[svn-commits] oej: trunk r216841 - /trunk/channels/chan_sip.c
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Mon Sep 7 11:31:38 CDT 2009
Author: oej
Date: Mon Sep 7 11:31:36 2009
New Revision: 216841
URL: http://svn.asterisk.org/svn-view/asterisk?view=rev&rev=216841
Log:
Move capability into sip_cfg. While at it, make sure we reset it at channel reload.
Modified:
trunk/channels/chan_sip.c
Modified: trunk/channels/chan_sip.c
URL: http://svn.asterisk.org/svn-view/asterisk/trunk/channels/chan_sip.c?view=diff&rev=216841&r1=216840&r2=216841
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Mon Sep 7 11:31:36 2009
@@ -1130,6 +1130,7 @@
#define DEFAULT_SDPSESSION "Asterisk PBX" /*!< Default SDP session name, (s=) header unless re-defined in sip.conf */
#define DEFAULT_SDPOWNER "root" /*!< Default SDP username field in (o=) header unless re-defined in sip.conf */
#define DEFAULT_ENGINE "asterisk" /*!< Default RTP engine to use for sessions */
+#define DEFAULT_CAPABILITY (AST_FORMAT_ULAW | AST_FORMAT_ALAW | AST_FORMAT_GSM | AST_FORMAT_H263);
#endif
/*@}*/
@@ -1197,6 +1198,7 @@
char default_context[AST_MAX_CONTEXT];
char default_subscribecontext[AST_MAX_CONTEXT];
struct ast_ha *contact_ha; /*! \brief Global list of addresses dynamic peers are not allowed to use */
+ int capability; /*!< Supported codecs */
};
static struct sip_settings sip_cfg; /*!< SIP configuration data.
@@ -1236,9 +1238,6 @@
static int global_qualify_gap; /*!< Time between our group of peer pokes */
static int global_qualify_peers; /*!< Number of peers to poke at a given time */
-
-/*! \brief Codecs that we support by default: */
-static int global_capability = AST_FORMAT_ULAW | AST_FORMAT_ALAW | AST_FORMAT_GSM | AST_FORMAT_H263;
static enum st_mode global_st_mode; /*!< Mode of operation for Session-Timers */
static enum st_refresher global_st_refresher; /*!< Session-Timer refresher */
@@ -6690,9 +6689,9 @@
video = i->capability & AST_FORMAT_VIDEO_MASK;
text = i->capability & AST_FORMAT_TEXT_MASK;
} else {
- what = global_capability; /* Global codec support */
- video = global_capability & AST_FORMAT_VIDEO_MASK;
- text = global_capability & AST_FORMAT_TEXT_MASK;
+ what = sip_cfg.capability; /* Global codec support */
+ video = sip_cfg.capability & AST_FORMAT_VIDEO_MASK;
+ text = sip_cfg.capability & AST_FORMAT_TEXT_MASK;
}
/* Set the native formats for audio and merge in video */
@@ -7231,7 +7230,7 @@
/* Assign default music on hold class */
ast_string_field_set(p, mohinterpret, default_mohinterpret);
ast_string_field_set(p, mohsuggest, default_mohsuggest);
- p->capability = global_capability;
+ p->capability = sip_cfg.capability;
p->allowtransfer = sip_cfg.allowtransfer;
if ((ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833) ||
(ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO))
@@ -16233,7 +16232,7 @@
ast_cli(a->fd, "\nGlobal Signalling Settings:\n");
ast_cli(a->fd, "---------------------------\n");
ast_cli(a->fd, " Codecs: ");
- ast_getformatname_multiple(codec_buf, sizeof(codec_buf) -1, global_capability);
+ ast_getformatname_multiple(codec_buf, sizeof(codec_buf) -1, sip_cfg.capability);
ast_cli(a->fd, "%s\n", codec_buf);
ast_cli(a->fd, " Codec Order: ");
print_codec_to_cli(a->fd, &default_prefs);
@@ -23526,7 +23525,7 @@
*/
format &= AST_FORMAT_AUDIO_MASK;
if (!format) {
- ast_log(LOG_NOTICE, "Asked to get a channel of unsupported format %s while capability is %s\n", ast_getformatname(oldformat), ast_getformatname(global_capability));
+ ast_log(LOG_NOTICE, "Asked to get a channel of unsupported format %s while capability is %s\n", ast_getformatname(oldformat), ast_getformatname(sip_cfg.capability));
*cause = AST_CAUSE_BEARERCAPABILITY_NOTAVAIL; /* Can't find codec to connect to host */
return NULL;
}
@@ -24037,7 +24036,7 @@
ast_string_field_set(peer, engine, default_engine);
peer->addr.sin_family = AF_INET;
peer->defaddr.sin_family = AF_INET;
- peer->capability = global_capability;
+ peer->capability = sip_cfg.capability;
peer->maxcallbitrate = default_maxcallbitrate;
peer->rtptimeout = global_rtptimeout;
peer->rtpholdtimeout = global_rtpholdtimeout;
@@ -24806,6 +24805,7 @@
/* Reset channel settings to default before re-configuring */
sip_cfg.allow_external_domains = DEFAULT_ALLOW_EXT_DOM; /* Allow external invites */
sip_cfg.regcontext[0] = '\0';
+ sip_cfg.capability = DEFAULT_CAPABILITY;
sip_cfg.regextenonqualify = DEFAULT_REGEXTENONQUALIFY;
sip_cfg.notifyringing = DEFAULT_NOTIFYRINGING;
sip_cfg.notifycid = DEFAULT_NOTIFYCID;
@@ -25161,11 +25161,11 @@
externrefresh = 10;
}
} else if (!strcasecmp(v->name, "allow")) {
- int error = ast_parse_allow_disallow(&default_prefs, &global_capability, v->value, TRUE);
+ int error = ast_parse_allow_disallow(&default_prefs, &sip_cfg.capability, v->value, TRUE);
if (error)
ast_log(LOG_WARNING, "Codec configuration errors found in line %d : %s = %s\n", v->lineno, v->name, v->value);
} else if (!strcasecmp(v->name, "disallow")) {
- int error = ast_parse_allow_disallow(&default_prefs, &global_capability, v->value, FALSE);
+ int error = ast_parse_allow_disallow(&default_prefs, &sip_cfg.capability, v->value, FALSE);
if (error)
ast_log(LOG_WARNING, "Codec configuration errors found in line %d : %s = %s\n", v->lineno, v->name, v->value);
} else if (!strcasecmp(v->name, "preferred_codec_only")) {
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