[asterisk-commits] oej: branch oej/teapot-1.8 r384889 - in /team/oej/teapot-1.8/channels: ./ sip...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Mon Apr 8 09:06:24 CDT 2013
Author: oej
Date: Mon Apr 8 09:06:20 2013
New Revision: 384889
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=384889
Log:
Backing up
Modified:
team/oej/teapot-1.8/channels/chan_sip.c
team/oej/teapot-1.8/channels/sip/include/sip_utils.h
Modified: team/oej/teapot-1.8/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/teapot-1.8/channels/chan_sip.c?view=diff&rev=384889&r1=384888&r2=384889
==============================================================================
--- team/oej/teapot-1.8/channels/chan_sip.c (original)
+++ team/oej/teapot-1.8/channels/chan_sip.c Mon Apr 8 09:06:20 2013
@@ -271,7 +271,6 @@
#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"
@@ -579,8 +578,6 @@
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 */
@@ -687,7 +684,6 @@
static char default_notifymime[AST_MAX_EXTENSION]; /*!< Default MIME media type for MWI notify messages */
static char default_vmexten[AST_MAX_EXTENSION]; /*!< Default From Username on MWI updates */
static int default_qualify; /*!< Default Qualify= setting */
-static int private_sip2cause; /*!< Indication of private sip2cause conversion tables */
static char default_mohinterpret[MAX_MUSICCLASS]; /*!< Global setting for moh class to use when put on hold */
static char default_mohsuggest[MAX_MUSICCLASS]; /*!< Global setting for moh class to suggest when putting
* a bridged channel on hold */
@@ -6459,6 +6455,175 @@
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_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)
{
@@ -18679,7 +18844,6 @@
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);
@@ -29558,18 +29722,6 @@
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;
- sip2cause_init(); /* Initialize standard settings */
- } else {
- sip2cause_free(); /* If it's a reload, free existing settings */
- sip2cause_init(); /* Initialize standard settings */
- 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);
@@ -31847,7 +31999,6 @@
ast_cc_agent_unregister(&sip_cc_agent_callbacks);
sip_reqresp_parser_exit();
- sip2cause_free();
sip_unregister_tests();
return 0;
Modified: team/oej/teapot-1.8/channels/sip/include/sip_utils.h
URL: http://svnview.digium.com/svn/asterisk/team/oej/teapot-1.8/channels/sip/include/sip_utils.h?view=diff&rev=384889&r1=384888&r2=384889
==============================================================================
--- team/oej/teapot-1.8/channels/sip/include/sip_utils.h (original)
+++ team/oej/teapot-1.8/channels/sip/include/sip_utils.h Mon Apr 8 09:06:20 2013
@@ -42,4 +42,42 @@
*/
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
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