diff options
author | markster <markster@f38db490-d61c-443f-a65b-d21fe96a405b> | 2003-09-30 23:03:57 +0000 |
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committer | markster <markster@f38db490-d61c-443f-a65b-d21fe96a405b> | 2003-09-30 23:03:57 +0000 |
commit | ce62ce8278d62887bc6e0a6be5c01f9c52a549d7 (patch) | |
tree | 33160c369c7ced799585cbd5b3934eea901f5d57 | |
parent | 0ff38e8c0320cafc703eaf36509951dc454e6cec (diff) |
Add sayunixtime, chan_sip updates for codec negotiation
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@1589 f38db490-d61c-443f-a65b-d21fe96a405b
-rwxr-xr-x | apps/Makefile | 2 | ||||
-rwxr-xr-x | apps/app_sayunixtime.c | 117 | ||||
-rwxr-xr-x | channels/chan_sip.c | 4 |
3 files changed, 120 insertions, 3 deletions
diff --git a/apps/Makefile b/apps/Makefile index 16850288f..0e89a7af6 100755 --- a/apps/Makefile +++ b/apps/Makefile @@ -24,7 +24,7 @@ APPS=app_dial.so app_playback.so app_voicemail.so app_directory.so app_intercom. app_authenticate.so app_softhangup.so app_lookupblacklist.so \ app_waitforring.so app_privacy.so app_db.so app_chanisavail.so \ app_enumlookup.so app_voicemail2.so app_transfer.so app_setcidnum.so app_cdr.so \ - app_hasnewvoicemail.so + app_hasnewvoicemail.so app_sayunixtime.so #APPS+=app_sql_postgres.so #APPS+=app_sql_odbc.so diff --git a/apps/app_sayunixtime.c b/apps/app_sayunixtime.c new file mode 100755 index 000000000..6a6fa6219 --- /dev/null +++ b/apps/app_sayunixtime.c @@ -0,0 +1,117 @@ +/* + * Asterisk -- A telephony toolkit for Linux. + * + * SayUnixTime application + * + * Copyright (c) 2003 Tilghman Lesher. All rights reserved. + * + * Tilghman Lesher <app_sayunixtime__200309@the-tilghman.com> + * + * This code is in the public domain. + * + */ + +#include <asterisk/file.h> +#include <asterisk/logger.h> +#include <asterisk/options.h> +#include <asterisk/channel.h> +#include <asterisk/pbx.h> +#include <asterisk/module.h> +#include <asterisk/say.h> +#include <stdio.h> +#include <stdlib.h> +#include <unistd.h> +#include <string.h> + + +static char *tdesc = "Say time"; + +static char *app_sayunixtime = "SayUnixTime"; + +static char *sayunixtime_synopsis = "Says a specified time in a custom format"; + +static char *sayunixtime_descrip = +"SayUnixTime([unixtime][|[timezone][|format]])\n" +" unixtime: time, in seconds since Jan 1, 1970. May be negative.\n" +" defaults to now.\n" +" timezone: timezone, see /usr/share/zoneinfo for a list.\n" +" defaults to machine default.\n" +" format: a format the time is to be said in. See voicemail.conf.\n" +" defaults to \"ABdY 'digits/at' IMp\"\n" +" Returns 0 or -1 on hangup.\n"; + +STANDARD_LOCAL_USER; + +LOCAL_USER_DECL; + +static int sayunixtime_exec(struct ast_channel *chan, void *data) +{ + int res=0; + struct localuser *u; + char *s,*zone=NULL,*timec; + time_t unixtime; + char *format = "ABdY 'digits/at' IMp"; + struct timeval tv; + + LOCAL_USER_ADD(u); + + gettimeofday(&tv,NULL); + unixtime = (time_t)tv.tv_sec; + + if (data) { + s = data; + s = strdupa(s); + if (s) { + timec = strsep(&s,"|"); + if ((timec) && (*timec != '\0')) { + long timein; + if (sscanf(timec,"%ld",&timein) == 1) { + unixtime = (time_t)timein; + } + } + if (s) { + zone = strsep(&s,"|"); + if (zone && (*zone == '\0')) + zone = NULL; + if (s) { + format = s; + } + } else { + ast_log(LOG_ERROR, "Out of memory error\n"); + } + } + } + + res = ast_say_date_with_format(chan, unixtime, AST_DIGIT_ANY, chan->language, format, zone); + + LOCAL_USER_REMOVE(u); + return res; +} + +int unload_module(void) +{ + STANDARD_HANGUP_LOCALUSERS; + return ast_unregister_application(app_sayunixtime); +} + +int load_module(void) +{ + return ast_register_application(app_sayunixtime, sayunixtime_exec, sayunixtime_synopsis, sayunixtime_descrip); +} + +char *description(void) +{ + return tdesc; +} + +int usecount(void) +{ + int res; + STANDARD_USECOUNT(res); + return res; +} + +char *key() +{ + return ASTERISK_GPL_KEY; +} diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 38af7956f..d5d59e062 100755 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -2423,7 +2423,7 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p, struct ast_rtp * /* Start by sending our preferred codecs */ cur = prefs; while(cur) { - if (p->capability & cur->codec) { + if (p->jointcapability & cur->codec) { if (sipdebug) ast_verbose("Answering with preferred capability %d\n", cur->codec); codec = ast_rtp_lookup_code(p->rtp, 1, cur->codec); @@ -2445,7 +2445,7 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p, struct ast_rtp * } /* Now send any other common codecs, and non-codec formats: */ for (x = 1; x <= AST_FORMAT_MAX_AUDIO; x <<= 1) { - if ((p->capability & x) && !(alreadysent & x)) { + if ((p->jointcapability & x) && !(alreadysent & x)) { if (sipdebug) ast_verbose("Answering with capability %d\n", x); codec = ast_rtp_lookup_code(p->rtp, 1, x); |