[svn-commits] kpfleming: branch 1.6.2 r210191 - in /branches/1.6.2: ./ channels/ configs/ c...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Mon Aug 3 15:58:53 CDT 2009
Author: kpfleming
Date: Mon Aug 3 15:58:48 2009
New Revision: 210191
URL: http://svn.asterisk.org/svn-view/asterisk?view=rev&rev=210191
Log:
Merged revisions 210190 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
........
r210190 | kpfleming | 2009-08-03 15:48:48 -0500 (Mon, 03 Aug 2009) | 11 lines
Rename 'canreinvite' option to 'directmedia', with backwards compatibility.
It is clear from multiple mailing list, forum, wiki and other sorts of posts
that users don't really understand the effects that the 'canreinvite' config
option actually has, and that in some cases they think that setting it to 'no'
will actually cause various other features (T.38, MOH, etc.) to not work properly,
when in fact this is not the case. This patch changes the proper name of the
option to what it should have been from the beginning ('directmedia'), but
preserves backwards compatibility for existing configurations.
........
Modified:
branches/1.6.2/ (props changed)
branches/1.6.2/UPGRADE.txt
branches/1.6.2/channels/chan_mgcp.c
branches/1.6.2/channels/chan_sip.c
branches/1.6.2/channels/chan_skinny.c
branches/1.6.2/configs/mgcp.conf.sample
branches/1.6.2/configs/res_ldap.conf.sample
branches/1.6.2/configs/sip.conf.sample
branches/1.6.2/configs/skinny.conf.sample
branches/1.6.2/contrib/scripts/realtime_pgsql.sql
branches/1.6.2/doc/chan_sip-perf-testing.txt
branches/1.6.2/doc/res_config_sqlite.txt
branches/1.6.2/doc/tex/phoneprov.tex
branches/1.6.2/main/rtp.c
Propchange: branches/1.6.2/
------------------------------------------------------------------------------
Binary property 'trunk-merged' - no diff available.
Modified: branches/1.6.2/UPGRADE.txt
URL: http://svn.asterisk.org/svn-view/asterisk/branches/1.6.2/UPGRADE.txt?view=diff&rev=210191&r1=210190&r2=210191
==============================================================================
--- branches/1.6.2/UPGRADE.txt (original)
+++ branches/1.6.2/UPGRADE.txt Mon Aug 3 15:58:48 2009
@@ -29,6 +29,17 @@
were made to solve significant T.38 interoperability problems between
Asterisk and various SIP/T.38 endpoints identified by many users of
Asterisk.
+
+* The 'canreinvite' option support by the SIP, MGCP and Skinny channel drivers
+ has been renamed to 'directmedia', to better reflect what it actually does.
+ In the case of SIP, there are still re-INVITEs issued for T.38 negotiation,
+ starting and stopping music-on-hold, and other reasons, and the 'canreinvite'
+ option never had any effect on these cases, it only affected the re-INVITEs
+ used for direct media path setup. For MGCP and Skinny, the option was poorly
+ named because those protocols don't even use INVITE messages at all. For
+ backwards compatibility, the old option is still supported in both normal
+ and Realtime configuration files, but all of the sample configuration files,
+ Realtime/LDAP schemas, and other documentation refer to it using the new name.
* The default console now will use colors according to the default background
color, instead of forcing the background color to black. If you are using a
Modified: branches/1.6.2/channels/chan_mgcp.c
URL: http://svn.asterisk.org/svn-view/asterisk/branches/1.6.2/channels/chan_mgcp.c?view=diff&rev=210191&r1=210190&r2=210191
==============================================================================
--- branches/1.6.2/channels/chan_mgcp.c (original)
+++ branches/1.6.2/channels/chan_mgcp.c Mon Aug 3 15:58:48 2009
@@ -80,7 +80,7 @@
#define MGCPDUMPER
#define DEFAULT_EXPIRY 120
#define MAX_EXPIRY 3600
-#define CANREINVITE 1
+#define DIRECTMEDIA 1
#ifndef INADDR_NONE
#define INADDR_NONE (in_addr_t)(-1)
@@ -177,7 +177,7 @@
static int singlepath = 0;
-static int canreinvite = CANREINVITE;
+static int directmedia = DIRECTMEDIA;
static char accountcode[AST_MAX_ACCOUNT_CODE] = "";
@@ -330,7 +330,7 @@
int threewaycalling;
int singlepath;
int cancallforward;
- int canreinvite;
+ int directmedia;
int callreturn;
int dnd; /* How does this affect callwait? Do we just deny a mgcp_request if we're dnd? */
int hascallerid;
@@ -3557,7 +3557,7 @@
int i=0, y=0;
int gw_reload = 0;
int ep_reload = 0;
- canreinvite = CANREINVITE;
+ directmedia = DIRECTMEDIA;
/* locate existing gateway */
gw = gateways;
@@ -3667,8 +3667,8 @@
cancallforward = ast_true(v->value);
} else if (!strcasecmp(v->name, "singlepath")) {
singlepath = ast_true(v->value);
- } else if (!strcasecmp(v->name, "canreinvite")) {
- canreinvite = ast_true(v->value);
+ } else if (!strcasecmp(v->name, "directmedia") || !strcasecmp(v->name, "canreinvite")) {
+ directmedia = ast_true(v->value);
} else if (!strcasecmp(v->name, "mailbox")) {
ast_copy_string(mailbox, v->value, sizeof(mailbox));
} else if (!strcasecmp(v->name, "hasvoicemail")) {
@@ -3753,7 +3753,7 @@
e->callreturn = callreturn;
e->cancallforward = cancallforward;
e->singlepath = singlepath;
- e->canreinvite = canreinvite;
+ e->directmedia = directmedia;
e->callwaiting = callwaiting;
e->hascallwaiting = callwaiting;
e->slowsequence = slowsequence;
@@ -3856,7 +3856,7 @@
e->pickupgroup=cur_pickupgroup;
e->callreturn = callreturn;
e->cancallforward = cancallforward;
- e->canreinvite = canreinvite;
+ e->directmedia = directmedia;
e->singlepath = singlepath;
e->callwaiting = callwaiting;
e->hascallwaiting = callwaiting;
@@ -3949,7 +3949,7 @@
*rtp = sub->rtp;
- if (sub->parent->canreinvite)
+ if (sub->parent->directmedia)
return AST_RTP_TRY_NATIVE;
else
return AST_RTP_TRY_PARTIAL;
Modified: branches/1.6.2/channels/chan_sip.c
URL: http://svn.asterisk.org/svn-view/asterisk/branches/1.6.2/channels/chan_sip.c?view=diff&rev=210191&r1=210190&r2=210191
==============================================================================
--- branches/1.6.2/channels/chan_sip.c (original)
+++ branches/1.6.2/channels/chan_sip.c Mon Aug 3 15:58:48 2009
@@ -1335,8 +1335,8 @@
/* re-INVITE related settings */
#define SIP_REINVITE (7 << 20) /*!< DP: four settings, uses three bits */
#define SIP_REINVITE_NONE (0 << 20) /*!< DP: no reinvite allowed */
-#define SIP_CAN_REINVITE (1 << 20) /*!< DP: allow peers to be reinvited to send media directly p2p */
-#define SIP_CAN_REINVITE_NAT (2 << 20) /*!< DP: allow media reinvite when new peer is behind NAT */
+#define SIP_DIRECT_MEDIA (1 << 20) /*!< DP: allow peers to be reinvited to send media directly p2p */
+#define SIP_DIRECT_MEDIA_NAT (2 << 20) /*!< DP: allow media reinvite when new peer is behind NAT */
#define SIP_REINVITE_UPDATE (4 << 20) /*!< DP: use UPDATE (RFC3311) when reinviting this peer */
/* "insecure" settings - see insecure2str() */
@@ -14796,7 +14796,7 @@
ast_cli(fd, " Nat : %s\n", nat2str(ast_test_flag(&peer->flags[0], SIP_NAT)));
ast_cli(fd, " ACL : %s\n", cli_yesno(peer->ha != NULL));
ast_cli(fd, " T38 pt UDPTL : %s\n", cli_yesno(ast_test_flag(&peer->flags[1], SIP_PAGE2_T38SUPPORT)));
- ast_cli(fd, " CanReinvite : %s\n", cli_yesno(ast_test_flag(&peer->flags[0], SIP_CAN_REINVITE)));
+ ast_cli(fd, " DirectMedia : %s\n", cli_yesno(ast_test_flag(&peer->flags[0], SIP_DIRECT_MEDIA)));
ast_cli(fd, " PromiscRedir : %s\n", cli_yesno(ast_test_flag(&peer->flags[0], SIP_PROMISCREDIR)));
ast_cli(fd, " User=Phone : %s\n", cli_yesno(ast_test_flag(&peer->flags[0], SIP_USEREQPHONE)));
ast_cli(fd, " Video Support: %s\n", cli_yesno(ast_test_flag(&peer->flags[1], SIP_PAGE2_VIDEOSUPPORT)));
@@ -14899,7 +14899,8 @@
astman_append(s, "SIP-AuthInsecure: %s\r\n", insecure2str(ast_test_flag(&peer->flags[0], SIP_INSECURE)));
astman_append(s, "SIP-NatSupport: %s\r\n", nat2str(ast_test_flag(&peer->flags[0], SIP_NAT)));
astman_append(s, "ACL: %s\r\n", (peer->ha?"Y":"N"));
- astman_append(s, "SIP-CanReinvite: %s\r\n", (ast_test_flag(&peer->flags[0], SIP_CAN_REINVITE)?"Y":"N"));
+ astman_append(s, "SIP-CanReinvite: %s\r\n", (ast_test_flag(&peer->flags[0], SIP_DIRECT_MEDIA)?"Y":"N"));
+ astman_append(s, "SIP-DirectMedia: %s\r\n", (ast_test_flag(&peer->flags[0], SIP_DIRECT_MEDIA)?"Y":"N"));
astman_append(s, "SIP-PromiscRedir: %s\r\n", (ast_test_flag(&peer->flags[0], SIP_PROMISCREDIR)?"Y":"N"));
astman_append(s, "SIP-UserPhone: %s\r\n", (ast_test_flag(&peer->flags[0], SIP_USEREQPHONE)?"Y":"N"));
astman_append(s, "SIP-VideoSupport: %s\r\n", (ast_test_flag(&peer->flags[1], SIP_PAGE2_VIDEOSUPPORT)?"Y":"N"));
@@ -22535,11 +22536,11 @@
ast_set_flag(&flags[0], SIP_NAT_ALWAYS);
else
ast_set_flag(&flags[0], SIP_NAT_RFC3581);
- } else if (!strcasecmp(v->name, "canreinvite")) {
+ } else if (!strcasecmp(v->name, "directmedia") || !strcasecmp(v->name, "canreinvite")) {
ast_set_flag(&mask[0], SIP_REINVITE);
ast_clear_flag(&flags[0], SIP_REINVITE);
if (ast_true(v->value)) {
- ast_set_flag(&flags[0], SIP_CAN_REINVITE | SIP_CAN_REINVITE_NAT);
+ ast_set_flag(&flags[0], SIP_DIRECT_MEDIA | SIP_DIRECT_MEDIA_NAT);
} else if (!ast_false(v->value)) {
char buf[64];
char *word, *next = buf;
@@ -22547,12 +22548,12 @@
ast_copy_string(buf, v->value, sizeof(buf));
while ((word = strsep(&next, ","))) {
if (!strcasecmp(word, "update")) {
- ast_set_flag(&flags[0], SIP_REINVITE_UPDATE | SIP_CAN_REINVITE);
+ ast_set_flag(&flags[0], SIP_REINVITE_UPDATE | SIP_DIRECT_MEDIA);
} else if (!strcasecmp(word, "nonat")) {
- ast_set_flag(&flags[0], SIP_CAN_REINVITE);
- ast_clear_flag(&flags[0], SIP_CAN_REINVITE_NAT);
+ ast_set_flag(&flags[0], SIP_DIRECT_MEDIA);
+ ast_clear_flag(&flags[0], SIP_DIRECT_MEDIA_NAT);
} else {
- ast_log(LOG_WARNING, "Unknown canreinvite mode '%s' on line %d\n", v->value, v->lineno);
+ ast_log(LOG_WARNING, "Unknown directmedia mode '%s' on line %d\n", v->value, v->lineno);
}
}
}
@@ -23625,7 +23626,7 @@
ast_copy_string(default_vmexten, DEFAULT_VMEXTEN, sizeof(default_vmexten));
ast_set_flag(&global_flags[0], SIP_DTMF_RFC2833); /*!< Default DTMF setting: RFC2833 */
ast_set_flag(&global_flags[0], SIP_NAT_RFC3581); /*!< NAT support if requested by device with rport */
- ast_set_flag(&global_flags[0], SIP_CAN_REINVITE); /*!< Allow re-invites */
+ ast_set_flag(&global_flags[0], SIP_DIRECT_MEDIA); /*!< Allow re-invites */
/* Debugging settings, always default to off */
dumphistory = FALSE;
@@ -24331,7 +24332,7 @@
return NULL;
sip_pvt_lock(p);
- if (p->udptl && ast_test_flag(&p->flags[0], SIP_CAN_REINVITE))
+ if (p->udptl && ast_test_flag(&p->flags[0], SIP_DIRECT_MEDIA))
udptl = p->udptl;
sip_pvt_unlock(p);
return udptl;
@@ -24381,9 +24382,9 @@
*rtp = p->rtp;
- if (ast_rtp_getnat(*rtp) && !ast_test_flag(&p->flags[0], SIP_CAN_REINVITE_NAT))
+ if (ast_rtp_getnat(*rtp) && !ast_test_flag(&p->flags[0], SIP_DIRECT_MEDIA_NAT))
res = AST_RTP_TRY_PARTIAL;
- else if (ast_test_flag(&p->flags[0], SIP_CAN_REINVITE))
+ else if (ast_test_flag(&p->flags[0], SIP_DIRECT_MEDIA))
res = AST_RTP_TRY_NATIVE;
else if (ast_test_flag(&global_jbconf, AST_JB_FORCED))
res = AST_RTP_GET_FAILED;
@@ -24410,7 +24411,7 @@
*rtp = p->vrtp;
- if (ast_test_flag(&p->flags[0], SIP_CAN_REINVITE))
+ if (ast_test_flag(&p->flags[0], SIP_DIRECT_MEDIA))
res = AST_RTP_TRY_NATIVE;
sip_pvt_unlock(p);
@@ -24435,7 +24436,7 @@
*rtp = p->trtp;
- if (ast_test_flag(&p->flags[0], SIP_CAN_REINVITE))
+ if (ast_test_flag(&p->flags[0], SIP_DIRECT_MEDIA))
res = AST_RTP_TRY_NATIVE;
sip_pvt_unlock(p);
@@ -24467,10 +24468,10 @@
/* if this peer cannot handle reinvites of the media stream to devices
that are known to be behind a NAT, then stop the process now
*/
- if (nat_active && !ast_test_flag(&p->flags[0], SIP_CAN_REINVITE_NAT)) {
- sip_pvt_unlock(p);
- return 0;
- }
+ if (nat_active && !ast_test_flag(&p->flags[0], SIP_DIRECT_MEDIA_NAT)) {
+ sip_pvt_unlock(p);
+ return 0;
+ }
if (rtp) {
changed |= ast_rtp_get_peer(rtp, &p->redirip);
Modified: branches/1.6.2/channels/chan_skinny.c
URL: http://svn.asterisk.org/svn-view/asterisk/branches/1.6.2/channels/chan_skinny.c?view=diff&rev=210191&r1=210190&r2=210191
==============================================================================
--- branches/1.6.2/channels/chan_skinny.c (original)
+++ branches/1.6.2/channels/chan_skinny.c Mon Aug 3 15:58:48 2009
@@ -1182,7 +1182,7 @@
int immediate; \
int hookstate; \
int nat; \
- int canreinvite; \
+ int directmedia; \
int prune;
struct skinny_line {
@@ -1208,7 +1208,7 @@
.hidecallerid = 0,
.amaflags = 0,
.instance = 0,
- .canreinvite = 0,
+ .directmedia = 0,
.nat = 0,
.confcapability = AST_FORMAT_ULAW | AST_FORMAT_ALAW,
.capability = 0,
@@ -2593,7 +2593,7 @@
l = sub->parent;
- if (!l->canreinvite || l->nat){
+ if (!l->directmedia || l->nat){
res = AST_RTP_TRY_PARTIAL;
if (skinnydebug)
ast_verb(1, "skinny_get_rtp_peer() Using AST_RTP_TRY_PARTIAL \n");
@@ -2653,7 +2653,7 @@
req->data.startmedia.conferenceId = htolel(sub->callid);
req->data.startmedia.passThruPartyId = htolel(sub->callid);
- if (!(l->canreinvite) || (l->nat)){
+ if (!(l->directmedia) || (l->nat)){
ast_rtp_get_us(rtp, &us);
req->data.startmedia.remoteIp = htolel(d->ourip.s_addr);
req->data.startmedia.remotePort = htolel(ntohs(us.sin_port));
@@ -6631,9 +6631,9 @@
CLINE_OPTS->callwaiting = ast_true(v->value);
continue;
}
- } else if (!strcasecmp(v->name, "canreinvite")) {
+ } else if (!strcasecmp(v->name, "directmedia") || !strcasecmp(v->name, "canreinvite")) {
if (type & (TYPE_DEF_LINE | TYPE_LINE)) {
- CLINE_OPTS->canreinvite = ast_true(v->value);
+ CLINE_OPTS->directmedia = ast_true(v->value);
continue;
}
} else if (!strcasecmp(v->name, "nat")) {
Modified: branches/1.6.2/configs/mgcp.conf.sample
URL: http://svn.asterisk.org/svn-view/asterisk/branches/1.6.2/configs/mgcp.conf.sample?view=diff&rev=210191&r1=210190&r2=210191
==============================================================================
--- branches/1.6.2/configs/mgcp.conf.sample (original)
+++ branches/1.6.2/configs/mgcp.conf.sample Mon Aug 3 15:58:48 2009
@@ -41,7 +41,7 @@
;[dlinkgw]
;host = 192.168.0.64
;context = default
-;canreinvite = no
+;directmedia = no
;line => aaln/2
;line => aaln/1
@@ -96,7 +96,7 @@
;callwaiting = no
;callreturn = yes
;cancallforward = yes
-;canreinvite = no
+;directmedia = no
;transfer = no
;dtmfmode = inband
;line => aaln/1 ; now lets save this config to line1 aka aaln/1
@@ -104,7 +104,7 @@
;callwaiting = no
;callreturn = yes
;cancallforward = yes
-;canreinvite = no
+;directmedia = no
;transfer = no
;dtmfmode = inband
;line => aaln/2 ; now lets save this config to line2 aka aaln/2
Modified: branches/1.6.2/configs/res_ldap.conf.sample
URL: http://svn.asterisk.org/svn-view/asterisk/branches/1.6.2/configs/res_ldap.conf.sample?view=diff&rev=210191&r1=210190&r2=210191
==============================================================================
--- branches/1.6.2/configs/res_ldap.conf.sample (original)
+++ branches/1.6.2/configs/res_ldap.conf.sample Mon Aug 3 15:58:48 2009
@@ -60,7 +60,7 @@
amaflags = AstAccountAMAFlags
callgroup = AstAccountCallGroup
callerid = AstAccountCallerID
-canreinvite = AstAccountCanReinvite
+directmedia = AstAccountDirectMedia
context = AstAccountContext
dtmfmode = AstAccountDTMFMode
fromuser = AstAccountFromUser
@@ -131,7 +131,7 @@
amaflags = AstAccountAMAFlags
callgroup = AstAccountCallGroup
callerid = AstAccountCallerID
-canreinvite = AstAccountCanReinvite
+directmedia = AstAccountDirectMedia
context = AstAccountContext
dtmfmode = AstAccountDTMFMode
fromuser = AstAccountFromUser
Modified: branches/1.6.2/configs/sip.conf.sample
URL: http://svn.asterisk.org/svn-view/asterisk/branches/1.6.2/configs/sip.conf.sample?view=diff&rev=210191&r1=210190&r2=210191
==============================================================================
--- branches/1.6.2/configs/sip.conf.sample (original)
+++ branches/1.6.2/configs/sip.conf.sample Mon Aug 3 15:58:48 2009
@@ -611,17 +611,17 @@
; ; (work around more UNIDEN bugs)
;----------------------------------- MEDIA HANDLING --------------------------------
-; By default, Asterisk tries to re-invite the audio to an optimal path. If there's
+; By default, Asterisk tries to re-invite media streams to an optimal path. If there's
; no reason for Asterisk to stay in the media path, the media will be redirected.
-; This does not really work with in the case where Asterisk is outside and have
-; clients on the inside of a NAT. In that case, you want to set canreinvite=nonat
-;
-;canreinvite=yes ; Asterisk by default tries to redirect the
- ; RTP media stream (audio) to go directly from
+; This does not really work well in the case where Asterisk is outside and the
+; clients are on the inside of a NAT. In that case, you want to set directmedia=nonat.
+;
+;directmedia=yes ; Asterisk by default tries to redirect the
+ ; RTP media stream to go directly from
; the caller to the callee. Some devices do not
; support this (especially if one of them is behind a NAT).
; The default setting is YES. If you have all clients
- ; behind a NAT, or for some other reason wants Asterisk to
+ ; behind a NAT, or for some other reason want Asterisk to
; stay in the audio path, you may want to turn this off.
; This setting also affect direct RTP
@@ -633,18 +633,18 @@
; the call directly with media peer-2-peer without re-invites.
; Will not work for video and cases where the callee sends
; RTP payloads and fmtp headers in the 200 OK that does not match the
- ; callers INVITE. This will also fail if canreinvite is enabled when
+ ; callers INVITE. This will also fail if directmedia is enabled when
; the device is actually behind NAT.
-;canreinvite=nonat ; An additional option is to allow media path redirection
+;directmedia=nonat ; An additional option is to allow media path redirection
; (reinvite) but only when the peer where the media is being
; sent is known to not be behind a NAT (as the RTP core can
; determine it based on the apparent IP address the media
; arrives from).
-;canreinvite=update ; Yet a third option... use UPDATE for media path redirection,
+;directmedia=update ; Yet a third option... use UPDATE for media path redirection,
; instead of INVITE. This can be combined with 'nonat', as
- ; 'canreinvite=update,nonat'. It implies 'yes'.
+ ; 'directmedia=update,nonat'. It implies 'yes'.
;ignoresdpversion=yes ; By default, Asterisk will honor the session version
; number in SDP packets and will only modify the SDP
@@ -808,7 +808,7 @@
; remotesecret
; transport
; dtmfmode
-; canreinvite
+; directmedia
; nat
; callgroup
; pickupgroup
@@ -918,12 +918,12 @@
[natted-phone](!,basic-options) ; another template inheriting basic-options
nat=yes
- canreinvite=no
+ directmedia=no
host=dynamic
[public-phone](!,basic-options) ; another template inheriting basic-options
nat=no
- canreinvite=yes
+ directmedia=yes
[my-codecs](!) ; a template for my preferred codecs
disallow=all
@@ -958,7 +958,7 @@
;host=192.168.0.23 ; we have a static but private IP address
; No registration allowed
;nat=no ; there is not NAT between phone and Asterisk
-;canreinvite=yes ; allow RTP voice traffic to bypass Asterisk
+;directmedia=yes ; allow RTP voice traffic to bypass Asterisk
;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone
;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time
; from the phone to asterisk (deprecated)
@@ -988,7 +988,7 @@
;callerid="Jane Smith" <5678>
;host=dynamic ; This device needs to register
;nat=yes ; X-Lite is behind a NAT router
-;canreinvite=no ; Typically set to NO if behind NAT
+;directmedia=no ; Typically set to NO if behind NAT
;disallow=all
;allow=gsm ; GSM consumes far less bandwidth than ulaw
;allow=ulaw
@@ -1061,7 +1061,7 @@
; Send SIP and RTP to the IP address that packet is
; received from instead of trusting SIP headers
;host=dynamic ; This device registers with us
-;canreinvite=no ; Asterisk by default tries to redirect the
+;directmedia=no ; Asterisk by default tries to redirect the
; RTP media stream (audio) to go directly from
; the caller to the callee. Some devices do not
; support this (especially if one of them is
Modified: branches/1.6.2/configs/skinny.conf.sample
URL: http://svn.asterisk.org/svn-view/asterisk/branches/1.6.2/configs/skinny.conf.sample?view=diff&rev=210191&r1=210190&r2=210191
==============================================================================
--- branches/1.6.2/configs/skinny.conf.sample (original)
+++ branches/1.6.2/configs/skinny.conf.sample Mon Aug 3 15:58:48 2009
@@ -157,7 +157,7 @@
;device=SEP00D0BA847E6B
;version=P002G204 ; Thanks critch
;context=did
-;canreinvite=yes ; Allow media to go directly between two RTP endpoints.
+;directmedia=yes ; Allow media to go directly between two RTP endpoints.
;line=120 ; Dial(Skinny/120 at florian)
; Typical config for a 7910
Modified: branches/1.6.2/contrib/scripts/realtime_pgsql.sql
URL: http://svn.asterisk.org/svn-view/asterisk/branches/1.6.2/contrib/scripts/realtime_pgsql.sql?view=diff&rev=210191&r1=210190&r2=210191
==============================================================================
--- branches/1.6.2/contrib/scripts/realtime_pgsql.sql (original)
+++ branches/1.6.2/contrib/scripts/realtime_pgsql.sql Mon Aug 3 15:58:48 2009
@@ -37,7 +37,7 @@
amaflags character varying(7),
callgroup character varying(10),
callerid character varying(80),
-canreinvite character varying(3) DEFAULT 'yes',
+directmedia character varying(3) DEFAULT 'yes',
context character varying(80),
defaultip character varying(15),
dtmfmode character varying(7),
Modified: branches/1.6.2/doc/chan_sip-perf-testing.txt
URL: http://svn.asterisk.org/svn-view/asterisk/branches/1.6.2/doc/chan_sip-perf-testing.txt?view=diff&rev=210191&r1=210190&r2=210191
==============================================================================
--- branches/1.6.2/doc/chan_sip-perf-testing.txt (original)
+++ branches/1.6.2/doc/chan_sip-perf-testing.txt Mon Aug 3 15:58:48 2009
@@ -58,7 +58,7 @@
context=test11
host=192.168.134.240 ;; the address of the host you will be running sipp on
user=sipp
-canreinvite=no
+directmedia=no
disallow=all
allow=ulaw
Modified: branches/1.6.2/doc/res_config_sqlite.txt
URL: http://svn.asterisk.org/svn-view/asterisk/branches/1.6.2/doc/res_config_sqlite.txt?view=diff&rev=210191&r1=210190&r2=210191
==============================================================================
--- branches/1.6.2/doc/res_config_sqlite.txt (original)
+++ branches/1.6.2/doc/res_config_sqlite.txt Mon Aug 3 15:58:48 2009
@@ -70,7 +70,7 @@
callgroup VARCHAR(10) DEFAULT NULL,
callerid VARCHAR(80) DEFAULT NULL,
cancallforward CHAR(3) DEFAULT 'yes',
- canreinvite CHAR(3) DEFAULT 'yes',
+ directmedia CHAR(3) DEFAULT 'yes',
context VARCHAR(80) DEFAULT NULL,
defaultip VARCHAR(15) DEFAULT NULL,
dtmfmode VARCHAR(7) DEFAULT NULL,
Modified: branches/1.6.2/doc/tex/phoneprov.tex
URL: http://svn.asterisk.org/svn-view/asterisk/branches/1.6.2/doc/tex/phoneprov.tex?view=diff&rev=210191&r1=210190&r2=210191
==============================================================================
--- branches/1.6.2/doc/tex/phoneprov.tex (original)
+++ branches/1.6.2/doc/tex/phoneprov.tex Mon Aug 3 15:58:48 2009
@@ -146,7 +146,7 @@
deletevoicemail = no
autoprov = yes
profile = polycom
-canreinvite = no
+directmedia = no
nat = no
fullname = User Two ; ${DISPLAY_NAME}
secret = test ; ${SECRET}
Modified: branches/1.6.2/main/rtp.c
URL: http://svn.asterisk.org/svn-view/asterisk/branches/1.6.2/main/rtp.c?view=diff&rev=210191&r1=210190&r2=210191
==============================================================================
--- branches/1.6.2/main/rtp.c (original)
+++ branches/1.6.2/main/rtp.c Mon Aug 3 15:58:48 2009
@@ -2124,7 +2124,7 @@
text_src_res = srcpr->get_trtp_info ? srcpr->get_trtp_info(c1, &tsrcp) : AST_RTP_GET_FAILED;
}
- /* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */
+ /* Check if bridge is still possible (In SIP directmedia=no stops this, like NAT) */
if (audio_dest_res != AST_RTP_TRY_NATIVE || (video_dest_res != AST_RTP_GET_FAILED && video_dest_res != AST_RTP_TRY_NATIVE)) {
/* Somebody doesn't want to play... */
ast_channel_unlock(c0);
@@ -2211,7 +2211,7 @@
else
destcodec = 0;
- /* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */
+ /* Check if bridge is still possible (In SIP directmedia=no stops this, like NAT) */
if (audio_dest_res != AST_RTP_TRY_NATIVE || (video_dest_res != AST_RTP_GET_FAILED && video_dest_res != AST_RTP_TRY_NATIVE) || audio_src_res != AST_RTP_TRY_NATIVE || (video_src_res != AST_RTP_GET_FAILED && video_src_res != AST_RTP_TRY_NATIVE) || !(srccodec & destcodec)) {
/* Somebody doesn't want to play... */
ast_channel_unlock(dest);
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