!strcasecmp(v->name, "constantssrc")) {
ast_set2_flag(&global_flags[1], ast_true(v->value), SIP_PAGE2_CONSTANT_SSRC);
} else if (!strcasecmp(v->name, "session-timers")) {
@@ -23324,7 +23414,7 @@
return;
regspacing = default_expiry * 1000/regobjs;
if (regspacing > 100)
- regspacing = 100;
+ regspacing = 100 * 10;
ms = regspacing;
ASTOBJ_CONTAINER_TRAVERSE(®l, 1, do {
ASTOBJ_WRLOCK(iterator);