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Restore SIP DTMF overlap dialing method.
The recent fix for ASTERISK-17288 to get RFC3578 SIP overlap support working correctly removed a long standing ability to do overlap dialing using DTMF in the early media phase of a call. See ASTERISK-18702 it has a very good description of the issue. I started with Pavel Troller's chan_sip.diff patch on issue ASTERISK-18702. * Added 'dtmf' enum value to sip.conf allowoverlap config option. The new option value causes the Incomplte application to not send anything with chan_sip so the caller can supply more digits via DTMF. * Renames SIP_GET_DEST_PICKUP_EXTEN_FOUND to SIP_GET_DEST_EXTEN_MATCHMORE since that is what it really means. * Fixed get_destination() inconsistency with the pickup extension matching. * Fixed initialization of PAGE3 of global_flags in reload_config(). (closes issue ASTERISK-18702) Reported by: Pavel Troller Review: https://reviewboard.asterisk.org/r/1517/ Review: https://reviewboard.asterisk.org/r/1582/ ........ Merged revisions 345273 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 345275 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345276 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
@@ -1384,6 +1384,7 @@ static void print_group(int fd, ast_group_t group, int crlf);
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static const char *dtmfmode2str(int mode) attribute_const;
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static int str2dtmfmode(const char *str) attribute_unused;
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static const char *insecure2str(int mode) attribute_const;
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static const char *allowoverlap2str(int mode) attribute_const;
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static void cleanup_stale_contexts(char *new, char *old);
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static void print_codec_to_cli(int fd, struct ast_codec_pref *pref);
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static const char *domain_mode_to_text(const enum domain_mode mode);
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@@ -6837,17 +6838,25 @@ static int sip_indicate(struct ast_channel *ast, int condition, const void *data
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break;
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case AST_CONTROL_INCOMPLETE:
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if (ast->_state != AST_STATE_UP) {
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if (ast_test_flag(&p->flags[1], SIP_PAGE2_ALLOWOVERLAP)) {
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switch (ast_test_flag(&p->flags[1], SIP_PAGE2_ALLOWOVERLAP)) {
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case SIP_PAGE2_ALLOWOVERLAP_YES:
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transmit_response_reliable(p, "484 Address Incomplete", &p->initreq);
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} else {
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p->invitestate = INV_COMPLETED;
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sip_alreadygone(p);
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ast_softhangup_nolock(ast, AST_SOFTHANGUP_DEV);
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break;
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case SIP_PAGE2_ALLOWOVERLAP_DTMF:
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/* Just wait for inband DTMF digits */
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break;
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default:
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/* it actually means no support for overlap */
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transmit_response_reliable(p, "404 Not Found", &p->initreq);
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p->invitestate = INV_COMPLETED;
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sip_alreadygone(p);
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ast_softhangup_nolock(ast, AST_SOFTHANGUP_DEV);
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break;
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}
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p->invitestate = INV_COMPLETED;
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sip_alreadygone(p);
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ast_softhangup_nolock(ast, AST_SOFTHANGUP_DEV);
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break;
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}
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res = 0;
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break;
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case AST_CONTROL_PROCEEDING:
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if ((ast->_state != AST_STATE_UP) &&
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@@ -15506,18 +15515,23 @@ static enum sip_get_dest_result get_destination(struct sip_pvt *p, struct sip_re
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}
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} else {
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struct ast_cc_agent *agent;
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int which = 0;
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/* Check the dialplan for the username part of the request URI,
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the domain will be stored in the SIPDOMAIN variable
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Return 0 if we have a matching extension */
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if (ast_exists_extension(NULL, p->context, uri, 1, S_OR(p->cid_num, from)) ||
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(ast_exists_extension(NULL, p->context, decoded_uri, 1, S_OR(p->cid_num, from)) && (which = 1)) ||
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!strcmp(decoded_uri, ast_pickup_ext())) {
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if (ast_exists_extension(NULL, p->context, uri, 1, S_OR(p->cid_num, from))) {
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if (!oreq) {
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ast_string_field_set(p, exten, which ? decoded_uri : uri);
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ast_string_field_set(p, exten, uri);
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}
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return SIP_GET_DEST_EXTEN_FOUND;
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} else if ((agent = find_sip_cc_agent_by_notify_uri(tmp))) {
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}
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if (ast_exists_extension(NULL, p->context, decoded_uri, 1, S_OR(p->cid_num, from))
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|| !strcmp(decoded_uri, ast_pickup_ext())) {
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if (!oreq) {
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ast_string_field_set(p, exten, decoded_uri);
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}
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return SIP_GET_DEST_EXTEN_FOUND;
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}
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if ((agent = find_sip_cc_agent_by_notify_uri(tmp))) {
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struct sip_cc_agent_pvt *agent_pvt = agent->private_data;
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/* This is a CC recall. We can set p's extension to the exten from
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* the original INVITE
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@@ -15536,11 +15550,12 @@ static enum sip_get_dest_result get_destination(struct sip_pvt *p, struct sip_re
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}
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}
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/* Return 1 for pickup extension or overlap dialling support (if we support it) */
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if((ast_test_flag(&global_flags[1], SIP_PAGE2_ALLOWOVERLAP) &&
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ast_canmatch_extension(NULL, p->context, decoded_uri, 1, S_OR(p->cid_num, from))) ||
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!strncmp(decoded_uri, ast_pickup_ext(), strlen(decoded_uri))) {
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return SIP_GET_DEST_PICKUP_EXTEN_FOUND;
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if (ast_test_flag(&global_flags[1], SIP_PAGE2_ALLOWOVERLAP)
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&& (ast_canmatch_extension(NULL, p->context, uri, 1, S_OR(p->cid_num, from))
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|| ast_canmatch_extension(NULL, p->context, decoded_uri, 1, S_OR(p->cid_num, from))
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|| !strncmp(decoded_uri, ast_pickup_ext(), strlen(decoded_uri)))) {
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/* Overlap dialing is enabled and we need more digits to match an extension. */
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return SIP_GET_DEST_EXTEN_MATCHMORE;
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}
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return SIP_GET_DEST_EXTEN_NOT_FOUND;
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@@ -17150,6 +17165,19 @@ static const char *insecure2str(int mode)
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return map_x_s(insecurestr, mode, "<error>");
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}
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static const struct _map_x_s allowoverlapstr[] = {
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{ SIP_PAGE2_ALLOWOVERLAP_YES, "Yes" },
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{ SIP_PAGE2_ALLOWOVERLAP_DTMF, "DTMF" },
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{ SIP_PAGE2_ALLOWOVERLAP_NO, "No" },
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{ -1, NULL }, /* terminator */
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};
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/*! \brief Convert AllowOverlap setting to printable string */
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static const char *allowoverlap2str(int mode)
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{
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return map_x_s(allowoverlapstr, mode, "<error>");
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}
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/*! \brief Destroy disused contexts between reloads
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Only used in reload_config so the code for regcontext doesn't get ugly
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*/
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@@ -17696,7 +17724,7 @@ static char *_sip_show_peer(int type, int fd, struct mansession *s, const struct
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ast_cli(fd, " Trust RPID : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[0], SIP_TRUSTRPID)));
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ast_cli(fd, " Send RPID : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[0], SIP_SENDRPID)));
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ast_cli(fd, " Subscriptions: %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[1], SIP_PAGE2_ALLOWSUBSCRIBE)));
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ast_cli(fd, " Overlap dial : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[1], SIP_PAGE2_ALLOWOVERLAP)));
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ast_cli(fd, " Overlap dial : %s\n", allowoverlap2str(ast_test_flag(&peer->flags[1], SIP_PAGE2_ALLOWOVERLAP)));
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if (peer->outboundproxy)
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ast_cli(fd, " Outb. proxy : %s %s\n", ast_strlen_zero(peer->outboundproxy->name) ? "<not set>" : peer->outboundproxy->name,
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peer->outboundproxy->force ? "(forced)" : "");
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@@ -18253,7 +18281,7 @@ static char *sip_show_settings(struct ast_cli_entry *e, int cmd, struct ast_cli_
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ast_cli(a->fd, " Match Auth Username: %s\n", AST_CLI_YESNO(global_match_auth_username));
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ast_cli(a->fd, " Allow unknown access: %s\n", AST_CLI_YESNO(sip_cfg.allowguest));
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ast_cli(a->fd, " Allow subscriptions: %s\n", AST_CLI_YESNO(ast_test_flag(&global_flags[1], SIP_PAGE2_ALLOWSUBSCRIBE)));
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ast_cli(a->fd, " Allow overlap dialing: %s\n", AST_CLI_YESNO(ast_test_flag(&global_flags[1], SIP_PAGE2_ALLOWOVERLAP)));
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ast_cli(a->fd, " Allow overlap dialing: %s\n", allowoverlap2str(ast_test_flag(&global_flags[1], SIP_PAGE2_ALLOWOVERLAP)));
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ast_cli(a->fd, " Allow promisc. redir: %s\n", AST_CLI_YESNO(ast_test_flag(&global_flags[0], SIP_PROMISCREDIR)));
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ast_cli(a->fd, " Enable call counters: %s\n", AST_CLI_YESNO(global_callcounter));
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ast_cli(a->fd, " SIP domain support: %s\n", AST_CLI_YESNO(!AST_LIST_EMPTY(&domain_list)));
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@@ -21546,10 +21574,13 @@ static void handle_response(struct sip_pvt *p, int resp, const char *rest, struc
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break;
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case 484: /* Address Incomplete */
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if (owner && sipmethod != SIP_BYE) {
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if (ast_test_flag(&p->flags[1], SIP_PAGE2_ALLOWOVERLAP)) {
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switch (ast_test_flag(&p->flags[1], SIP_PAGE2_ALLOWOVERLAP)) {
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case SIP_PAGE2_ALLOWOVERLAP_YES:
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ast_queue_hangup_with_cause(p->owner, hangup_sip2cause(resp));
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} else {
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break;
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default:
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ast_queue_hangup_with_cause(p->owner, hangup_sip2cause(404));
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break;
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}
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}
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break;
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@@ -22262,7 +22293,7 @@ static int handle_request_options(struct sip_pvt *p, struct sip_request *req, st
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case SIP_GET_DEST_INVALID_URI:
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msg = "416 Unsupported URI scheme";
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break;
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case SIP_GET_DEST_PICKUP_EXTEN_FOUND:
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case SIP_GET_DEST_EXTEN_MATCHMORE:
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case SIP_GET_DEST_REFUSED:
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case SIP_GET_DEST_EXTEN_NOT_FOUND:
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//msg = "404 Not Found";
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@@ -22983,12 +23014,21 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
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case SIP_GET_DEST_INVALID_URI:
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transmit_response_reliable(p, "416 Unsupported URI scheme", req);
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break;
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case SIP_GET_DEST_PICKUP_EXTEN_FOUND:
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if (ast_test_flag(&p->flags[1], SIP_PAGE2_ALLOWOVERLAP)) {
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case SIP_GET_DEST_EXTEN_MATCHMORE:
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if (ast_test_flag(&p->flags[1], SIP_PAGE2_ALLOWOVERLAP)
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== SIP_PAGE2_ALLOWOVERLAP_YES) {
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transmit_response_reliable(p, "484 Address Incomplete", req);
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break;
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}
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/* INTENTIONAL FALL THROUGH */
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/*
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* XXX We would have to implement collecting more digits in
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* chan_sip for any other schemes of overlap dialing.
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*
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* For SIP_PAGE2_ALLOWOVERLAP_DTMF it is better to do this in
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* the dialplan using the Incomplete application rather than
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* having the channel driver do it.
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*/
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/* Fall through */
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case SIP_GET_DEST_EXTEN_NOT_FOUND:
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case SIP_GET_DEST_REFUSED:
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default:
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@@ -27244,7 +27284,12 @@ static int handle_common_options(struct ast_flags *flags, struct ast_flags *mask
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res = 1;
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} else if (!strcasecmp(v->name, "allowoverlap")) {
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ast_set_flag(&mask[1], SIP_PAGE2_ALLOWOVERLAP);
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ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_ALLOWOVERLAP);
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ast_clear_flag(&flags[1], SIP_PAGE2_ALLOWOVERLAP);
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if (ast_true(v->value)) {
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ast_set_flag(&flags[1], SIP_PAGE2_ALLOWOVERLAP_YES);
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} else if (!strcasecmp(v->value, "dtmf")){
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ast_set_flag(&flags[1], SIP_PAGE2_ALLOWOVERLAP_DTMF);
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}
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} else if (!strcasecmp(v->name, "allowsubscribe")) {
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ast_set_flag(&mask[1], SIP_PAGE2_ALLOWSUBSCRIBE);
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ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_ALLOWSUBSCRIBE);
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@@ -28459,6 +28504,7 @@ static int reload_config(enum channelreloadreason reason)
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sipdebug &= sip_debug_console;
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ast_clear_flag(&global_flags[0], AST_FLAGS_ALL);
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ast_clear_flag(&global_flags[1], AST_FLAGS_ALL);
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ast_clear_flag(&global_flags[2], AST_FLAGS_ALL);
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/* Reset IP addresses */
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ast_sockaddr_parse(&bindaddr, "0.0.0.0:0", 0);
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@@ -28534,7 +28580,7 @@ static int reload_config(enum channelreloadreason reason)
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sip_cfg.allowtransfer = TRANSFER_OPENFORALL; /* Merrily accept all transfers by default */
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sip_cfg.rtautoclear = 120;
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ast_set_flag(&global_flags[1], SIP_PAGE2_ALLOWSUBSCRIBE); /* Default for all devices: TRUE */
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ast_set_flag(&global_flags[1], SIP_PAGE2_ALLOWOVERLAP); /* Default for all devices: TRUE */
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ast_set_flag(&global_flags[1], SIP_PAGE2_ALLOWOVERLAP_YES); /* Default for all devices: Yes */
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sip_cfg.peer_rtupdate = TRUE;
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global_dynamic_exclude_static = 0; /* Exclude static peers */
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sip_cfg.tcp_enabled = FALSE;
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@@ -303,46 +303,52 @@
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a second page of flags (for flags[1] */
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/*@{*/
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/* realtime flags */
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#define SIP_PAGE2_RTCACHEFRIENDS (1 << 0) /*!< GP: Should we keep RT objects in memory for extended time? */
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#define SIP_PAGE2_RTAUTOCLEAR (1 << 1) /*!< GP: Should we clean memory from peers after expiry? */
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#define SIP_PAGE2_RPID_UPDATE (1 << 2)
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#define SIP_PAGE2_Q850_REASON (1 << 3) /*!< DP: Get/send cause code via Reason header */
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#define SIP_PAGE2_SYMMETRICRTP (1 << 4) /*!< GDP: Whether symmetric RTP is enabled or not */
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#define SIP_PAGE2_STATECHANGEQUEUE (1 << 5) /*!< D: Unsent state pending change exists */
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#define SIP_PAGE2_CONNECTLINEUPDATE_PEND (1 << 6)
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#define SIP_PAGE2_RPID_IMMEDIATE (1 << 7)
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#define SIP_PAGE2_RPORT_PRESENT (1 << 8) /*!< Was rport received in the Via header? */
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#define SIP_PAGE2_PREFERRED_CODEC (1 << 9) /*!< GDP: Only respond with single most preferred joint codec */
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#define SIP_PAGE2_VIDEOSUPPORT (1 << 10) /*!< DP: Video supported if offered? */
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#define SIP_PAGE2_TEXTSUPPORT (1 << 11) /*!< GDP: Global text enable */
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#define SIP_PAGE2_ALLOWSUBSCRIBE (1 << 12) /*!< GP: Allow subscriptions from this peer? */
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#define SIP_PAGE2_ALLOWOVERLAP (1 << 13) /*!< DP: Allow overlap dialing ? */
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#define SIP_PAGE2_SUBSCRIBEMWIONLY (1 << 14) /*!< GP: Only issue MWI notification if subscribed to */
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#define SIP_PAGE2_IGNORESDPVERSION (1 << 15) /*!< GDP: Ignore the SDP session version number we receive and treat all sessions as new */
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#define SIP_PAGE2_RTCACHEFRIENDS (1 << 0) /*!< GP: Should we keep RT objects in memory for extended time? */
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#define SIP_PAGE2_RTAUTOCLEAR (1 << 1) /*!< GP: Should we clean memory from peers after expiry? */
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#define SIP_PAGE2_RPID_UPDATE (1 << 2)
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#define SIP_PAGE2_Q850_REASON (1 << 3) /*!< DP: Get/send cause code via Reason header */
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#define SIP_PAGE2_SYMMETRICRTP (1 << 4) /*!< GDP: Whether symmetric RTP is enabled or not */
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#define SIP_PAGE2_STATECHANGEQUEUE (1 << 5) /*!< D: Unsent state pending change exists */
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#define SIP_PAGE2_CONNECTLINEUPDATE_PEND (1 << 6)
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#define SIP_PAGE2_RPID_IMMEDIATE (1 << 7)
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#define SIP_PAGE2_RPORT_PRESENT (1 << 8) /*!< Was rport received in the Via header? */
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#define SIP_PAGE2_PREFERRED_CODEC (1 << 9) /*!< GDP: Only respond with single most preferred joint codec */
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#define SIP_PAGE2_VIDEOSUPPORT (1 << 10) /*!< DP: Video supported if offered? */
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#define SIP_PAGE2_TEXTSUPPORT (1 << 11) /*!< GDP: Global text enable */
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#define SIP_PAGE2_ALLOWSUBSCRIBE (1 << 12) /*!< GP: Allow subscriptions from this peer? */
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#define SIP_PAGE2_T38SUPPORT (3 << 16) /*!< GDP: T.38 Fax Support */
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#define SIP_PAGE2_T38SUPPORT_UDPTL (1 << 16) /*!< GDP: T.38 Fax Support (no error correction) */
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#define SIP_PAGE2_T38SUPPORT_UDPTL_FEC (2 << 16) /*!< GDP: T.38 Fax Support (FEC error correction) */
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#define SIP_PAGE2_T38SUPPORT_UDPTL_REDUNDANCY (3 << 16) /*!< GDP: T.38 Fax Support (redundancy error correction) */
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#define SIP_PAGE2_ALLOWOVERLAP (3 << 13) /*!< DP: Allow overlap dialing ? */
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#define SIP_PAGE2_ALLOWOVERLAP_NO (0 << 13) /*!< No, terminate with 404 Not found */
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#define SIP_PAGE2_ALLOWOVERLAP_YES (1 << 13) /*!< Yes, using the 484 Address Incomplete response */
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#define SIP_PAGE2_ALLOWOVERLAP_DTMF (2 << 13) /*!< Yes, using the DTMF transmission through Early Media */
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#define SIP_PAGE2_ALLOWOVERLAP_SPARE (3 << 13) /*!< Spare (reserved for another dialling transmission mechanisms like KPML) */
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#define SIP_PAGE2_CALL_ONHOLD (3 << 18) /*!< D: Call hold states: */
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#define SIP_PAGE2_CALL_ONHOLD_ACTIVE (1 << 18) /*!< D: Active hold */
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#define SIP_PAGE2_CALL_ONHOLD_ONEDIR (2 << 18) /*!< D: One directional hold */
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#define SIP_PAGE2_CALL_ONHOLD_INACTIVE (3 << 18) /*!< D: Inactive hold */
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#define SIP_PAGE2_SUBSCRIBEMWIONLY (1 << 15) /*!< GP: Only issue MWI notification if subscribed to */
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#define SIP_PAGE2_IGNORESDPVERSION (1 << 16) /*!< GDP: Ignore the SDP session version number we receive and treat all sessions as new */
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#define SIP_PAGE2_RFC2833_COMPENSATE (1 << 20) /*!< DP: Compensate for buggy RFC2833 implementations */
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#define SIP_PAGE2_BUGGY_MWI (1 << 21) /*!< DP: Buggy CISCO MWI fix */
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#define SIP_PAGE2_DIALOG_ESTABLISHED (1 << 22) /*!< 29: Has a dialog been established? */
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#define SIP_PAGE2_T38SUPPORT (3 << 17) /*!< GDP: T.38 Fax Support */
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#define SIP_PAGE2_T38SUPPORT_UDPTL (1 << 17) /*!< GDP: T.38 Fax Support (no error correction) */
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#define SIP_PAGE2_T38SUPPORT_UDPTL_FEC (2 << 17) /*!< GDP: T.38 Fax Support (FEC error correction) */
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#define SIP_PAGE2_T38SUPPORT_UDPTL_REDUNDANCY (3 << 17) /*!< GDP: T.38 Fax Support (redundancy error correction) */
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#define SIP_PAGE2_FAX_DETECT (3 << 23) /*!< DP: Fax Detection support */
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#define SIP_PAGE2_FAX_DETECT_CNG (1 << 23) /*!< DP: Fax Detection support - detect CNG in audio */
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#define SIP_PAGE2_FAX_DETECT_T38 (2 << 23) /*!< DP: Fax Detection support - detect T.38 reinvite from peer */
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#define SIP_PAGE2_FAX_DETECT_BOTH (3 << 23) /*!< DP: Fax Detection support - detect both */
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#define SIP_PAGE2_CALL_ONHOLD (3 << 19) /*!< D: Call hold states: */
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#define SIP_PAGE2_CALL_ONHOLD_ACTIVE (1 << 19) /*!< D: Active hold */
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#define SIP_PAGE2_CALL_ONHOLD_ONEDIR (2 << 19) /*!< D: One directional hold */
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#define SIP_PAGE2_CALL_ONHOLD_INACTIVE (3 << 19) /*!< D: Inactive hold */
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||||
|
||||
#define SIP_PAGE2_UDPTL_DESTINATION (1 << 25) /*!< DP: Use source IP of RTP as destination if NAT is enabled */
|
||||
#define SIP_PAGE2_VIDEOSUPPORT_ALWAYS (1 << 26) /*!< DP: Always set up video, even if endpoints don't support it */
|
||||
#define SIP_PAGE2_HAVEPEERCONTEXT (1 << 27) /*< Are we associated with a configured peer context? */
|
||||
#define SIP_PAGE2_USE_SRTP (1 << 28) /*!< DP: Whether we should offer (only) SRTP */
|
||||
#define SIP_PAGE2_RFC2833_COMPENSATE (1 << 21) /*!< DP: Compensate for buggy RFC2833 implementations */
|
||||
#define SIP_PAGE2_BUGGY_MWI (1 << 22) /*!< DP: Buggy CISCO MWI fix */
|
||||
#define SIP_PAGE2_DIALOG_ESTABLISHED (1 << 23) /*!< 29: Has a dialog been established? */
|
||||
|
||||
#define SIP_PAGE2_FAX_DETECT (3 << 24) /*!< DP: Fax Detection support */
|
||||
#define SIP_PAGE2_FAX_DETECT_CNG (1 << 24) /*!< DP: Fax Detection support - detect CNG in audio */
|
||||
#define SIP_PAGE2_FAX_DETECT_T38 (2 << 24) /*!< DP: Fax Detection support - detect T.38 reinvite from peer */
|
||||
#define SIP_PAGE2_FAX_DETECT_BOTH (3 << 24) /*!< DP: Fax Detection support - detect both */
|
||||
|
||||
#define SIP_PAGE2_UDPTL_DESTINATION (1 << 26) /*!< DP: Use source IP of RTP as destination if NAT is enabled */
|
||||
#define SIP_PAGE2_VIDEOSUPPORT_ALWAYS (1 << 27) /*!< DP: Always set up video, even if endpoints don't support it */
|
||||
#define SIP_PAGE2_HAVEPEERCONTEXT (1 << 28) /*< Are we associated with a configured peer context? */
|
||||
#define SIP_PAGE2_USE_SRTP (1 << 29) /*!< DP: Whether we should offer (only) SRTP */
|
||||
|
||||
#define SIP_PAGE2_FLAGS_TO_COPY \
|
||||
(SIP_PAGE2_ALLOWSUBSCRIBE | SIP_PAGE2_ALLOWOVERLAP | SIP_PAGE2_IGNORESDPVERSION | \
|
||||
@@ -466,7 +472,7 @@ enum sip_auth_type {
|
||||
|
||||
/*! \brief Result from get_destination function */
|
||||
enum sip_get_dest_result {
|
||||
SIP_GET_DEST_PICKUP_EXTEN_FOUND = 1,
|
||||
SIP_GET_DEST_EXTEN_MATCHMORE = 1,
|
||||
SIP_GET_DEST_EXTEN_FOUND = 0,
|
||||
SIP_GET_DEST_EXTEN_NOT_FOUND = -1,
|
||||
SIP_GET_DEST_REFUSED = -2,
|
||||
|
Reference in New Issue
Block a user