Merge Russell's formatting patch (bug #3838)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@5234 f38db490-d61c-443f-a65b-d21fe96a405b
This commit is contained in:
markster
parent
2b84c4638e
commit
4a7019a16a
16
app.c
16
app.c
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@ -58,9 +58,9 @@ int ast_app_dtget(struct ast_channel *chan, const char *context, char *collect,
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timeout = 5;
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ts = ast_get_indication_tone(chan->zone,"dial");
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if (ts && ts->data[0]) {
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if (ts && ts->data[0])
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res = ast_playtones_start(chan, 0, ts->data, 0);
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} else
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else
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ast_log(LOG_NOTICE,"Huh....? no dial for indications?\n");
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for (x = strlen(collect); strlen(collect) < maxlen; ) {
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@ -105,8 +105,10 @@ int ast_app_getdata(struct ast_channel *c, char *prompt, char *s, int maxlen, in
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fto = c->pbx ? c->pbx->rtimeout * 1000 : 6000;
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to = c->pbx ? c->pbx->dtimeout * 1000 : 2000;
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if (timeout > 0) fto = to = timeout;
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if (timeout < 0) fto = to = 1000000000;
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if (timeout > 0)
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fto = to = timeout;
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if (timeout < 0)
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fto = to = 1000000000;
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res = ast_readstring(c, s, maxlen, to, fto, "#");
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return res;
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}
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@ -122,8 +124,10 @@ int ast_app_getdata_full(struct ast_channel *c, char *prompt, char *s, int maxle
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}
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fto = 6000;
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to = 2000;
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if (timeout > 0) fto = to = timeout;
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if (timeout < 0) fto = to = 1000000000;
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if (timeout > 0)
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fto = to = timeout;
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if (timeout < 0)
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fto = to = 1000000000;
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res = ast_readstring_full(c, s, maxlen, to, fto, "#", audiofd, ctrlfd);
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return res;
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}
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@ -44,7 +44,7 @@ static char *descrip =
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" or 'noanswer' to read digits even if the line is not up.\n"
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" attempts -- if greater than 1, that many attempts will be made in the \n"
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" event no data is entered.\n"
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" timeout -- if greater than 0, that value will override the default timeoft.\n\n"
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" timeout -- if greater than 0, that value will override the default timeout.\n\n"
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"Returns -1 on hangup or error and 0 otherwise.\n";
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STANDARD_LOCAL_USER;
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4
astmm.c
4
astmm.c
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@ -3,9 +3,9 @@
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*
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* Channel Variables
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*
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* Copyright (C) 2002, Mark Spencer
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* Copyright (C) 2002-2005, Mark Spencer
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*
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* Mark Spencer <markster@linux-support.net>
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* Mark Spencer <markster@digium.com>
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*
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* This program is free software, distributed under the terms of
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* the GNU General Public License
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3
cli.c
3
cli.c
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@ -377,7 +377,7 @@ static int handle_modlist(int fd, int argc, char *argv[])
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if (argc == 3)
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return RESULT_SHOWUSAGE;
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else if (argc >= 4) {
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if ( strcmp(argv[2],"like") )
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if (strcmp(argv[2],"like"))
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return RESULT_SHOWUSAGE;
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like = argv[3];
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}
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@ -732,6 +732,7 @@ static int handle_showchan(int fd, int argc, char *argv[])
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c->context, c->exten, c->priority, c->callgroup, c->pickupgroup, ( c->appl ? c->appl : "(N/A)" ),
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( c-> data ? (!ast_strlen_zero(c->data) ? c->data : "(Empty)") : "(None)"),
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(ast_test_flag(c, AST_FLAG_BLOCKING) ? c->blockproc : "(Not Blocking)"));
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if(pbx_builtin_serialize_variables(c,buf,sizeof(buf)))
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ast_cli(fd," Variables:\n%s\n",buf);
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if(c->cdr && ast_cdr_serialize_variables(c->cdr,buf, sizeof(buf), '=', '\n', 1))
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@ -330,20 +330,20 @@ ${UNIQUEID} * Current call unique identifier
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Various application variables
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-----------------------------
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${CURL} Resulting page content for curl()
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${ENUM} Result of application EnumLookup
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${CURL} * Resulting page content for curl()
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${ENUM} * Result of application EnumLookup
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${EXITCONTEXT} Context to exit to in IVR menu (app background())
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or in the RetryDial() application
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${GROUPCOUNT} Result from groupcount()
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${MONITOR} Set to "TRUE" if the channel is/has been monitored (app monitor())
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${GROUPCOUNT} * Result from groupcount()
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${MONITOR} * Set to "TRUE" if the channel is/has been monitored (app monitor())
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${MONITOR_EXEC} Application to execute after monitoring a call
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${MONITOR_EXEC_ARGS} Arguments to application
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${MONITOR_FILENAME} File for monitoring (recording) calls in queue
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${QUEUE_PRIO} Queue priority
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${RECORDED_FILE} Recorded file in record()
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${TALK_DETECED} Result from talkdetect()
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${RECORDED_FILE} * Recorded file in record()
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${TALK_DETECED} * Result from talkdetect()
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${TOUCH_MONITOR} The filename base to use with Touch Monitor (auto record)
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${TXTCIDNAME} Result of application TXTCIDName
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${TXTCIDNAME} * Result of application TXTCIDName
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${VPB_GETDTMF} chan_vpb
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The MeetMe Conference Bridge uses the following variables:
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@ -353,53 +353,47 @@ ${MEETME_RECORDINGFILE} Name of file for recording a conference with
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${MEETME_RECORDINGFORMAT} Format of file to be recorded
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${MEETME_EXIT_CONTEXT} Context for exit out of meetme meeting
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${MEETME_AGI_BACKGROUND} AGI script for Meetme (zap only)
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Meetme() sets the following variable:
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${MEETMESECS} Number of seconds a user participated in a MeetMe conference
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${MEETMESECS} * Number of seconds a user participated in a MeetMe conference
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The voicemail() application uses the following variables:
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---------------------------------------------------------
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${VM_CATEGORY} Sets voicemail category
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${VM_NAME} * Full name in voicemail
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${VM_DUR} * Voicemail duration
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${VM_MSGNUM} * Number of voicemail message in mailbox
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${VM_CALLERID} * Voicemail Caller ID (Person leaving vm)
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${VM_CIDNAME} * Voicemail Caller ID Name
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${VM_CIDNUM} * Voicemail Caller ID Number
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${VM_DATE} * Voicemail Date
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The following variables are set by voicemail()
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${VM_NAME} Full name in voicemail
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${VM_DUR} Voicemail duration
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${VM_MSGNUM} Number of voicemail message in mailbox
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${VM_CALLERID} Voicemail Caller ID (Person leaving vm)
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${VM_CIDNAME} Voicemail Caller ID Name
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${VM_CIDNUM} Voicemail Caller ID Number
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${VM_DATE} Voicemail Date
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The following variables are set by vmauthenticate()
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${AUTH_MAILBOX} Authenticated mailbox
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${AUTH_CONTEXT} Authenticated mailbox context
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${DIFF_DAY} Day difference (internal use)
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Dundi() uses the following variables
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The vmauthenticate() application uses the following variables:
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---------------------------------------------------------
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${DUNDTECH} Technology
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${DUNDDEST} Destination
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${AUTH_MAILBOX} * Authenticated mailbox
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${AUTH_CONTEXT} * Authenticated mailbox context
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DUNDiLookup() uses the following variables
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---------------------------------------------------------
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${DUNDTECH} * The Technology of the result from a call to DUNDiLookup()
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${DUNDDEST} * The Destination of the result from a call to DUNDiLookup()
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The Zaptel channel sets the following variables:
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---------------------------------------------------------
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${ANI2} The ANI2 Code provided by the network on the incoming call.
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${ANI2} * The ANI2 Code provided by the network on the incoming call.
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(ie, Code 29 identifies call as a Prison/Inmate Call)
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${CALLTYPE} Type of call (Speech, Digital, etc)
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${CALLEDTON} Type of number for incoming PRI extension
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${CALLTYPE} * Type of call (Speech, Digital, etc)
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${CALLEDTON} * Type of number for incoming PRI extension
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i.e. 0=unknown, 1=international, 2=domestic, 3=net_specific,
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4=subscriber, 6=abbreviated, 7=reserved
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${CALLINGSUBADDR} Called PRI Subaddress
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${FAXEXTEN} The extension called before being redirected to "fax"
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${PRIREDIRECTREASON} Reason for redirect, if a call was directed
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${CALLINGSUBADDR} * Called PRI Subaddress
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${FAXEXTEN} * The extension called before being redirected to "fax"
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${PRIREDIRECTREASON} * Reason for redirect, if a call was directed
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The SIP channel sets the following variables:
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---------------------------------------------------------
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${SIPCALLID} SIP Call-ID: header verbatim (for logging or CDR matching)
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${SIPDOMAIN} SIP destination domain of an inbound call (if appropriate)
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${SIPUSERAGENT} SIP user agent
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${SIPURI} SIP uri
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The SIP channel reads the following variable:
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${SIPCALLID} * SIP Call-ID: header verbatim (for logging or CDR matching)
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${SIPDOMAIN} * SIP destination domain of an inbound call (if appropriate)
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${SIPUSERAGENT} * SIP user agent
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${SIPURI} * SIP uri
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${SIP_CODEC} Set the SIP codec for a call
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The Agent channel uses the following variables:
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@ -410,20 +404,18 @@ ${AGENTGOODBYE} Sound file to use for "Good Bye" when agent logs out
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${AGENTACKCALL} Whether the agent should acknowledge the incoming call
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${AGENTAUTOLOGOFF} Auto logging off for an agent
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${AGENTWRAPUPTIME} Setting the time for wrapup between incoming calls
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${AGENTNUMBER} Agent number (username) set at login
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${AGENTSTATUS} Status of login ( fail | on | off )
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${AGENTEXTEN} Extension for logged in agent
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${AGENTNUMBER} * Agent number (username) set at login
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${AGENTSTATUS} * Status of login ( fail | on | off )
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${AGENTEXTEN} * Extension for logged in agent
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The Dial() application sets the following variables:
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The Dial() application uses the following variables:
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---------------------------------------------------------
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${DIALEDPEERNAME} Dialed peer name
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${DIALEDPEERNUMBER} Dialed peer number
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${DIALEDTIME} Time for the call (seconds)
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${ANSWEREDTIME} Time from dial to answer (seconds)
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${DIALSTATUS} Status of the call, one of:
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CHANUNAVAIL | CONGESTION | BUSY | NOANSWER | ANSWER | CANCEL
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The dial() application reads the following variables
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${DIALEDPEERNAME} * Dialed peer name
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${DIALEDPEERNUMBER} * Dialed peer number
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${DIALEDTIME} * Time for the call (seconds)
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${ANSWEREDTIME} * Time from dial to answer (seconds)
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${DIALSTATUS} * Status of the call, one of:
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(CHANUNAVAIL | CONGESTION | BUSY | NOANSWER | ANSWER | CANCEL)
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${LIMIT_PLAYAUDIO_CALLER} Soundfile for call limits
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${LIMIT_PLAYAUDIO_CALLEE} Soundfile for call limits
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${LIMIT_WARNING_FILE} Soundfile for call limits
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@ -432,19 +424,17 @@ ${LIMIT_CONNECT_FILE} Soundfile for call limits
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${OUTBOUND_GROUP} Default groups for peer channels (as in SetGroup)
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* See "show application dial" for more information
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The chanisavail() application sets the following variables:
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-----------------------------------------------------------
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${AVAILCHAN} return value
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${AVAILORIGCHAN} return value
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${AVAILSTATUS} Status of requested channel
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${AVAILCHAN} * the name of the available channel if one was found
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${AVAILORIGCHAN} * the canonical channel name that was used to create the channel
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${AVAILSTATUS} * Status of requested channel
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When using macros in the dialplan, these variables are available
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---------------------------------------------------------
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${MACRO_EXTEN} The calling extensions
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${MACRO_CONTEXT} The calling context
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${MACRO_PRIORITY} The calling priority
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${MACRO_EXTEN} * The calling extensions
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${MACRO_CONTEXT} * The calling context
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${MACRO_PRIORITY} * The calling priority
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${MACRO_OFFSET} Offset to add to priority at return from macro
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If you compile with OSP support in the SIP channel, these
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@ -1343,7 +1343,7 @@ static void *accept_thread(void *ignore)
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continue;
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}
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p = getprotobyname("tcp");
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if( p ) {
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if (p) {
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if( setsockopt(as, p->p_proto, TCP_NODELAY, (char *)&arg, sizeof(arg) ) < 0 ) {
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ast_log(LOG_WARNING, "Failed to set manager tcp connection to TCP_NODELAY mode: %s\n", strerror(errno));
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}
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