[svn-commits] oej: branch oej/earl-grey-sip2cause-configurable-trunk r377019 - in /team/oej...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Mon Dec 3 08:51:20 CST 2012
Author: oej
Date: Mon Dec 3 08:51:16 2012
New Revision: 377019
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=377019
Log:
Adding trunk version of Earl Grey.
Oh what a lovely cup of tea in the afternoon.
Added:
team/oej/earl-grey-sip2cause-configurable-trunk/README.earl-grey (with props)
team/oej/earl-grey-sip2cause-configurable-trunk/channels/sip/include/sip2cause.h (with props)
team/oej/earl-grey-sip2cause-configurable-trunk/channels/sip/sip2cause.c (with props)
team/oej/earl-grey-sip2cause-configurable-trunk/configs/sip2cause.conf.sample (with props)
Modified:
team/oej/earl-grey-sip2cause-configurable-trunk/channels/chan_sip.c
team/oej/earl-grey-sip2cause-configurable-trunk/channels/sip/include/sip_utils.h
Added: team/oej/earl-grey-sip2cause-configurable-trunk/README.earl-grey
URL: http://svnview.digium.com/svn/asterisk/team/oej/earl-grey-sip2cause-configurable-trunk/README.earl-grey?view=auto&rev=377019
==============================================================================
--- team/oej/earl-grey-sip2cause-configurable-trunk/README.earl-grey (added)
+++ team/oej/earl-grey-sip2cause-configurable-trunk/README.earl-grey Mon Dec 3 08:51:16 2012
@@ -1,0 +1,32 @@
+Edvina AB
+Olle E. Johansson
+
+
+Project start: 2012-11-28
+
+
+
+Earl Grey - configuring sip2cause tables
+==========================================
+
+Asterisk internally use ISDN cause codes. In the SIP channel, we do have two
+tables - one for translating from an ISDN cause code at hangup, to a SIP response
+code and one for translating the other way. In some edge cases, not very common,
+one wants to modify these tables.
+
+This branch will make the table a configurable option and add a new configuration
+file, sip2cause.conf, that is not needed in normal usage of Asterisk.
+
+When do you need this?
+======================
+If you operate a SIP2ISDN gateway and need to fine-tune conversion between ISDN
+cause codes and Asterisk SIP channel responses, you need a dynamic way to
+handle the conversion.
+
+If you operate a SIP server outside of Asterisk, you might want to add extra
+non-standard SIP response codes that is converted to a separate cause code
+that you handle in the dial plan.
+
+Please note the warning in sip2cause.conf.sample. Changing the conversion table
+can lead to all kinds of damage to your Asterisk installation. In most cases,
+no animals will be hurt.
Propchange: team/oej/earl-grey-sip2cause-configurable-trunk/README.earl-grey
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svn:eol-style = native
Propchange: team/oej/earl-grey-sip2cause-configurable-trunk/README.earl-grey
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svn:keywords = Author Date Id Revision
Propchange: team/oej/earl-grey-sip2cause-configurable-trunk/README.earl-grey
------------------------------------------------------------------------------
svn:mime-type = text/plain
Modified: team/oej/earl-grey-sip2cause-configurable-trunk/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/earl-grey-sip2cause-configurable-trunk/channels/chan_sip.c?view=diff&rev=377019&r1=377018&r2=377019
==============================================================================
--- team/oej/earl-grey-sip2cause-configurable-trunk/channels/chan_sip.c (original)
+++ team/oej/earl-grey-sip2cause-configurable-trunk/channels/chan_sip.c Mon Dec 3 08:51:16 2012
@@ -286,6 +286,7 @@
#include "sip/include/sip_utils.h"
#include "sip/include/srtp.h"
#include "sip/include/sdp_crypto.h"
+#include "sip/include/sip2cause.h"
#include "asterisk/ccss.h"
#include "asterisk/xml.h"
#include "sip/include/dialog.h"
@@ -644,6 +645,8 @@
static const char config[] = "sip.conf"; /*!< Main configuration file */
static const char notify_config[] = "sip_notify.conf"; /*!< Configuration file for sending Notify with CLI commands to reconfigure or reboot phones */
+static const char sip2cause_config[] = "sip2cause.conf"; /*!< Configuration file for configuration of sip2cause conversions */
+static struct ast_config *sip2cause = NULL; /*!< Configuration file for sip2cause conversions */
/*! \brief Readable descriptions of device states.
* \note Should be aligned to above table as index */
@@ -761,8 +764,9 @@
static char default_zone[MAX_TONEZONE_COUNTRY]; /*!< Default tone zone for channels created from the SIP driver */
static unsigned int default_transports; /*!< Default Transports (enum sip_transport) that are acceptable */
static unsigned int default_primary_transport; /*!< Default primary Transport (enum sip_transport) for outbound connections to devices */
-
-static struct sip_settings sip_cfg; /*!< SIP configuration data.
+static int private_sip2cause; /*!< Indication of private sip2cause conversion tables */
+
+static struct sip_settings sip_cfg; /*!< SIP configuration data.
\note in the future we could have multiple of these (per domain, per device group etc) */
/*!< use this macro when ast_uri_decode is dependent on pedantic checking to be on. */
@@ -6843,178 +6847,6 @@
ast_debug(3, "Destroying SIP dialog %s\n", p->callid);
__sip_destroy(p, TRUE, TRUE);
return NULL;
-}
-
-/*! \brief Convert SIP hangup causes to Asterisk hangup causes */
-int hangup_sip2cause(int cause)
-{
- /* Possible values taken from causes.h */
-
- switch(cause) {
- case 401: /* Unauthorized */
- return AST_CAUSE_CALL_REJECTED;
- case 403: /* Not found */
- return AST_CAUSE_CALL_REJECTED;
- case 404: /* Not found */
- return AST_CAUSE_UNALLOCATED;
- case 405: /* Method not allowed */
- return AST_CAUSE_INTERWORKING;
- case 407: /* Proxy authentication required */
- return AST_CAUSE_CALL_REJECTED;
- case 408: /* No reaction */
- return AST_CAUSE_NO_USER_RESPONSE;
- case 409: /* Conflict */
- return AST_CAUSE_NORMAL_TEMPORARY_FAILURE;
- case 410: /* Gone */
- return AST_CAUSE_NUMBER_CHANGED;
- case 411: /* Length required */
- return AST_CAUSE_INTERWORKING;
- case 413: /* Request entity too large */
- return AST_CAUSE_INTERWORKING;
- case 414: /* Request URI too large */
- return AST_CAUSE_INTERWORKING;
- case 415: /* Unsupported media type */
- return AST_CAUSE_INTERWORKING;
- case 420: /* Bad extension */
- return AST_CAUSE_NO_ROUTE_DESTINATION;
- case 480: /* No answer */
- return AST_CAUSE_NO_ANSWER;
- case 481: /* No answer */
- return AST_CAUSE_INTERWORKING;
- case 482: /* Loop detected */
- return AST_CAUSE_INTERWORKING;
- case 483: /* Too many hops */
- return AST_CAUSE_NO_ANSWER;
- case 484: /* Address incomplete */
- return AST_CAUSE_INVALID_NUMBER_FORMAT;
- case 485: /* Ambiguous */
- return AST_CAUSE_UNALLOCATED;
- case 486: /* Busy everywhere */
- return AST_CAUSE_BUSY;
- case 487: /* Request terminated */
- return AST_CAUSE_INTERWORKING;
- case 488: /* No codecs approved */
- return AST_CAUSE_BEARERCAPABILITY_NOTAVAIL;
- case 491: /* Request pending */
- return AST_CAUSE_INTERWORKING;
- case 493: /* Undecipherable */
- return AST_CAUSE_INTERWORKING;
- case 500: /* Server internal failure */
- return AST_CAUSE_FAILURE;
- case 501: /* Call rejected */
- return AST_CAUSE_FACILITY_REJECTED;
- case 502:
- return AST_CAUSE_DESTINATION_OUT_OF_ORDER;
- case 503: /* Service unavailable */
- return AST_CAUSE_CONGESTION;
- case 504: /* Gateway timeout */
- return AST_CAUSE_RECOVERY_ON_TIMER_EXPIRE;
- case 505: /* SIP version not supported */
- return AST_CAUSE_INTERWORKING;
- case 600: /* Busy everywhere */
- return AST_CAUSE_USER_BUSY;
- case 603: /* Decline */
- return AST_CAUSE_CALL_REJECTED;
- case 604: /* Does not exist anywhere */
- return AST_CAUSE_UNALLOCATED;
- case 606: /* Not acceptable */
- return AST_CAUSE_BEARERCAPABILITY_NOTAVAIL;
- default:
- if (cause < 500 && cause >= 400) {
- /* 4xx class error that is unknown - someting wrong with our request */
- return AST_CAUSE_INTERWORKING;
- } else if (cause < 600 && cause >= 500) {
- /* 5xx class error - problem in the remote end */
- return AST_CAUSE_CONGESTION;
- } else if (cause < 700 && cause >= 600) {
- /* 6xx - global errors in the 4xx class */
- return AST_CAUSE_INTERWORKING;
- }
- return AST_CAUSE_NORMAL;
- }
- /* Never reached */
- return 0;
-}
-
-/*! \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)
-{
- switch (cause) {
- case AST_CAUSE_UNALLOCATED: /* 1 */
- case AST_CAUSE_NO_ROUTE_DESTINATION: /* 3 IAX2: Can't find extension in context */
- case AST_CAUSE_NO_ROUTE_TRANSIT_NET: /* 2 */
- return "404 Not Found";
- case AST_CAUSE_CONGESTION: /* 34 */
- case AST_CAUSE_SWITCH_CONGESTION: /* 42 */
- return "503 Service Unavailable";
- case AST_CAUSE_NO_USER_RESPONSE: /* 18 */
- return "408 Request Timeout";
- case AST_CAUSE_NO_ANSWER: /* 19 */
- case AST_CAUSE_UNREGISTERED: /* 20 */
- return "480 Temporarily unavailable";
- case AST_CAUSE_CALL_REJECTED: /* 21 */
- return "403 Forbidden";
- case AST_CAUSE_NUMBER_CHANGED: /* 22 */
- return "410 Gone";
- case AST_CAUSE_NORMAL_UNSPECIFIED: /* 31 */
- return "480 Temporarily unavailable";
- case AST_CAUSE_INVALID_NUMBER_FORMAT:
- return "484 Address incomplete";
- case AST_CAUSE_USER_BUSY:
- return "486 Busy here";
- case AST_CAUSE_FAILURE:
- return "500 Server internal failure";
- case AST_CAUSE_FACILITY_REJECTED: /* 29 */
- return "501 Not Implemented";
- case AST_CAUSE_CHAN_NOT_IMPLEMENTED:
- return "503 Service Unavailable";
- /* Used in chan_iax2 */
- case AST_CAUSE_DESTINATION_OUT_OF_ORDER:
- return "502 Bad Gateway";
- case AST_CAUSE_BEARERCAPABILITY_NOTAVAIL: /* Can't find codec to connect to host */
- return "488 Not Acceptable Here";
- case AST_CAUSE_INTERWORKING: /* Unspecified Interworking issues */
- return "500 Network error";
-
- case AST_CAUSE_NOTDEFINED:
- default:
- ast_debug(1, "AST hangup cause %d (no match found in SIP)\n", cause);
- return NULL;
- }
-
- /* Never reached */
- return 0;
}
static int reinvite_timeout(const void *data)
@@ -20569,6 +20401,7 @@
ast_cli(a->fd, " MWI NOTIFY mime type: %s\n", default_notifymime);
ast_cli(a->fd, " DNS SRV lookup: %s\n", AST_CLI_YESNO(sip_cfg.srvlookup));
ast_cli(a->fd, " Pedantic SIP support: %s\n", AST_CLI_YESNO(sip_cfg.pedanticsipchecking));
+ ast_cli(a->fd, " Private SIP2cause: %s\n", AST_CLI_YESNO(private_sip2cause));
ast_cli(a->fd, " Reg. min duration %d secs\n", min_expiry);
ast_cli(a->fd, " Reg. max duration: %d secs\n", max_expiry);
ast_cli(a->fd, " Reg. default duration: %d secs\n", default_expiry);
@@ -32210,6 +32043,17 @@
ast_log(LOG_ERROR, "Contents of %s are invalid and cannot be parsed.\n", notify_config);
notify_types = NULL;
}
+ private_sip2cause = 0;
+ if ((sip2cause = ast_config_load(sip2cause_config, config_flags)) == CONFIG_STATUS_FILEINVALID) {
+ ast_log(LOG_ERROR, "Contents of %s are invalid and cannot be parsed.\n", sip2cause_config);
+ sip2cause = NULL;
+ } else {
+ if(sip2cause != NULL) {
+ private_sip2cause = sip2cause_load(sip2cause);
+ }
+ /* Now clean up */
+ ast_config_destroy(sip2cause);
+ }
/* Done, tell the manager */
manager_event(EVENT_FLAG_SYSTEM, "ChannelReload", "ChannelType: SIP\r\nReloadReason: %s\r\nRegistry_Count: %d\r\nPeer_Count: %d\r\n", channelreloadreason2txt(reason), registry_count, peer_count);
@@ -34183,6 +34027,7 @@
sip_reloadreason = CHANNEL_MODULE_LOAD;
can_parse_xml = sip_is_xml_parsable();
+ sip2cause_init();
if (reload_config(sip_reloadreason)) { /* Load the configuration from sip.conf */
return AST_MODULE_LOAD_DECLINE;
}
@@ -34479,6 +34324,7 @@
ast_cc_agent_unregister(&sip_cc_agent_callbacks);
sip_reqresp_parser_exit();
+ sip2cause_free();
sip_unregister_tests();
ast_format_cap_destroy(sip_tech.capabilities);
Added: team/oej/earl-grey-sip2cause-configurable-trunk/channels/sip/include/sip2cause.h
URL: http://svnview.digium.com/svn/asterisk/team/oej/earl-grey-sip2cause-configurable-trunk/channels/sip/include/sip2cause.h?view=auto&rev=377019
==============================================================================
--- team/oej/earl-grey-sip2cause-configurable-trunk/channels/sip/include/sip2cause.h (added)
+++ team/oej/earl-grey-sip2cause-configurable-trunk/channels/sip/include/sip2cause.h Mon Dec 3 08:51:16 2012
@@ -1,0 +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 sip2cause header file
+ */
+
+#include "sip.h"
+
+#ifndef _SIP_CAUSE_H
+#define _SIP_CAUSE_H
+
+/*! \brief Convert SIP response code to ISDN or Asterisk-specific cause code */
+int hangup_sip2cause(int cause);
+
+/*! \brief Convert ISDN or Asterisk-specific cause code to SIP response code */
+char *hangup_cause2sip(int cause);
+
+/*! \brief Initialized sip2cause tables */
+void sip2cause_init(void);
+
+/*! \brief Free sip2cause tables */
+void sip2cause_free(void);
+
+/*! \brief Load configuration */
+int sip2cause_load(struct ast_config *s2c_config);
+
+#endif /* defined(_SIP_CAUSE_H) */
Propchange: team/oej/earl-grey-sip2cause-configurable-trunk/channels/sip/include/sip2cause.h
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: team/oej/earl-grey-sip2cause-configurable-trunk/channels/sip/include/sip2cause.h
------------------------------------------------------------------------------
svn:keywords = Author Date Id Revision
Propchange: team/oej/earl-grey-sip2cause-configurable-trunk/channels/sip/include/sip2cause.h
------------------------------------------------------------------------------
svn:mime-type = text/plain
Modified: team/oej/earl-grey-sip2cause-configurable-trunk/channels/sip/include/sip_utils.h
URL: http://svnview.digium.com/svn/asterisk/team/oej/earl-grey-sip2cause-configurable-trunk/channels/sip/include/sip_utils.h?view=diff&rev=377019&r1=377018&r2=377019
==============================================================================
--- team/oej/earl-grey-sip2cause-configurable-trunk/channels/sip/include/sip_utils.h (original)
+++ team/oej/earl-grey-sip2cause-configurable-trunk/channels/sip/include/sip_utils.h Mon Dec 3 08:51:16 2012
@@ -42,44 +42,6 @@
*/
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);
-
/*! \brief Return a string describing the force_rport value for the given flags */
const char *force_rport_string(struct ast_flags *flags);
Added: team/oej/earl-grey-sip2cause-configurable-trunk/channels/sip/sip2cause.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/earl-grey-sip2cause-configurable-trunk/channels/sip/sip2cause.c?view=auto&rev=377019
==============================================================================
--- team/oej/earl-grey-sip2cause-configurable-trunk/channels/sip/sip2cause.c (added)
+++ team/oej/earl-grey-sip2cause-configurable-trunk/channels/sip/sip2cause.c Mon Dec 3 08:51:16 2012
@@ -1,0 +1,281 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2012, Digium, inc and Edvina AB
+ *
+ * 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-to-ISDN cause code conversions
+ *
+ * \author Olle E. Johansson <oej at edvina.net>
+ */
+
+/*** MODULEINFO
+ <support_level>core</support_level>
+ ***/
+
+#include "asterisk.h"
+#include "asterisk/causes.h"
+#include "asterisk/strings.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+#include "include/sip2cause.h"
+
+/*! \brief structure for conversion between ISDN and SIP codes */
+struct sip2causestruct {
+ int sip; /*!< SIP code (200-699) - no provisional codes */
+ int cause; /*!< ISDN cause code */
+ char reason[64]; /*!< SIP reason text, like "486 Busy", "404 Inte min domän" or "500 Que?" */
+ int private; /*!< If 1 = private extension */
+ struct sip2causestruct *next; /*!< Pointer to next entry */
+};
+
+/*! \brief Main structure for tables, including default values */
+struct sip2causetable {
+ struct sip2causestruct *table;
+ int defaultcode;
+ char *defaultreason;
+};
+
+/*! \brief Actual table for sip => ISDN lookups */
+struct sip2causetable sip2causelookup;
+
+/*! \brief Actual table for ISDN => sip lookups */
+struct sip2causetable cause2siplookup;
+
+/*! \brief Add conversion entry to table */
+static struct sip2causestruct *newsip2cause(int sip, int cause, const char *reason, int private, struct sip2causestruct *next)
+{
+ struct sip2causestruct *s2c = ast_calloc(1, sizeof(struct sip2causestruct));
+
+ if (!s2c) {
+ return NULL;
+ }
+ s2c->sip = sip;
+ s2c->cause = cause;
+ ast_copy_string(s2c->reason, reason, sizeof(s2c->reason));
+ s2c->next = next;
+ ast_debug(4, "SIP2CAUSE adding %d %s <=> %d (%s) \n", sip, reason, cause, ast_cause2str(cause));
+ return(s2c);
+ }
+
+/*! \brief Initialize structure with default values */
+void sip2cause_init(void)
+{
+ /* Initialize table for SIP => ISDN codes */
+ sip2causelookup.table = newsip2cause(401, /* Unauthorized */ AST_CAUSE_CALL_REJECTED, "", 0, NULL);
+ sip2causelookup.table = newsip2cause(403, /* Not found */ AST_CAUSE_CALL_REJECTED, "", 0, sip2causelookup.table);
+ sip2causelookup.table = newsip2cause(404, /* Not found */ AST_CAUSE_UNALLOCATED, "", 0, sip2causelookup.table);
+ sip2causelookup.table = newsip2cause(405, /* Method not allowed */ AST_CAUSE_INTERWORKING, "", 0, sip2causelookup.table);
+ sip2causelookup.table = newsip2cause(407, /* Proxy authentication required */ AST_CAUSE_CALL_REJECTED, "", 0, sip2causelookup.table);
+ sip2causelookup.table = newsip2cause(408, /* No reaction */ AST_CAUSE_NO_USER_RESPONSE, "", 0, sip2causelookup.table);
+ sip2causelookup.table = newsip2cause(409, /* Conflict */ AST_CAUSE_NORMAL_TEMPORARY_FAILURE, "", 0, sip2causelookup.table);
+ sip2causelookup.table = newsip2cause(410, /* Gone */ AST_CAUSE_NUMBER_CHANGED, "", 0, sip2causelookup.table);
+ sip2causelookup.table = newsip2cause(411, /* Length required */ AST_CAUSE_INTERWORKING, "", 0, sip2causelookup.table);
+ sip2causelookup.table = newsip2cause(413, /* Request entity too large */ AST_CAUSE_INTERWORKING, "", 0, sip2causelookup.table);
+ sip2causelookup.table = newsip2cause(414, /* Request URI too large */ AST_CAUSE_INTERWORKING, "", 0, sip2causelookup.table);
+ sip2causelookup.table = newsip2cause(415, /* Unsupported media type */ AST_CAUSE_INTERWORKING, "", 0, sip2causelookup.table);
+ sip2causelookup.table = newsip2cause(420, /* Bad extension */ AST_CAUSE_NO_ROUTE_DESTINATION, "", 0, sip2causelookup.table);
+ sip2causelookup.table = newsip2cause(480, /* No answer */ AST_CAUSE_NO_ANSWER, "", 0, sip2causelookup.table);
+ sip2causelookup.table = newsip2cause(481, /* No answer */ AST_CAUSE_INTERWORKING, "", 0, sip2causelookup.table);
+ sip2causelookup.table = newsip2cause(482, /* Loop detected */ AST_CAUSE_INTERWORKING, "", 0, sip2causelookup.table);
+ sip2causelookup.table = newsip2cause(483, /* Too many hops */ AST_CAUSE_NO_ANSWER, "", 0, sip2causelookup.table);
+ sip2causelookup.table = newsip2cause(484, /* Address incomplete */ AST_CAUSE_INVALID_NUMBER_FORMAT, "", 0, sip2causelookup.table);
+ sip2causelookup.table = newsip2cause(485, /* Ambiguous */ AST_CAUSE_UNALLOCATED, "", 0, sip2causelookup.table);
+ sip2causelookup.table = newsip2cause(486, /* Busy everywhere */ AST_CAUSE_BUSY, "", 0, sip2causelookup.table);
+ sip2causelookup.table = newsip2cause(487, /* Request terminated */ AST_CAUSE_INTERWORKING, "", 0, sip2causelookup.table);
+ sip2causelookup.table = newsip2cause(488, /* No codecs approved */ AST_CAUSE_BEARERCAPABILITY_NOTAVAIL, "", 0, sip2causelookup.table);
+ sip2causelookup.table = newsip2cause(491, /* Request pending */ AST_CAUSE_INTERWORKING, "", 0, sip2causelookup.table);
+ sip2causelookup.table = newsip2cause(493, /* Undecipherable */ AST_CAUSE_INTERWORKING, "", 0, sip2causelookup.table);
+ sip2causelookup.table = newsip2cause(500, /* Server internal failure */ AST_CAUSE_FAILURE, "", 0, sip2causelookup.table);
+ sip2causelookup.table = newsip2cause(501, /* Call rejected */ AST_CAUSE_FACILITY_REJECTED, "", 0, sip2causelookup.table);
+ sip2causelookup.table = newsip2cause(502, AST_CAUSE_DESTINATION_OUT_OF_ORDER, "", 0, sip2causelookup.table);
+ sip2causelookup.table = newsip2cause(503, /* Service unavailable */ AST_CAUSE_CONGESTION, "", 0, sip2causelookup.table);
+ sip2causelookup.table = newsip2cause(504, /* Gateway timeout */ AST_CAUSE_RECOVERY_ON_TIMER_EXPIRE, "", 0, sip2causelookup.table);
+ sip2causelookup.table = newsip2cause(505, /* SIP version not supported */ AST_CAUSE_INTERWORKING, "", 0, sip2causelookup.table);
+ sip2causelookup.table = newsip2cause(600, /* Busy everywhere */ AST_CAUSE_USER_BUSY, "", 0, sip2causelookup.table);
+ sip2causelookup.table = newsip2cause(603, /* Decline */ AST_CAUSE_CALL_REJECTED, "", 0, sip2causelookup.table);
+ sip2causelookup.table = newsip2cause(604, /* Does not exist anywhere */ AST_CAUSE_UNALLOCATED, "", 0, sip2causelookup.table);
+ sip2causelookup.table = newsip2cause(606, /* Not acceptable */ AST_CAUSE_BEARERCAPABILITY_NOTAVAIL, "", 0, sip2causelookup.table);
+
+ /* Add the reverse table */
+ cause2siplookup.table = newsip2cause(404, AST_CAUSE_UNALLOCATED, "404 Not Found", 0, NULL);
+ cause2siplookup.table = newsip2cause(404, AST_CAUSE_NO_ROUTE_DESTINATION, "404 Not Found", 0, cause2siplookup.table);
+ cause2siplookup.table = newsip2cause(404, AST_CAUSE_NO_ROUTE_TRANSIT_NET, "404 Not Found", 0, cause2siplookup.table);
+ cause2siplookup.table = newsip2cause(503, AST_CAUSE_CONGESTION, "503 Service Unavailable", 0, cause2siplookup.table);
+ cause2siplookup.table = newsip2cause(503, AST_CAUSE_SWITCH_CONGESTION, "503 Service Unavailable", 0, cause2siplookup.table);
+ cause2siplookup.table = newsip2cause(408, AST_CAUSE_NO_USER_RESPONSE, "408 Request Timeout", 0, cause2siplookup.table);
+ cause2siplookup.table = newsip2cause(480, AST_CAUSE_NO_ANSWER, "480 Temporarily unavailable", 0, cause2siplookup.table);
+ cause2siplookup.table = newsip2cause(480, AST_CAUSE_UNREGISTERED, "480 Temporarily unavailable", 0, cause2siplookup.table);
+ cause2siplookup.table = newsip2cause(403, AST_CAUSE_CALL_REJECTED, "403 Forbidden", 0, cause2siplookup.table);
+ cause2siplookup.table = newsip2cause(410, AST_CAUSE_NUMBER_CHANGED, "410 Gone", 0, cause2siplookup.table);
+ cause2siplookup.table = newsip2cause(480, AST_CAUSE_NORMAL_UNSPECIFIED, "480 Temporarily unavailable", 0, cause2siplookup.table);
+ cause2siplookup.table = newsip2cause(484, AST_CAUSE_INVALID_NUMBER_FORMAT, "484 Address Incomplete", 0, cause2siplookup.table);
+ cause2siplookup.table = newsip2cause(486, AST_CAUSE_USER_BUSY, "486 Busy here", 0, cause2siplookup.table);
+ cause2siplookup.table = newsip2cause(500, AST_CAUSE_FAILURE, "500 Server internal failure", 0, cause2siplookup.table);
+ cause2siplookup.table = newsip2cause(501, AST_CAUSE_FACILITY_REJECTED, "501 Not implemented", 0, cause2siplookup.table);
+ cause2siplookup.table = newsip2cause(503, AST_CAUSE_CHAN_NOT_IMPLEMENTED, "503 Service unavailable", 0, cause2siplookup.table);
+ cause2siplookup.table = newsip2cause(502, AST_CAUSE_DESTINATION_OUT_OF_ORDER, "502 Bad gateway", 0, cause2siplookup.table);
+ cause2siplookup.table = newsip2cause(488, AST_CAUSE_BEARERCAPABILITY_NOTAVAIL, "488 Not Acceptable Here", 0, cause2siplookup.table);
+ cause2siplookup.table = newsip2cause(500, AST_CAUSE_INTERWORKING, "500 Network error", 0, cause2siplookup.table);
+
+}
+
+
+/*! \brief Make sure we free the cause code list from memory */
+void sip2cause_free(void)
+{
+ struct sip2causestruct *s2c = sip2causelookup.table;
+ while (s2c) {
+ struct sip2causestruct *next = s2c->next;
+ ast_free(s2c);
+ s2c = next;
+ }
+ s2c = cause2siplookup.table;
+ while (s2c) {
+ struct sip2causestruct *next = s2c->next;
+ ast_free(s2c);
+ s2c = next;
+ }
+}
+
+
+/*! \brief Convert SIP hangup causes to Asterisk hangup causes */
+int hangup_sip2cause(int sipcode)
+{
+ struct sip2causestruct *s2c = sip2causelookup.table;
+ while (s2c) {
+ if (s2c->sip == sipcode) {
+ ast_debug(1, "SIP2CAUSE returning %d (%s) based on SIP code %d [%s]\n", s2c->cause, ast_cause2str(s2c->cause), sipcode, s2c->private ? "config" : "default");
+ return s2c->cause;
+ }
+ s2c = s2c->next;
+ }
+
+ /* Possible values taken from causes.h */
+
+ if (sipcode < 500 && sipcode >= 400) {
+ /* 4xx class error that is unknown - someting wrong with our request */
+ ast_debug(4, "SIP2CAUSE returning default %d (%s) based on SIP code %d\n", AST_CAUSE_INTERWORKING, ast_cause2str(AST_CAUSE_INTERWORKING), sipcode);
+ return AST_CAUSE_INTERWORKING;
+ } else if (sipcode < 600 && sipcode >= 500) {
+ ast_debug(4, "SIP2CAUSE returning default %d (%s) based on SIP code %d\n", AST_CAUSE_CONGESTION, ast_cause2str(AST_CAUSE_CONGESTION), sipcode);
+ /* 5xx class error - problem in the remote end */
+ return AST_CAUSE_CONGESTION;
+ } else if (sipcode < 700 && sipcode >= 600) {
+ ast_debug(4, "SIP2CAUSE returning default %d (%s) based on SIP code %d\n", AST_CAUSE_INTERWORKING, ast_cause2str(AST_CAUSE_INTERWORKING), sipcode);
+ /* 6xx - global errors in the 4xx class */
+ return AST_CAUSE_INTERWORKING;
+ }
+ ast_debug(4, "SIP2CAUSE returning default %d (%s) based on SIP code %d\n", s2c->cause, ast_cause2str(s2c->cause), sipcode);
+ return AST_CAUSE_NORMAL;
+}
+
+/*! \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
+*/
+char *hangup_cause2sip(int cause)
+{
+ struct sip2causestruct *s2c = cause2siplookup.table;
+ while (s2c) {
+ if (s2c->cause == cause) {
+ ast_debug(4, "cause2sip returning %s based on ISDN cause %d - %s [%s]\n", s2c->reason, cause, ast_cause2str(cause), s2c->private ? "config" : "default");
+ return s2c->reason;
+ }
+ s2c = s2c->next;
+ }
+ ast_debug(1, "AST hangup cause %d (no match found in SIP)\n", cause);
+ return NULL;
+}
+
+/*! \brief Load configuration
+
+- Check if SIP code is valid
+- Check if we can parse the cause code using functions in channel.c
+
+*/
+int sip2cause_load(struct ast_config *s2c_config)
+{
+ struct ast_variable *v;
+ int respcode;
+ int cause;
+ int number=0;
+
+ ast_debug(2, "AST sip2cause configuration parser");
+ for (v = ast_variable_browse(s2c_config, "sip2cause"); v; v = v->next) {
+ ast_debug(1, "====> SIP2cause ::: Name %s Value %s \n", v->name, v->value);
+ respcode = 42;
+ cause = 0;
+ number = sscanf(v->name, "%d", &respcode);
+ if (number != 1) {
+ ast_log(LOG_ERROR, "Unknown SIP response code format %s in sip2cause.conf section [sip2cause] Respcode %d Number %d\n", v->name, respcode, number);
+ continue;
+ }
+ if (respcode < 200 || respcode > 699) {
+ ast_log(LOG_ERROR, "Bad SIP response code: Asterisk cause code \'%s=>%s\' in sip2cause.conf section [sip2cause] \n", v->name, v->value);
+ continue;
+ }
+ if ((cause = ast_str2cause(v->value)) == -1) {
+ ast_log(LOG_ERROR, "Unknown Asterisk cause code %s in sip2cause.conf section [sip2cause] Cause %d\n", v->value, cause);
+ continue;
+ }
+ sip2causelookup.table = newsip2cause(respcode, cause, "", 1, sip2causelookup.table);
+ }
+ for (v = ast_variable_browse(s2c_config, "cause2sip"); v; v = v->next) {
+ ast_debug(1, "====> CAUSE2SIP ::: Name %s Value %s \n", v->name, v->value);
+ if ((cause = ast_str2cause(v->name)) == -1) {
+ ast_log(LOG_ERROR, "Unknown Asterisk cause code %s in sip2cause.conf section [cause2sip] \n", v->name);
+ continue;
+ }
+ if (sscanf(v->value, "%d ", &respcode) != 1) {
+ ast_log(LOG_ERROR, "Bad syntax: Asterisk cause code \'%s=>%s\' in sip2cause.conf section [cause2sip] \n", v->name, v->value);
+ continue;
+ }
+ if (respcode < 200 || respcode > 699) {
+ ast_log(LOG_ERROR, "Bad SIP response code: \'%s=>%s\' in sip2cause.conf section [cause2sip] \n", v->name, v->value);
+ continue;
+ }
+ cause2siplookup.table = newsip2cause(respcode, cause, v->value, 1, cause2siplookup.table);
+ }
+ return 1;
+}
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Added: team/oej/earl-grey-sip2cause-configurable-trunk/configs/sip2cause.conf.sample
URL: http://svnview.digium.com/svn/asterisk/team/oej/earl-grey-sip2cause-configurable-trunk/configs/sip2cause.conf.sample?view=auto&rev=377019
==============================================================================
--- team/oej/earl-grey-sip2cause-configurable-trunk/configs/sip2cause.conf.sample (added)
+++ team/oej/earl-grey-sip2cause-configurable-trunk/configs/sip2cause.conf.sample Mon Dec 3 08:51:16 2012
@@ -1,0 +1,24 @@
+;
+; Conversion table additions for the translation between SIP hangup codes and ISDN cause codes
+; ============================================================================================
+; This table is used to convert a negative SIP response to an ISDN cause code, that is sent
+; over the PBX call bridge to the calling party, or to convert a received ISDN hangup cause
+; from the called party to a SIP response. Changes will not affect how Asterisk treats a SIP
+; response, but how the other party reacts to it.
+;
+; For reference, check Asterisk source code include/asterisk/causes.h
+; and channels/sip/sip2cause.c
+;
+; Note also that this translation is not used in all cases in the SIP implementation in Asterisk.
+; Fixes are always welcome.
+;
+; !!!!!!!!!!!!!!!!!!!! READ THIS SERIOUSLY !
+; Note: By changing the default translation, you can seriously jeopardize your Asterisk server functionality.
+; Make sure you understand what you are doing, and that you are testing carefully.
+; In 99.99% of Asterisk installations, you will not need to add ANYTHING to this file.
+;
+[sip2cause]
+; 404 => USER_BUSY
+
+[cause2sip]
+; USER_BUSY => 403 Forbidden
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