[asterisk-commits] murf: branch group/bug7433 r48292 - in
/team/group/bug7433: ./ channels/ conf...
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Tue Dec 5 23:30:46 MST 2006
Author: murf
Date: Wed Dec 6 00:30:46 2006
New Revision: 48292
URL: http://svn.digium.com/view/asterisk?view=rev&rev=48292
Log:
Merged revisions 48264,48268,48270,48279,48281 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r48264 | oej | 2006-12-05 05:39:30 -0700 (Tue, 05 Dec 2006) | 11 lines
Updating sip.conf.sample with information about T38 not working
when chan_local or chan_agent is involved in the call.
I don't know how big a fix that would be to solve, but this is
the current state of affairs.
(Chan_sip currently checks if the other side of the bridge
has a SIP tech. We could/should implement another check,
possibly for udptl_write or some flag in the ast_channel
structure).
........
r48268 | oej | 2006-12-05 08:59:05 -0700 (Tue, 05 Dec 2006) | 2 lines
Add missing s from another repository. (thanks jcmoore!)
........
r48270 | oej | 2006-12-05 10:29:43 -0700 (Tue, 05 Dec 2006) | 4 lines
Merging the invitestate-1.4 branch after successful testing.
Will check if I can solve this with less changes in 1.2.
........
r48279 | qwell | 2006-12-05 13:42:52 -0700 (Tue, 05 Dec 2006) | 4 lines
Fix curl version number testing to be much more friendly to non-bash shells.
Issue 8508, patch by me. This *SHOULD* be POSIX compliant now..
........
r48281 | file | 2006-12-05 13:45:28 -0700 (Tue, 05 Dec 2006) | 2 lines
Regenerate configure from Qwell's last commit.
........
Modified:
team/group/bug7433/ (props changed)
team/group/bug7433/channels/chan_sip.c
team/group/bug7433/configs/sip.conf.sample
team/group/bug7433/configure
team/group/bug7433/configure.ac
Propchange: team/group/bug7433/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Wed Dec 6 00:30:46 2006
@@ -1,1 +1,1 @@
-/branches/1.4:1-48255
+/branches/1.4:1-48291
Modified: team/group/bug7433/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/group/bug7433/channels/chan_sip.c?view=diff&rev=48292&r1=48291&r2=48292
==============================================================================
--- team/group/bug7433/channels/chan_sip.c (original)
+++ team/group/bug7433/channels/chan_sip.c Wed Dec 6 00:30:46 2006
@@ -244,6 +244,21 @@
AST_FAILURE = -1,
};
+/*! \brief States for the INVITE transaction, not the dialog
+ \note this is for the INVITE that sets up the dialog
+*/
+enum invitestates {
+ INV_NONE = 0, /*!< No state at all, maybe not an INVITE dialog */
+ INV_CALLING = 1, /*!< Invite sent, no answer */
+ INV_PROCEEDING = 2, /*!< We got/sent 1xx message */
+ INV_EARLY_MEDIA = 3, /*!< We got 18x message with to-tag back */
+ INV_COMPLETED = 4, /*!< Got final response with error. Wait for ACK, then CONFIRMED */
+ INV_CONFIRMED = 5, /*!< Confirmed response - we've got an ack (Incoming calls only) */
+ INV_TERMINATED = 6, /*!< Transaction done - either successful (AST_STATE_UP) or failed, but done
+ The only way out of this is a BYE from one side */
+ INV_CANCELLED = 7, /*!< Transaction cancelled by client or server in non-terminated state */
+};
+
/* Do _NOT_ make any changes to this enum, or the array following it;
if you think you are doing the right thing, you are probably
not doing the right thing. If you think there are changes
@@ -703,7 +718,7 @@
#define SIP_REALTIME (1 << 11) /*!< Flag for realtime users */
#define SIP_USECLIENTCODE (1 << 12) /*!< Trust X-ClientCode info message */
#define SIP_OUTGOING (1 << 13) /*!< Direction of the last transaction in this dialog */
-#define SIP_CAN_BYE (1 << 14) /*!< Can we send BYE on this dialog? */
+#define SIP_FREE_BIT (1 << 14) /*!< ---- */
#define SIP_DEFER_BYE_ON_TRANSFER (1 << 15) /*!< Do not hangup at first ast_hangup */
#define SIP_DTMF (3 << 16) /*!< DTMF Support: four settings, uses two bits */
#define SIP_DTMF_RFC2833 (0 << 16) /*!< DTMF Support: RTP DTMF - "rfc2833" */
@@ -877,6 +892,7 @@
static struct sip_pvt {
ast_mutex_t lock; /*!< Dialog private lock */
int method; /*!< SIP method that opened this dialog */
+ enum invitestates invitestate; /*!< The state of the INVITE transaction only */
AST_DECLARE_STRING_FIELDS(
AST_STRING_FIELD(callid); /*!< Global CallID */
AST_STRING_FIELD(randdata); /*!< Random data */
@@ -1593,6 +1609,13 @@
ast_verbose("%d headers, %d lines\n", p->initreq.headers, p->initreq.lines);
}
+static void sip_alreadygone(struct sip_pvt *dialog)
+{
+ if (option_debug > 2)
+ ast_log(LOG_DEBUG, "Setting SIP_ALREADYGONE on dialog %s\n", dialog->callid);
+ ast_set_flag(&dialog->flags[0], SIP_ALREADYGONE);
+}
+
/*! \brief returns true if 'name' (with optional trailing whitespace)
* matches the sip method 'id'.
@@ -1871,7 +1894,7 @@
ast_mutex_lock(&pkt->owner->lock);
}
if (pkt->owner->owner) {
- ast_set_flag(&pkt->owner->flags[0], SIP_ALREADYGONE);
+ sip_alreadygone(pkt->owner);
ast_log(LOG_WARNING, "Hanging up call %s - no reply to our critical packet.\n", pkt->owner->callid);
ast_queue_hangup(pkt->owner->owner);
ast_channel_unlock(pkt->owner->owner);
@@ -2802,6 +2825,7 @@
if (option_debug > 1)
ast_log(LOG_DEBUG,"Our T38 capability (%d), joint T38 capability (%d)\n", p->t38.capability, p->t38.jointcapability);
transmit_invite(p, SIP_INVITE, 1, 2);
+ p->invitestate = INV_CALLING;
/* Initialize auto-congest time */
p->initid = ast_sched_add(sched, p->maxtime ? (p->maxtime * 4) : SIP_TRANS_TIMEOUT, auto_congest, p);
@@ -3271,7 +3295,7 @@
return 0;
}
/* If the call is not UP, we need to send CANCEL instead of BYE */
- if (ast->_state == AST_STATE_RING || ast->_state == AST_STATE_RINGING) {
+ if (ast->_state == AST_STATE_RING || ast->_state == AST_STATE_RINGING || p->invitestate < INV_COMPLETED) {
needcancel = TRUE;
if (option_debug > 3)
ast_log(LOG_DEBUG, "Hanging up channel in state %s (not UP)\n", ast_state2str(ast->_state));
@@ -3295,7 +3319,7 @@
*/
if (ast_test_flag(&p->flags[0], SIP_ALREADYGONE))
needdestroy = 1; /* Set destroy flag at end of this function */
- else
+ else if (p->invitestate != INV_CALLING)
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
/* Start the process if it's not already started */
@@ -3306,7 +3330,8 @@
__sip_pretend_ack(p);
/* if we can't send right now, mark it pending */
- if (!ast_test_flag(&p->flags[0], SIP_CAN_BYE)) {
+ if (p->invitestate == INV_CALLING) {
+ /* We can't send anything in CALLING state */
ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
/* Do we need a timer here if we don't hear from them at all? */
} else {
@@ -3356,6 +3381,7 @@
but we can't send one while we have "INVITE" outstanding. */
ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
ast_clear_flag(&p->flags[0], SIP_NEEDREINVITE);
+ sip_cancel_destroy(p);
}
}
}
@@ -3610,6 +3636,7 @@
switch(condition) {
case AST_CONTROL_RINGING:
if (ast->_state == AST_STATE_RING) {
+ p->invitestate = INV_EARLY_MEDIA;
if (!ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) ||
(ast_test_flag(&p->flags[0], SIP_PROG_INBAND) == SIP_PROG_INBAND_NEVER)) {
/* Send 180 ringing if out-of-band seems reasonable */
@@ -3626,7 +3653,8 @@
case AST_CONTROL_BUSY:
if (ast->_state != AST_STATE_UP) {
transmit_response(p, "486 Busy Here", &p->initreq);
- ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
+ p->invitestate = INV_TERMINATED;
+ sip_alreadygone(p);
ast_softhangup_nolock(ast, AST_SOFTHANGUP_DEV);
break;
}
@@ -3635,7 +3663,8 @@
case AST_CONTROL_CONGESTION:
if (ast->_state != AST_STATE_UP) {
transmit_response(p, "503 Service Unavailable", &p->initreq);
- ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
+ p->invitestate = INV_TERMINATED;
+ sip_alreadygone(p);
ast_softhangup_nolock(ast, AST_SOFTHANGUP_DEV);
break;
}
@@ -3646,6 +3675,7 @@
!ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) &&
!ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
transmit_response(p, "100 Trying", &p->initreq);
+ p->invitestate = INV_PROCEEDING;
break;
}
res = -1;
@@ -3654,6 +3684,7 @@
if ((ast->_state != AST_STATE_UP) &&
!ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) &&
!ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
+ p->invitestate = INV_EARLY_MEDIA;
transmit_response_with_sdp(p, "183 Session Progress", &p->initreq, XMIT_UNRELIABLE);
ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
break;
@@ -7376,6 +7407,9 @@
{
struct sip_request resp;
+ if (sipmethod == SIP_ACK)
+ p->invitestate = INV_CONFIRMED;
+
reqprep(&resp, p, sipmethod, seqno, newbranch);
add_header_contentLength(&resp, 0);
return send_request(p, &resp, reliable, seqno ? seqno : p->ocseq);
@@ -11464,7 +11498,7 @@
{
if (ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
/* if we can't BYE, then this is really a pending CANCEL */
- if (!ast_test_flag(&p->flags[0], SIP_CAN_BYE))
+ if (p->invitestate == INV_PROCEEDING || p->invitestate == INV_EARLY_MEDIA)
transmit_request(p, SIP_CANCEL, p->ocseq, XMIT_RELIABLE, FALSE);
/* Actually don't destroy us yet, wait for the 487 on our original
INVITE, but do set an autodestruct just in case we never get it. */
@@ -11515,6 +11549,15 @@
if (resp > 100 && resp < 200 && resp!=101 && resp != 180 && resp != 183)
resp = 183;
+ /* Any response between 100 and 199 is PROCEEDING */
+ if (resp >= 100 && resp < 200 && p->invitestate == INV_CALLING)
+ p->invitestate = INV_PROCEEDING;
+
+ /* Final response, not 200 ? */
+ if (resp >= 300 && (p->invitestate == INV_CALLING || p->invitestate == INV_PROCEEDING || p->invitestate == INV_EARLY_MEDIA ))
+ p->invitestate = INV_COMPLETED;
+
+
switch (resp) {
case 100: /* Trying */
case 101: /* Dialog establishment */
@@ -11533,13 +11576,13 @@
}
}
if (find_sdp(req)) {
+ p->invitestate = INV_EARLY_MEDIA;
res = process_sdp(p, req);
if (!ast_test_flag(req, SIP_PKT_IGNORE) && p->owner) {
/* Queue a progress frame only if we have SDP in 180 */
ast_queue_control(p->owner, AST_CONTROL_PROGRESS);
}
}
- ast_set_flag(&p->flags[0], SIP_CAN_BYE);
check_pendings(p);
break;
@@ -11548,13 +11591,13 @@
sip_cancel_destroy(p);
/* Ignore 183 Session progress without SDP */
if (find_sdp(req)) {
+ p->invitestate = INV_EARLY_MEDIA;
res = process_sdp(p, req);
if (!ast_test_flag(req, SIP_PKT_IGNORE) && p->owner) {
/* Queue a progress frame */
ast_queue_control(p->owner, AST_CONTROL_PROGRESS);
}
}
- ast_set_flag(&p->flags[0], SIP_CAN_BYE);
check_pendings(p);
break;
@@ -11655,8 +11698,8 @@
ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
}
/* If I understand this right, the branch is different for a non-200 ACK only */
+ p->invitestate = INV_TERMINATED;
transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, TRUE);
- ast_set_flag(&p->flags[0], SIP_CAN_BYE);
check_pendings(p);
break;
case 407: /* Proxy authentication */
@@ -11674,7 +11717,7 @@
if ((p->authtries == MAX_AUTHTRIES) || do_proxy_auth(p, req, authenticate, authorization, SIP_INVITE, 1)) {
ast_log(LOG_NOTICE, "Failed to authenticate on INVITE to '%s'\n", get_header(&p->initreq, "From"));
ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
- ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
+ sip_alreadygone(p);
if (p->owner)
ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
}
@@ -11688,14 +11731,14 @@
if (!ast_test_flag(req, SIP_PKT_IGNORE) && p->owner)
ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
- ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
+ sip_alreadygone(p);
break;
case 404: /* Not found */
transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
if (p->owner && !ast_test_flag(req, SIP_PKT_IGNORE))
ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
- ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
+ sip_alreadygone(p);
break;
case 481: /* Call leg does not exist */
@@ -12152,7 +12195,6 @@
/* Fatal response */
if ((option_verbose > 2) && (resp != 487))
ast_verbose(VERBOSE_PREFIX_3 "Got SIP response %d \"%s\" back from %s\n", resp, rest, ast_inet_ntoa(p->sa.sin_addr));
- ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
stop_media_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */
@@ -12211,7 +12253,7 @@
/* ACK on invite */
if (sipmethod == SIP_INVITE)
transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
- ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
+ sip_alreadygone(p);
if (!p->owner)
ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
} else if ((resp >= 100) && (resp < 200)) {
@@ -13258,6 +13300,7 @@
if (option_debug > 1)
ast_log(LOG_DEBUG, "%s: New call is still down.... Trying... \n", c->name);
transmit_response(p, "100 Trying", req);
+ p->invitestate = INV_PROCEEDING;
ast_setstate(c, AST_STATE_RING);
if (strcmp(p->exten, ast_pickup_ext())) { /* Call to extension -start pbx on this call */
enum ast_pbx_result res;
@@ -13267,6 +13310,7 @@
switch(res) {
case AST_PBX_FAILED:
ast_log(LOG_WARNING, "Failed to start PBX :(\n");
+ p->invitestate = INV_COMPLETED;
if (ast_test_flag(req, SIP_PKT_IGNORE))
transmit_response(p, "503 Unavailable", req);
else
@@ -13274,6 +13318,7 @@
break;
case AST_PBX_CALL_LIMIT:
ast_log(LOG_WARNING, "Failed to start PBX (call limit reached) \n");
+ p->invitestate = INV_COMPLETED;
if (ast_test_flag(req, SIP_PKT_IGNORE))
transmit_response(p, "480 Temporarily Unavailable", req);
else
@@ -13301,7 +13346,7 @@
transmit_response(p, "503 Unavailable", req); /* OEJ - Right answer? */
else
transmit_response_reliable(p, "503 Unavailable", req);
- ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
+ sip_alreadygone(p);
/* Unlock locks so ast_hangup can do its magic */
ast_mutex_unlock(&p->lock);
c->hangupcause = AST_CAUSE_CALL_REJECTED;
@@ -13310,6 +13355,7 @@
ast_setstate(c, AST_STATE_DOWN);
c->hangupcause = AST_CAUSE_NORMAL_CLEARING;
}
+ p->invitestate = INV_COMPLETED;
ast_hangup(c);
ast_mutex_lock(&p->lock);
c = NULL;
@@ -13317,9 +13363,11 @@
break;
case AST_STATE_RING:
transmit_response(p, "100 Trying", req);
+ p->invitestate = INV_PROCEEDING;
break;
case AST_STATE_RINGING:
transmit_response(p, "180 Ringing", req);
+ p->invitestate = INV_PROCEEDING;
break;
case AST_STATE_UP:
if (option_debug > 1)
@@ -13405,6 +13453,7 @@
transmit_response_with_sdp(p, "200 OK", req, XMIT_CRITICAL);
}
+ p->invitestate = INV_TERMINATED;
break;
default:
ast_log(LOG_WARNING, "Don't know how to handle INVITE in state %d\n", c->_state);
@@ -13425,6 +13474,7 @@
transmit_response(p, msg, req);
else
transmit_response_reliable(p, msg, req);
+ p->invitestate = INV_COMPLETED;
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
}
}
@@ -13620,7 +13670,7 @@
transmit_response(p, "603 Declined (No dialog)", req);
if (!ast_test_flag(req, SIP_PKT_IGNORE)) {
append_history(p, "Xfer", "Refer failed. Outside of dialog.");
- ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
+ sip_alreadygone(p);
ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
}
return 0;
@@ -13879,7 +13929,8 @@
{
check_via(p, req);
- ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
+ sip_alreadygone(p);
+ p->invitestate = INV_CANCELLED;
if (p->owner && p->owner->_state == AST_STATE_UP) {
/* This call is up, cancel is ignored, we need a bye */
@@ -13912,12 +13963,14 @@
struct ast_channel *bridged_to;
/* If we have an INCOMING invite that we haven't answered, terminate that transaction */
- if (p->pendinginvite && !ast_test_flag(&p->flags[0], SIP_OUTGOING) && !ast_test_flag(req, SIP_PKT_IGNORE) && !p->owner)
+ if (p->pendinginvite && !ast_test_flag(&p->flags[0], SIP_OUTGOING) && !ast_test_flag(req, SIP_PKT_IGNORE) && !p->owner)
transmit_response_reliable(p, "487 Request Terminated", &p->initreq);
+
+ p->invitestate = INV_TERMINATED;
copy_request(&p->initreq, req);
check_via(p, req);
- ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
+ sip_alreadygone(p);
/* Get RTCP quality before end of call */
if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY) || p->owner) {
@@ -14483,6 +14536,7 @@
case SIP_ACK:
/* Make sure we don't ignore this */
if (seqno == p->pendinginvite) {
+ p->invitestate = INV_CONFIRMED;
p->pendinginvite = 0;
__sip_ack(p, seqno, FLAG_RESPONSE, 0);
if (find_sdp(req)) {
Modified: team/group/bug7433/configs/sip.conf.sample
URL: http://svn.digium.com/view/asterisk/team/group/bug7433/configs/sip.conf.sample?view=diff&rev=48292&r1=48291&r2=48292
==============================================================================
--- team/group/bug7433/configs/sip.conf.sample (original)
+++ team/group/bug7433/configs/sip.conf.sample Wed Dec 6 00:30:46 2006
@@ -200,7 +200,7 @@
;notifyhold = yes ; Notify subscriptions on HOLD state (default: no)
; Turning on notifyringing and notifyhold will add a lot
; more database transactions if you are using realtime.
-;limitonpeer = yes ; Apply call limits on peers only. This will improve
+;limitonpeers = yes ; Apply call limits on peers only. This will improve
; status notification when you are using type=friend
; Inbound calls, that really apply to the user part
; of a friend will now be added to and compared with
@@ -214,6 +214,8 @@
; both parties have T38 support enabled in their Asterisk configuration
; This has to be enabled in the general section for all devices to work. You can then
; disable it on a per device basis.
+;
+; T.38 faxing only works in SIP to SIP calls, with no local or agent channel being used.
;
; t38pt_udptl = yes ; Default false
;
Modified: team/group/bug7433/configure
URL: http://svn.digium.com/view/asterisk/team/group/bug7433/configure?view=diff&rev=48292&r1=48291&r2=48292
==============================================================================
--- team/group/bug7433/configure (original)
+++ team/group/bug7433/configure Wed Dec 6 00:30:46 2006
@@ -1,5 +1,5 @@
#! /bin/sh
-# From configure.ac Revision: 47758 .
+# From configure.ac Revision: 48279 .
# Guess values for system-dependent variables and create Makefiles.
# Generated by GNU Autoconf 2.60a.
#
@@ -31046,30 +31046,17 @@
if test ! x"${CURL}" = xNo; then
# check for version
- if test "${host_os}" = "SunOS"; then
- if [ 0x`curl-config --vernum` -ge 0x70907 ]; then
- CURL_INCLUDE=$(${CURL} --cflags)
- CURL_LIB=$(${CURL} --libs)
- PBX_CURL=1
+ if test $(printf "%d" 0x$(curl-config --vernum)) -ge $(printf "%d" 0x070907); then
+ CURL_INCLUDE=$(${CURL} --cflags)
+ CURL_LIB=$(${CURL} --libs)
+ PBX_CURL=1
cat >>confdefs.h <<\_ACEOF
#define HAVE_CURL 1
_ACEOF
- fi
- else
- if [[ 0x`curl-config --vernum` -ge 0x70907 ]]; then
- CURL_INCLUDE=$(${CURL} --cflags)
- CURL_LIB=$(${CURL} --libs)
- PBX_CURL=1
-
-cat >>confdefs.h <<\_ACEOF
-#define HAVE_CURL 1
-_ACEOF
-
- fi
- fi
- fi
+ fi
+ fi
fi
ac_config_files="$ac_config_files build_tools/menuselect-deps makeopts channels/h323/Makefile"
Modified: team/group/bug7433/configure.ac
URL: http://svn.digium.com/view/asterisk/team/group/bug7433/configure.ac?view=diff&rev=48292&r1=48291&r2=48292
==============================================================================
--- team/group/bug7433/configure.ac (original)
+++ team/group/bug7433/configure.ac Wed Dec 6 00:30:46 2006
@@ -1033,22 +1033,13 @@
AC_PATH_TOOL([CURL], [curl-config], No)
if test ! x"${CURL}" = xNo; then
# check for version
- if test "${host_os}" = "SunOS"; then
- if [[ 0x`curl-config --vernum` -ge 0x70907 ]]; then
- CURL_INCLUDE=$(${CURL} --cflags)
- CURL_LIB=$(${CURL} --libs)
- PBX_CURL=1
- AC_DEFINE([HAVE_CURL], 1, [Define if your system has the curl libraries.])
- fi
- else
- if [[[ 0x`curl-config --vernum` -ge 0x70907 ]]]; then
- CURL_INCLUDE=$(${CURL} --cflags)
- CURL_LIB=$(${CURL} --libs)
- PBX_CURL=1
- AC_DEFINE([HAVE_CURL], 1, [Define if your system has the curl libraries.])
- fi
- fi
- fi
+ if test $(printf "%d" 0x$(curl-config --vernum)) -ge $(printf "%d" 0x070907); then
+ CURL_INCLUDE=$(${CURL} --cflags)
+ CURL_LIB=$(${CURL} --libs)
+ PBX_CURL=1
+ AC_DEFINE([HAVE_CURL], 1, [Define if your system has the curl libraries.])
+ fi
+ fi
fi
AC_CONFIG_FILES([build_tools/menuselect-deps makeopts channels/h323/Makefile])
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