diff options
author | tilghman <tilghman@f38db490-d61c-443f-a65b-d21fe96a405b> | 2010-02-17 06:25:15 +0000 |
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committer | tilghman <tilghman@f38db490-d61c-443f-a65b-d21fe96a405b> | 2010-02-17 06:25:15 +0000 |
commit | eae21bd2ffbe4dd555d4034aa12db55bf2ebbe8c (patch) | |
tree | 71fd1d739915bdaf45c20dc0cd3fda632194f03b /channels/sip | |
parent | fe7b2347a46062eb0d5e2a0247a1f5cabdd515b7 (diff) |
Make all of the various rtpqos parameters in this branch available from the CHANNEL function.
Also includes a test for retrieving rtpqos parameters, including a NULL RTP
driver. Additionally, some further separation of the SIP internal API into
headers was necessary.
(closes issue #16652)
Reported by: kkm
Patches:
20100204__issue16652.diff.txt uploaded by tilghman (license 14)
Review: https://reviewboard.asterisk.org/r/501/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@247124 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'channels/sip')
-rw-r--r-- | channels/sip/dialplan_functions.c | 366 | ||||
-rw-r--r-- | channels/sip/include/config_parser.h | 2 | ||||
-rw-r--r-- | channels/sip/include/dialog.h | 75 | ||||
-rw-r--r-- | channels/sip/include/dialplan_functions.h | 41 | ||||
-rw-r--r-- | channels/sip/include/globals.h | 42 | ||||
-rw-r--r-- | channels/sip/include/sip_utils.h | 44 |
6 files changed, 567 insertions, 3 deletions
diff --git a/channels/sip/dialplan_functions.c b/channels/sip/dialplan_functions.c new file mode 100644 index 000000000..473199c71 --- /dev/null +++ b/channels/sip/dialplan_functions.c @@ -0,0 +1,366 @@ +/* + * Asterisk -- An open source telephony toolkit. + * + * Copyright (C) 2010, Digium, Inc. + * + * See http://www.asterisk.org for more information about + * the Asterisk project. Please do not directly contact + * any of the maintainers of this project for assistance; + * the project provides a web site, mailing lists and IRC + * channels for your use. + * + * This program is free software, distributed under the terms of + * the GNU General Public License Version 2. See the LICENSE file + * at the top of the source tree. + */ + +/*! + * \file + * \brief sip channel dialplan functions and unit tests + */ + +#include "asterisk.h" + +ASTERISK_FILE_VERSION(__FILE__, "$Revision$") + +#include <math.h> + +#include "asterisk/channel.h" +#include "asterisk/rtp_engine.h" +#include "asterisk/pbx.h" + +#include "include/sip.h" +#include "include/globals.h" +#include "include/dialog.h" +#include "include/dialplan_functions.h" +#include "include/sip_utils.h" + + +int sip_acf_channel_read(struct ast_channel *chan, const char *funcname, char *preparse, char *buf, size_t buflen) +{ + struct sip_pvt *p = chan->tech_pvt; + char *parse = ast_strdupa(preparse); + int res = 0; + AST_DECLARE_APP_ARGS(args, + AST_APP_ARG(param); + AST_APP_ARG(type); + AST_APP_ARG(field); + ); + AST_STANDARD_APP_ARGS(args, parse); + + /* Sanity check */ + if (!IS_SIP_TECH(chan->tech)) { + ast_log(LOG_ERROR, "Cannot call %s on a non-SIP channel\n", funcname); + return 0; + } + + memset(buf, 0, buflen); + + if (p == NULL) { + return -1; + } + + if (!strcasecmp(args.param, "peerip")) { + ast_copy_string(buf, p->sa.sin_addr.s_addr ? ast_inet_ntoa(p->sa.sin_addr) : "", buflen); + } else if (!strcasecmp(args.param, "recvip")) { + ast_copy_string(buf, p->recv.sin_addr.s_addr ? ast_inet_ntoa(p->recv.sin_addr) : "", buflen); + } else if (!strcasecmp(args.param, "from")) { + ast_copy_string(buf, p->from, buflen); + } else if (!strcasecmp(args.param, "uri")) { + ast_copy_string(buf, p->uri, buflen); + } else if (!strcasecmp(args.param, "useragent")) { + ast_copy_string(buf, p->useragent, buflen); + } else if (!strcasecmp(args.param, "peername")) { + ast_copy_string(buf, p->peername, buflen); + } else if (!strcasecmp(args.param, "t38passthrough")) { + ast_copy_string(buf, (p->t38.state == T38_DISABLED) ? "0" : "1", buflen); + } else if (!strcasecmp(args.param, "rtpdest")) { + struct sockaddr_in sin; + + if (ast_strlen_zero(args.type)) + args.type = "audio"; + + if (!strcasecmp(args.type, "audio")) + ast_rtp_instance_get_remote_address(p->rtp, &sin); + else if (!strcasecmp(args.type, "video")) + ast_rtp_instance_get_remote_address(p->vrtp, &sin); + else if (!strcasecmp(args.type, "text")) + ast_rtp_instance_get_remote_address(p->trtp, &sin); + else + return -1; + + snprintf(buf, buflen, "%s:%d", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port)); + } else if (!strcasecmp(args.param, "rtpqos")) { + struct ast_rtp_instance *rtp = NULL; + + if (ast_strlen_zero(args.type)) { + args.type = "audio"; + } + + if (!strcasecmp(args.type, "audio")) { + rtp = p->rtp; + } else if (!strcasecmp(args.type, "video")) { + rtp = p->vrtp; + } else if (!strcasecmp(args.type, "text")) { + rtp = p->trtp; + } else { + return -1; + } + + if (ast_strlen_zero(args.field) || !strcasecmp(args.field, "all")) { + char quality_buf[AST_MAX_USER_FIELD], *quality; + + if (!(quality = ast_rtp_instance_get_quality(rtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf)))) { + return -1; + } + + ast_copy_string(buf, quality_buf, buflen); + return res; + } else { + struct ast_rtp_instance_stats stats; + int i; + struct { + const char *name; + enum { INT, DBL } type; + union { + unsigned int *i4; + double *d8; + }; + } lookup[] = { + { "txcount", INT, { .i4 = &stats.txcount, }, }, + { "rxcount", INT, { .i4 = &stats.rxcount, }, }, + { "txjitter", INT, { .i4 = &stats.txjitter, }, }, + { "rxjitter", INT, { .i4 = &stats.rxjitter, }, }, + { "remote_maxjitter", DBL, { .d8 = &stats.remote_maxjitter, }, }, + { "remote_minjitter", DBL, { .d8 = &stats.remote_minjitter, }, }, + { "remote_normdevjitter", DBL, { .d8 = &stats.remote_normdevjitter, }, }, + { "remote_stdevjitter", DBL, { .d8 = &stats.remote_stdevjitter, }, }, + { "local_maxjitter", DBL, { .d8 = &stats.local_maxjitter, }, }, + { "local_minjitter", DBL, { .d8 = &stats.local_minjitter, }, }, + { "local_normdevjitter", DBL, { .d8 = &stats.local_normdevjitter, }, }, + { "local_stdevjitter", DBL, { .d8 = &stats.local_stdevjitter, }, }, + { "txploss", INT, { .i4 = &stats.txploss, }, }, + { "rxploss", INT, { .i4 = &stats.rxploss, }, }, + { "remote_maxrxploss", DBL, { .d8 = &stats.remote_maxrxploss, }, }, + { "remote_minrxploss", DBL, { .d8 = &stats.remote_minrxploss, }, }, + { "remote_normdevrxploss", DBL, { .d8 = &stats.remote_normdevrxploss, }, }, + { "remote_stdevrxploss", DBL, { .d8 = &stats.remote_stdevrxploss, }, }, + { "local_maxrxploss", DBL, { .d8 = &stats.local_maxrxploss, }, }, + { "local_minrxploss", DBL, { .d8 = &stats.local_minrxploss, }, }, + { "local_normdevrxploss", DBL, { .d8 = &stats.local_normdevrxploss, }, }, + { "local_stdevrxploss", DBL, { .d8 = &stats.local_stdevrxploss, }, }, + { "rtt", INT, { .i4 = &stats.rtt, }, }, + { "maxrtt", DBL, { .d8 = &stats.maxrtt, }, }, + { "minrtt", DBL, { .d8 = &stats.minrtt, }, }, + { "normdevrtt", DBL, { .d8 = &stats.normdevrtt, }, }, + { "stdevrtt", DBL, { .d8 = &stats.stdevrtt, }, }, + { "local_ssrc", INT, { .i4 = &stats.local_ssrc, }, }, + { "remote_ssrc", INT, { .i4 = &stats.remote_ssrc, }, }, + { NULL, }, + }; + + if (ast_rtp_instance_get_stats(rtp, &stats, AST_RTP_INSTANCE_STAT_ALL)) { + return -1; + } + + for (i = 0; !ast_strlen_zero(lookup[i].name); i++) { + if (!strcasecmp(args.field, lookup[i].name)) { + if (lookup[i].type == INT) { + snprintf(buf, buflen, "%u", *lookup[i].i4); + } else { + snprintf(buf, buflen, "%f", *lookup[i].d8); + } + return 0; + } + } + ast_log(LOG_WARNING, "Unrecognized argument '%s' to %s\n", preparse, funcname); + return -1; + } + } else { + res = -1; + } + return res; +} + +#ifdef TEST_FRAMEWORK +static int test_sip_rtpqos_1_new(struct ast_rtp_instance *instance, struct sched_context *sched, struct sockaddr_in *sin, void *data) +{ + /* Needed to pass sanity checks */ + ast_rtp_instance_set_data(instance, data); + return 0; +} + +static int test_sip_rtpqos_1_destroy(struct ast_rtp_instance *instance) +{ + /* Needed to pass sanity checks */ + return 0; +} + +static struct ast_frame *test_sip_rtpqos_1_read(struct ast_rtp_instance *instance, int rtcp) +{ + /* Needed to pass sanity checks */ + return &ast_null_frame; +} + +static int test_sip_rtpqos_1_write(struct ast_rtp_instance *instance, struct ast_frame *frame) +{ + /* Needed to pass sanity checks */ + return 0; +} + +static int test_sip_rtpqos_1_get_stat(struct ast_rtp_instance *instance, struct ast_rtp_instance_stats *stats, enum ast_rtp_instance_stat stat) +{ + struct ast_rtp_instance_stats *s = ast_rtp_instance_get_data(instance); + memcpy(stats, s, sizeof(*stats)); + return 0; +} + +AST_TEST_DEFINE(test_sip_rtpqos_1) +{ + int i, res = AST_TEST_PASS; + struct ast_rtp_engine test_engine = { + .name = "test", + .new = test_sip_rtpqos_1_new, + .destroy = test_sip_rtpqos_1_destroy, + .read = test_sip_rtpqos_1_read, + .write = test_sip_rtpqos_1_write, + .get_stat = test_sip_rtpqos_1_get_stat, + }; + struct sockaddr_in sin = { .sin_port = 31337, .sin_addr = { 4 * 16777216 + 3 * 65536 + 2 * 256 + 1 } }; + struct ast_rtp_instance_stats mine = { 0, }; + struct sip_pvt *p = NULL; + struct ast_channel *chan = NULL; + struct ast_str *varstr = NULL, *buffer = NULL; + struct { + const char *name; + enum { INT, DBL } type; + union { + unsigned int *i4; + double *d8; + }; + } lookup[] = { + { "txcount", INT, { .i4 = &mine.txcount, }, }, + { "rxcount", INT, { .i4 = &mine.rxcount, }, }, + { "txjitter", INT, { .i4 = &mine.txjitter, }, }, + { "rxjitter", INT, { .i4 = &mine.rxjitter, }, }, + { "remote_maxjitter", DBL, { .d8 = &mine.remote_maxjitter, }, }, + { "remote_minjitter", DBL, { .d8 = &mine.remote_minjitter, }, }, + { "remote_normdevjitter", DBL, { .d8 = &mine.remote_normdevjitter, }, }, + { "remote_stdevjitter", DBL, { .d8 = &mine.remote_stdevjitter, }, }, + { "local_maxjitter", DBL, { .d8 = &mine.local_maxjitter, }, }, + { "local_minjitter", DBL, { .d8 = &mine.local_minjitter, }, }, + { "local_normdevjitter", DBL, { .d8 = &mine.local_normdevjitter, }, }, + { "local_stdevjitter", DBL, { .d8 = &mine.local_stdevjitter, }, }, + { "txploss", INT, { .i4 = &mine.txploss, }, }, + { "rxploss", INT, { .i4 = &mine.rxploss, }, }, + { "remote_maxrxploss", DBL, { .d8 = &mine.remote_maxrxploss, }, }, + { "remote_minrxploss", DBL, { .d8 = &mine.remote_minrxploss, }, }, + { "remote_normdevrxploss", DBL, { .d8 = &mine.remote_normdevrxploss, }, }, + { "remote_stdevrxploss", DBL, { .d8 = &mine.remote_stdevrxploss, }, }, + { "local_maxrxploss", DBL, { .d8 = &mine.local_maxrxploss, }, }, + { "local_minrxploss", DBL, { .d8 = &mine.local_minrxploss, }, }, + { "local_normdevrxploss", DBL, { .d8 = &mine.local_normdevrxploss, }, }, + { "local_stdevrxploss", DBL, { .d8 = &mine.local_stdevrxploss, }, }, + { "rtt", INT, { .i4 = &mine.rtt, }, }, + { "maxrtt", DBL, { .d8 = &mine.maxrtt, }, }, + { "minrtt", DBL, { .d8 = &mine.minrtt, }, }, + { "normdevrtt", DBL, { .d8 = &mine.normdevrtt, }, }, + { "stdevrtt", DBL, { .d8 = &mine.stdevrtt, }, }, + { "local_ssrc", INT, { .i4 = &mine.local_ssrc, }, }, + { "remote_ssrc", INT, { .i4 = &mine.remote_ssrc, }, }, + { NULL, }, + }; + + switch (cmd) { + case TEST_INIT: + info->name = "test_sip_rtpqos"; + info->category = "channels/chan_sip/"; + info->summary = "Test retrieval of SIP RTP QOS stats"; + info->description = + "Verify values in the RTP instance structure can be accessed through the dialplan."; + return AST_TEST_NOT_RUN; + case TEST_EXECUTE: + break; + } + + ast_rtp_engine_register2(&test_engine, NULL); + /* Have to associate this with a SIP pvt and an ast_channel */ + if (!(p = sip_alloc(NULL, NULL, 0, SIP_NOTIFY, NULL))) { + res = AST_TEST_NOT_RUN; + goto done; + } + if (!(p->rtp = ast_rtp_instance_new("test", sched, &bindaddr, &mine))) { + res = AST_TEST_NOT_RUN; + goto done; + } + ast_rtp_instance_set_remote_address(p->rtp, &sin); + if (!(chan = ast_dummy_channel_alloc())) { + res = AST_TEST_NOT_RUN; + goto done; + } + chan->tech = &sip_tech; + chan->tech_pvt = p; + p->owner = chan; + + varstr = ast_str_create(16); + buffer = ast_str_create(16); + if (!varstr || !buffer) { + res = AST_TEST_NOT_RUN; + goto done; + } + + /* Populate "mine" with values, then retrieve them with the CHANNEL dialplan function */ + for (i = 0; !ast_strlen_zero(lookup[i].name); i++) { + ast_str_set(&varstr, 0, "${CHANNEL(rtpqos,audio,%s)}", lookup[i].name); + if (lookup[i].type == INT) { + int j; + char cmpstr[256]; + for (j = 1; j < 25; j++) { + *lookup[i].i4 = j; + ast_str_substitute_variables(&buffer, 0, chan, ast_str_buffer(varstr)); + snprintf(cmpstr, sizeof(cmpstr), "%d", j); + if (strcmp(cmpstr, ast_str_buffer(buffer))) { + res = AST_TEST_FAIL; + ast_test_status_update(test, "%s != %s != %s\n", ast_str_buffer(varstr), cmpstr, ast_str_buffer(buffer)); + break; + } + } + } else { + double j, cmpdbl = 0.0; + for (j = 1.0; j < 10.0; j += 0.3) { + *lookup[i].d8 = j; + ast_str_substitute_variables(&buffer, 0, chan, ast_str_buffer(varstr)); + if (sscanf(ast_str_buffer(buffer), "%lf", &cmpdbl) != 1 || fabs(j - cmpdbl > .05)) { + res = AST_TEST_FAIL; + ast_test_status_update(test, "%s != %f != %s\n", ast_str_buffer(varstr), j, ast_str_buffer(buffer)); + break; + } + } + } + } + +done: + ast_free(varstr); + ast_free(buffer); + + /* This unref will take care of destroying the channel, RTP instance, and SIP pvt */ + if (p) { + dialog_unref(p, "Destroy test object"); + } + ast_rtp_engine_unregister(&test_engine); + return res; +} +#endif + +/*! \brief SIP test registration */ +void sip_dialplan_function_register_tests(void) +{ + AST_TEST_REGISTER(test_sip_rtpqos_1); +} + +/*! \brief SIP test registration */ +void sip_dialplan_function_unregister_tests(void) +{ + AST_TEST_UNREGISTER(test_sip_rtpqos_1); +} + diff --git a/channels/sip/include/config_parser.h b/channels/sip/include/config_parser.h index 0b86188c6..76fefc2c6 100644 --- a/channels/sip/include/config_parser.h +++ b/channels/sip/include/config_parser.h @@ -43,7 +43,7 @@ int sip_parse_register_line(struct sip_registry *reg, int default_expiry, const */ int sip_parse_host(char *line, int lineno, char **hostname, int *portnum, enum sip_transport *transport); -/*! +/*! * \brief register config parsing tests */ void sip_config_parser_register_tests(void); diff --git a/channels/sip/include/dialog.h b/channels/sip/include/dialog.h new file mode 100644 index 000000000..554aa5e6a --- /dev/null +++ b/channels/sip/include/dialog.h @@ -0,0 +1,75 @@ +/* + * Asterisk -- An open source telephony toolkit. + * + * Copyright (C) 2010, Digium, Inc. + * + * See http://www.asterisk.org for more information about + * the Asterisk project. Please do not directly contact + * any of the maintainers of this project for assistance; + * the project provides a web site, mailing lists and IRC + * channels for your use. + * + * This program is free software, distributed under the terms of + * the GNU General Public License Version 2. See the LICENSE file + * at the top of the source tree. + */ + +/*! + * \file + * \brief sip dialog management header file + */ + +#include "sip.h" + +#ifndef _SIP_DIALOG_H +#define _SIP_DIALOG_H + +/*! \brief + * when we create or delete references, make sure to use these + * functions so we keep track of the refcounts. + * To simplify the code, we allow a NULL to be passed to dialog_unref(). + */ +#define dialog_ref(arg1,arg2) dialog_ref_debug((arg1),(arg2), __FILE__, __LINE__, __PRETTY_FUNCTION__) +#define dialog_unref(arg1,arg2) dialog_unref_debug((arg1),(arg2), __FILE__, __LINE__, __PRETTY_FUNCTION__) +struct sip_pvt *dialog_ref_debug(struct sip_pvt *p, char *tag, char *file, int line, const char *func); +struct sip_pvt *dialog_unref_debug(struct sip_pvt *p, char *tag, char *file, int line, const char *func); + +struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *sin, + int useglobal_nat, const int intended_method, struct sip_request *req); +void sip_scheddestroy(struct sip_pvt *p, int ms); +int sip_cancel_destroy(struct sip_pvt *p); + +/*! \brief Destroy SIP call structure. + * Make it return NULL so the caller can do things like + * foo = sip_destroy(foo); + * and reduce the chance of bugs due to dangling pointers. + */ +struct sip_pvt *sip_destroy(struct sip_pvt *p); + +/*! \brief Destroy SIP call structure. + * Make it return NULL so the caller can do things like + * foo = sip_destroy(foo); + * and reduce the chance of bugs due to dangling pointers. + */ +void __sip_destroy(struct sip_pvt *p, int lockowner, int lockdialoglist); +/*! + * \brief Unlink a dialog from the dialogs container, as well as any other places + * that it may be currently stored. + * + * \note A reference to the dialog must be held before calling this function, and this + * function does not release that reference. + */ +void *dialog_unlink_all(struct sip_pvt *dialog, int lockowner, int lockdialoglist); + +/*! \brief Acknowledges receipt of a packet and stops retransmission + * called with p locked*/ +int __sip_ack(struct sip_pvt *p, int seqno, int resp, int sipmethod); + +/*! \brief Pretend to ack all packets + * called with p locked */ +void __sip_pretend_ack(struct sip_pvt *p); + +/*! \brief Acks receipt of packet, keep it around (used for provisional responses) */ +int __sip_semi_ack(struct sip_pvt *p, int seqno, int resp, int sipmethod); + +#endif /* defined(_SIP_DIALOG_H) */ diff --git a/channels/sip/include/dialplan_functions.h b/channels/sip/include/dialplan_functions.h new file mode 100644 index 000000000..1600d4314 --- /dev/null +++ b/channels/sip/include/dialplan_functions.h @@ -0,0 +1,41 @@ +/* + * Asterisk -- An open source telephony toolkit. + * + * Copyright (C) 2010, Digium, Inc. + * + * See http://www.asterisk.org for more information about + * the Asterisk project. Please do not directly contact + * any of the maintainers of this project for assistance; + * the project provides a web site, mailing lists and IRC + * channels for your use. + * + * This program is free software, distributed under the terms of + * the GNU General Public License Version 2. See the LICENSE file + * at the top of the source tree. + */ + +/*! + * \file + * \brief SIP dialplan functions header file + */ + +#include "sip.h" + +#ifndef _SIP_DIALPLAN_FUNCTIONS_H +#define _SIP_DIALPLAN_FUNCTIONS_H + +/*! + * \brief Channel read dialplan function for SIP + */ +int sip_acf_channel_read(struct ast_channel *chan, const char *funcname, char *preparse, char *buf, size_t buflen); + +/*! + * \brief register dialplan function tests + */ +void sip_dialplan_function_register_tests(void); +/*! + * \brief unregister dialplan function tests + */ +void sip_dialplan_function_unregister_tests(void); + +#endif /* !defined(_SIP_DIALPLAN_FUNCTIONS_H) */ diff --git a/channels/sip/include/globals.h b/channels/sip/include/globals.h new file mode 100644 index 000000000..0d7131d87 --- /dev/null +++ b/channels/sip/include/globals.h @@ -0,0 +1,42 @@ +/* + * Asterisk -- An open source telephony toolkit. + * + * Copyright (C) 2010, Digium, Inc. + * + * See http://www.asterisk.org for more information about + * the Asterisk project. Please do not directly contact + * any of the maintainers of this project for assistance; + * the project provides a web site, mailing lists and IRC + * channels for your use. + * + * This program is free software, distributed under the terms of + * the GNU General Public License Version 2. See the LICENSE file + * at the top of the source tree. + */ + +/*! + * \file + * \brief sip global declaration header file + */ + +#include "sip.h" + +#ifndef _SIP_GLOBALS_H +#define _SIP_GLOBALS_H + +extern struct sockaddr_in bindaddr; /*!< UDP: The address we bind to */ +extern struct sched_context *sched; /*!< The scheduling context */ + +/*! \brief Definition of this channel for PBX channel registration */ +extern const struct ast_channel_tech sip_tech; + +/*! \brief This version of the sip channel tech has no send_digit_begin + * callback so that the core knows that the channel does not want + * DTMF BEGIN frames. + * The struct is initialized just before registering the channel driver, + * and is for use with channels using SIP INFO DTMF. + */ +extern struct ast_channel_tech sip_tech_info; + +#endif /* !defined(SIP_GLOBALS_H) */ + diff --git a/channels/sip/include/sip_utils.h b/channels/sip/include/sip_utils.h index 549f9392f..47297ab68 100644 --- a/channels/sip/include/sip_utils.h +++ b/channels/sip/include/sip_utils.h @@ -22,8 +22,11 @@ #ifndef _SIP_UTILS_H #define _SIP_UTILS_H -/*! - * \brief converts ascii port to int representation. +/* wrapper macro to tell whether t points to one of the sip_tech descriptors */ +#define IS_SIP_TECH(t) ((t) == &sip_tech || (t) == &sip_tech_info) + +/*! + * \brief converts ascii port to int representation. * * \arg pt[in] string that contains a port. * \arg standard[in] port to return in case the port string input is NULL @@ -40,4 +43,41 @@ unsigned int port_str2int(const char *pt, unsigned int standard); const char *find_closing_quote(const char *start, const char *lim); +/*! \brief Convert SIP hangup causes to Asterisk hangup causes */ +int hangup_sip2cause(int cause); + +/*! \brief Convert Asterisk hangup causes to SIP codes +\verbatim + Possible values from causes.h + AST_CAUSE_NOTDEFINED AST_CAUSE_NORMAL AST_CAUSE_BUSY + AST_CAUSE_FAILURE AST_CAUSE_CONGESTION AST_CAUSE_UNALLOCATED + + In addition to these, a lot of PRI codes is defined in causes.h + ...should we take care of them too ? + + Quote RFC 3398 + + ISUP Cause value SIP response + ---------------- ------------ + 1 unallocated number 404 Not Found + 2 no route to network 404 Not found + 3 no route to destination 404 Not found + 16 normal call clearing --- (*) + 17 user busy 486 Busy here + 18 no user responding 408 Request Timeout + 19 no answer from the user 480 Temporarily unavailable + 20 subscriber absent 480 Temporarily unavailable + 21 call rejected 403 Forbidden (+) + 22 number changed (w/o diagnostic) 410 Gone + 22 number changed (w/ diagnostic) 301 Moved Permanently + 23 redirection to new destination 410 Gone + 26 non-selected user clearing 404 Not Found (=) + 27 destination out of order 502 Bad Gateway + 28 address incomplete 484 Address incomplete + 29 facility rejected 501 Not implemented + 31 normal unspecified 480 Temporarily unavailable +\endverbatim +*/ +const char *hangup_cause2sip(int cause); + #endif |