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Author SHA1 Message Date
Asterisk Autobuilder
67d4994dec Importing release summary for 10.1.0-rc2 release.
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/10.1.0-rc2@352347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-24 18:51:49 +00:00
Asterisk Autobuilder
fe5270a92f Updated with test results
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/10.1.0-rc2@352346 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-24 18:51:00 +00:00
Matthew Jordan
907ae33024 Merged 349732, 350553, 352228, 352015, 351505, 351289, 351308
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/10.1.0-rc2@352290 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-24 16:38:46 +00:00
Matthew Jordan
a2f6f5de34 Create 10.1.0-rc2
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/10.1.0-rc2@352285 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-24 14:45:42 +00:00
Asterisk Autobuilder
cb517c90de Updated release date
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/10.1.0-rc1@349402 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-30 15:25:10 +00:00
Asterisk Autobuilder
9137f7408b Updated ChangeLog with test results
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/10.1.0-rc1@349401 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-30 14:58:22 +00:00
Asterisk Autobuilder
91e047df57 Use autotagged externals
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/10.1.0-rc1@349399 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-29 19:41:47 +00:00
Asterisk Autobuilder
83665af6bc Importing release summary for 10.1.0-rc1 release.
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/10.1.0-rc1@349398 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-29 19:41:37 +00:00
Asterisk Autobuilder
9b7c240428 Importing files for 10.1.0-rc1 release.
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/10.1.0-rc1@349397 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-29 19:41:31 +00:00
Asterisk Autobuilder
3b1f3d20dc Creating tag for the release of asterisk-10.1.0-rc1
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/10.1.0-rc1@349396 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-29 19:38:09 +00:00
12 changed files with 20661 additions and 41 deletions

3
.lastclean Normal file
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@@ -0,0 +1,3 @@
39

1
.version Normal file
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@@ -0,0 +1 @@
10.1.0-rc2

15
CHANGES
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@@ -8,6 +8,21 @@
===
==============================================================================
------------------------------------------------------------------------------
--- Functionality changes since Asterisk 10.0.0 ------------------------------
------------------------------------------------------------------------------
RTP changes
-------------
* A new option, 'probation' has been added to rtp.conf
RTP in strictrtp mode can now require more than 1 packet to exit learning
mode with a new source (and by default requires 4). The probation option
allows the user to change the required number of packets in sequence to any
desired value. Use a value of 1 to essentially restore the old behavior.
Also, with strictrtp on, Asterisk will now drop all packets until learning
mode has successfully exited. These changes are based on how pjmedia handles
media sources and source changes.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 1.8 to Asterisk 10 -------------------
------------------------------------------------------------------------------

20342
ChangeLog Normal file

File diff suppressed because it is too large Load Diff

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@@ -2397,9 +2397,8 @@ static struct call_queue *find_load_queue_rt_friendly(const char *queuename)
if (queue_vars) {
member_config = ast_load_realtime_multientry("queue_members", "interface LIKE", "%", "queue_name", queuename, SENTINEL);
if (!member_config) {
ast_log(LOG_ERROR, "no queue_members defined in your config (extconfig.conf).\n");
ast_variables_destroy(queue_vars);
return NULL;
ast_debug(1, "No queue_members defined in config extconfig.conf\n");
member_config = ast_config_new();
}
}
if (q) {

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@@ -0,0 +1,68 @@
<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN" http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd">
<html xmlns="http://www.w3.org/1999/xhtml">
<head><meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1" /><title>Release Summary - asterisk-10.1.0-rc2</title></head>
<body>
<h1 align="center"><a name="top">Release Summary</a></h1>
<h3 align="center">asterisk-10.1.0-rc2</h3>
<h3 align="center">Date: 2012-01-24</h3>
<h3 align="center">&lt;asteriskteam@digium.com&gt;</h3>
<hr/>
<h2 align="center">Table of Contents</h2>
<ol>
<li><a href="#summary">Summary</a></li>
<li><a href="#contributors">Contributors</a></li>
<li><a href="#commits">Other Changes</a></li>
<li><a href="#diffstat">Diffstat</a></li>
</ol>
<hr/>
<a name="summary"><h2 align="center">Summary</h2></a>
<center><a href="#top">[Back to Top]</a></center><br/><p>This release includes only bug fixes. The changes included were made only to address problems that have been identified in this release series. Users should be able to safely upgrade to this version if this release series is already in use. Users considering upgrading from a previous release series are strongly encouraged to review the UPGRADE.txt document as well as the CHANGES document for information about upgrading to this release series.</p>
<p>The data in this summary reflects changes that have been made since the previous release, asterisk-10.1.0-rc1.</p>
<hr/>
<a name="contributors"><h2 align="center">Contributors</h2></a>
<center><a href="#top">[Back to Top]</a></center><br/><p>This table lists the people who have submitted code, those that have tested patches, as well as those that reported issues on the issue tracker that were resolved in this release. For coders, the number is how many of their patches (of any size) were committed into this release. For testers, the number is the number of times their name was listed as assisting with testing a patch. Finally, for reporters, the number is the number of issues that they reported that were closed by commits that went into this release.</p>
<table width="100%" border="0">
<tr>
<td width="33%"><h3>Coders</h3></td>
<td width="33%"><h3>Testers</h3></td>
<td width="33%"><h3>Reporters</h3></td>
</tr>
<tr valign="top">
<td>
2 mjordan<br/>
1 bebuild<br/>
</td>
<td>
</td>
<td>
</td>
</tr>
</table>
<hr/>
<a name="commits"><h2 align="center">Commits Not Associated with an Issue</h2></a>
<center><a href="#top">[Back to Top]</a></center><br/><p>This is a list of all changes that went into this release that did not directly close an issue from the issue tracker. The commits may have been marked as being related to an issue. If that is the case, the issue numbers are listed here, as well.</p>
<table width="100%" border="1">
<tr><td><b>Revision</b></td><td><b>Author</b></td><td><b>Summary</b></td><td><b>Issues Referenced</b></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=352285">352285</a></td><td>mjordan</td><td>Create 10.1.0-rc2</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=352290">352290</a></td><td>mjordan</td><td>Merged 349732, 350553, 352228, 352015, 351505, 351289, 351308</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=352346">352346</a></td><td>bebuild</td><td>Updated with test results</td>
<td></td></tr></table>
<hr/>
<a name="diffstat"><h2 align="center">Diffstat Results</h2></a>
<center><a href="#top">[Back to Top]</a></center><br/><p>This is a summary of the changes to the source code that went into this release that was generated using the diffstat utility.</p>
<pre>
.version | 2
CHANGES | 15 +
ChangeLog | 48 +++
apps/app_queue.c | 5
asterisk-10.1.0-rc1-summary.html | 275 -------------------
asterisk-10.1.0-rc1-summary.txt | 553 ---------------------------------------
channels/chan_sip.c | 58 +---
configs/rtp.conf.sample | 7
main/features.c | 11
main/file.c | 16 -
res/res_rtp_asterisk.c | 77 +++++
11 files changed, 197 insertions(+), 870 deletions(-)
</pre><br/>
<hr/>
</body>
</html>

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@@ -0,0 +1,99 @@
Release Summary
asterisk-10.1.0-rc2
Date: 2012-01-24
<asteriskteam@digium.com>
----------------------------------------------------------------------
Table of Contents
1. Summary
2. Contributors
3. Other Changes
4. Diffstat
----------------------------------------------------------------------
Summary
[Back to Top]
This release includes only bug fixes. The changes included were made only
to address problems that have been identified in this release series.
Users should be able to safely upgrade to this version if this release
series is already in use. Users considering upgrading from a previous
release series are strongly encouraged to review the UPGRADE.txt document
as well as the CHANGES document for information about upgrading to this
release series.
The data in this summary reflects changes that have been made since the
previous release, asterisk-10.1.0-rc1.
----------------------------------------------------------------------
Contributors
[Back to Top]
This table lists the people who have submitted code, those that have
tested patches, as well as those that reported issues on the issue tracker
that were resolved in this release. For coders, the number is how many of
their patches (of any size) were committed into this release. For testers,
the number is the number of times their name was listed as assisting with
testing a patch. Finally, for reporters, the number is the number of
issues that they reported that were closed by commits that went into this
release.
Coders Testers Reporters
2 mjordan
1 bebuild
----------------------------------------------------------------------
Commits Not Associated with an Issue
[Back to Top]
This is a list of all changes that went into this release that did not
directly close an issue from the issue tracker. The commits may have been
marked as being related to an issue. If that is the case, the issue
numbers are listed here, as well.
+------------------------------------------------------------------------+
| Revision | Author | Summary | Issues Referenced |
|----------+---------+-------------------------------+-------------------|
| 352285 | mjordan | Create 10.1.0-rc2 | |
|----------+---------+-------------------------------+-------------------|
| | | Merged 349732, 350553, | |
| 352290 | mjordan | 352228, 352015, 351505, | |
| | | 351289, 351308 | |
|----------+---------+-------------------------------+-------------------|
| 352346 | bebuild | Updated with test results | |
+------------------------------------------------------------------------+
----------------------------------------------------------------------
Diffstat Results
[Back to Top]
This is a summary of the changes to the source code that went into this
release that was generated using the diffstat utility.
.version | 2
CHANGES | 15 +
ChangeLog | 48 +++
apps/app_queue.c | 5
asterisk-10.1.0-rc1-summary.html | 275 -------------------
asterisk-10.1.0-rc1-summary.txt | 553 ---------------------------------------
channels/chan_sip.c | 58 +---
configs/rtp.conf.sample | 7
main/features.c | 11
main/file.c | 16 -
res/res_rtp_asterisk.c | 77 +++++
11 files changed, 197 insertions(+), 870 deletions(-)
----------------------------------------------------------------------

View File

@@ -3882,6 +3882,7 @@ static int __sip_autodestruct(const void *data)
ast_channel_unref(owner);
} else if (p->refer && !p->alreadygone) {
ast_debug(3, "Finally hanging up channel after transfer: %s\n", p->callid);
stop_media_flows(p);
transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, 1);
append_history(p, "ReferBYE", "Sending BYE on transferer call leg %s", p->callid);
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
@@ -20714,15 +20715,22 @@ static void handle_response_notify(struct sip_pvt *p, int resp, const char *rest
case 200: /* Notify accepted */
/* They got the notify, this is the end */
if (p->owner) {
if (!p->refer) {
ast_log(LOG_WARNING, "Notify answer on an owned channel? - %s\n", p->owner->name);
ast_queue_hangup_with_cause(p->owner, AST_CAUSE_NORMAL_UNSPECIFIED);
if (p->refer) {
ast_log(LOG_NOTICE, "Got OK on REFER Notify message\n");
} else {
ast_debug(4, "Got OK on REFER Notify message\n");
ast_log(LOG_WARNING, "Notify answer on an owned channel? - %s\n", p->owner->name);
/*
* XXX There is discrepancy on whether a hangup should be queued
* or not. This code used to be duplicated in two places, and the more
* frequently hit area had this disabled, making it the de facto
* "correct" way to go.
*
* ast_queue_hangup_with_cause(p->owner, AST_CAUSE_NORMAL_UNSPECIFIED);
*/
}
} else {
if (p->subscribed == NONE) {
ast_debug(4, "Got 200 accepted on NOTIFY\n");
if (p->subscribed == NONE && !p->refer) {
ast_debug(4, "Got 200 accepted on NOTIFY %s\n", p->callid);
pvt_set_needdestroy(p, "received 200 response");
}
if (ast_test_flag(&p->flags[1], SIP_PAGE2_STATECHANGEQUEUE)) {
@@ -20747,6 +20755,9 @@ static void handle_response_notify(struct sip_pvt *p, int resp, const char *rest
pvt_set_needdestroy(p, "failed to authenticate NOTIFY");
}
break;
case 481: /* Call leg does not exist */
pvt_set_needdestroy(p, "Received 481 response for NOTIFY");
break;
}
}
@@ -21389,6 +21400,9 @@ static void handle_response(struct sip_pvt *p, int resp, const char *rest, struc
} else if (sipmethod == SIP_MESSAGE) {
/* More good gravy! */
handle_response_message(p, resp, rest, req, seqno);
} else if (sipmethod == SIP_NOTIFY) {
/* The gravy train continues to roll */
handle_response_notify(p, resp, rest, req, seqno);
} else if (ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
switch(resp) {
case 100: /* 100 Trying */
@@ -21404,8 +21418,6 @@ static void handle_response(struct sip_pvt *p, int resp, const char *rest, struc
p->authtries = 0; /* Reset authentication counter */
if (sipmethod == SIP_INVITE) {
handle_response_invite(p, resp, rest, req, seqno);
} else if (sipmethod == SIP_NOTIFY) {
handle_response_notify(p, resp, rest, req, seqno);
} else if (sipmethod == SIP_REGISTER) {
handle_response_register(p, resp, rest, req, seqno);
} else if (sipmethod == SIP_SUBSCRIBE) {
@@ -21420,8 +21432,6 @@ static void handle_response(struct sip_pvt *p, int resp, const char *rest, struc
case 407: /* Proxy auth required */
if (sipmethod == SIP_INVITE)
handle_response_invite(p, resp, rest, req, seqno);
else if (sipmethod == SIP_NOTIFY)
handle_response_notify(p, resp, rest, req, seqno);
else if (sipmethod == SIP_SUBSCRIBE)
handle_response_subscribe(p, resp, rest, req, seqno);
else if (p->registry && sipmethod == SIP_REGISTER)
@@ -21496,8 +21506,6 @@ static void handle_response(struct sip_pvt *p, int resp, const char *rest, struc
handle_response_invite(p, resp, rest, req, seqno);
} else if (sipmethod == SIP_SUBSCRIBE) {
handle_response_subscribe(p, resp, rest, req, seqno);
} else if (sipmethod == SIP_NOTIFY) {
pvt_set_needdestroy(p, "received 481 response");
} else if (sipmethod == SIP_BYE) {
/* The other side has no transaction to bye,
just assume it's all right then */
@@ -21658,24 +21666,6 @@ static void handle_response(struct sip_pvt *p, int resp, const char *rest, struc
ast_debug(1, "Got 200 OK on CANCEL\n");
/* Wait for 487, then destroy */
} else if (sipmethod == SIP_NOTIFY) {
/* They got the notify, this is the end */
if (p->owner) {
if (p->refer) {
ast_debug(1, "Got 200 OK on NOTIFY for transfer\n");
} else
ast_log(LOG_WARNING, "Notify answer on an owned channel?\n");
/* ast_queue_hangup(p->owner); Disabled */
} else {
if (!p->subscribed && !p->refer) {
pvt_set_needdestroy(p, "transaction completed");
}
if (ast_test_flag(&p->flags[1], SIP_PAGE2_STATECHANGEQUEUE)) {
/* Ready to send the next state we have on queue */
ast_clear_flag(&p->flags[1], SIP_PAGE2_STATECHANGEQUEUE);
cb_extensionstate((char *)p->context, (char *)p->exten, p->laststate, (void *) p);
}
}
} else if (sipmethod == SIP_BYE) {
pvt_set_needdestroy(p, "transaction completed");
}
@@ -21697,8 +21687,6 @@ static void handle_response(struct sip_pvt *p, int resp, const char *rest, struc
handle_response_invite(p, resp, rest, req, seqno);
} else if (sipmethod == SIP_BYE) {
pvt_set_needdestroy(p, "received 481 response");
} else if (sipmethod == SIP_NOTIFY) {
pvt_set_needdestroy(p, "received 481 response");
} else if (sipdebug) {
ast_debug(1, "Remote host can't match request %s to call '%s'. Giving up\n", sip_methods[sipmethod].text, p->callid);
}
@@ -30103,6 +30091,12 @@ static int setup_srtp(struct sip_srtp **srtp)
static int process_crypto(struct sip_pvt *p, struct ast_rtp_instance *rtp, struct sip_srtp **srtp, const char *a)
{
/* If no RTP instance exists for this media stream don't bother processing the crypto line */
if (!rtp) {
ast_debug(3, "Received offer with crypto line for media stream that is not enabled\n");
return FALSE;
}
if (strncasecmp(a, "crypto:", 7)) {
return FALSE;
}

View File

@@ -25,3 +25,10 @@ rtpend=20000
; do not come from the source of the RTP stream. This option is
; enabled by default.
; strictrtp=yes
;
; Number of packets containing consecutive sequence values needed
; to change the RTP source socket address. This option only comes
; into play while using strictrtp=yes. Consider changing this value
; if rtp packets are dropped from one or both ends after a call is
; connected. This option is set to 4 by default.
; probation=8

View File

@@ -4110,6 +4110,17 @@ int ast_bridge_call(struct ast_channel *chan, struct ast_channel *peer, struct a
if (!f || (f->frametype == AST_FRAME_CONTROL &&
(f->subclass.integer == AST_CONTROL_HANGUP || f->subclass.integer == AST_CONTROL_BUSY ||
f->subclass.integer == AST_CONTROL_CONGESTION))) {
/*
* If the bridge was broken for a hangup that isn't real, then
* then don't run the h extension, because the channel isn't
* really hung up. This should really only happen with AST_SOFTHANGUP_ASYNCGOTO,
* but it doesn't hurt to check AST_SOFTHANGUP_UNBRIDGE either.
*/
ast_channel_lock(chan);
if (chan->_softhangup & (AST_SOFTHANGUP_ASYNCGOTO | AST_SOFTHANGUP_UNBRIDGE)) {
ast_set_flag(chan, AST_FLAG_BRIDGE_HANGUP_DONT);
}
ast_channel_unlock(chan);
res = -1;
break;
}

View File

@@ -1012,6 +1012,7 @@ int ast_streamfile(struct ast_channel *chan, const char *filename, const char *p
struct ast_filestream *fs;
struct ast_filestream *vfs=NULL;
char fmt[256];
off_t pos;
int seekattempt;
int res;
@@ -1024,12 +1025,17 @@ int ast_streamfile(struct ast_channel *chan, const char *filename, const char *p
/* check to see if there is any data present (not a zero length file),
* done this way because there is no where for ast_openstream_full to
* return the file had no data. */
seekattempt = fseek(fs->f, -1, SEEK_END);
if (seekattempt && errno == EINVAL) {
/* Zero-length file, as opposed to a pipe */
return 0;
pos = ftello(fs->f);
seekattempt = fseeko(fs->f, -1, SEEK_END);
if (seekattempt) {
if (errno == EINVAL) {
/* Zero-length file, as opposed to a pipe */
return 0;
} else {
ast_seekstream(fs, 0, SEEK_SET);
}
} else {
ast_seekstream(fs, 0, SEEK_SET);
fseeko(fs->f, pos, SEEK_SET);
}
vfs = ast_openvstream(chan, filename, preflang);

View File

@@ -80,6 +80,8 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#define ZFONE_PROFILE_ID 0x505a
#define DEFAULT_LEARNING_MIN_SEQUENTIAL 4
extern struct ast_srtp_res *res_srtp;
static int dtmftimeout = DEFAULT_DTMF_TIMEOUT;
@@ -96,7 +98,8 @@ static int rtcpdebugport; /*< Debug only RTCP packets from IP or IP+Port if por
#ifdef SO_NO_CHECK
static int nochecksums;
#endif
static int strictrtp;
static int strictrtp; /*< Only accept RTP frames from a defined source. If we receive an indication of a changing source, enter learning mode. */
static int learning_min_sequential; /*< Number of sequential RTP frames needed from a single source during learning mode to accept new source. */
enum strict_rtp_state {
STRICT_RTP_OPEN = 0, /*! No RTP packets should be dropped, all sources accepted */
@@ -176,6 +179,13 @@ struct ast_rtp {
struct ast_sockaddr strict_rtp_address; /*!< Remote address information for strict RTP purposes */
struct ast_sockaddr alt_rtp_address; /*!<Alternate remote address information */
/*
* Learning mode values based on pjmedia's probation mode. Many of these values are redundant to the above,
* but these are in place to keep learning mode sequence values sealed from their normal counterparts.
*/
uint16_t learning_max_seq; /*!< Highest sequence number heard */
int learning_probation; /*!< Sequential packets untill source is valid */
struct rtp_red *red;
};
@@ -460,6 +470,50 @@ static int create_new_socket(const char *type, int af)
return sock;
}
/*!
* \internal
* \brief Initializes sequence values and probation for learning mode.
* \note This is an adaptation of pjmedia's pjmedia_rtp_seq_init function.
*
* \param rtp pointer to rtp struct used with the received rtp packet.
* \param seq sequence number read from the rtp header
*/
static void rtp_learning_seq_init(struct ast_rtp *rtp, uint16_t seq)
{
rtp->learning_max_seq = seq - 1;
rtp->learning_probation = learning_min_sequential;
}
/*!
* \internal
* \brief Updates sequence information for learning mode and determines if probation/learning mode should remain in effect.
* \note This function was adapted from pjmedia's pjmedia_rtp_seq_update function.
*
* \param rtp pointer to rtp struct used with the received rtp packet.
* \param seq sequence number read from the rtp header
* \return boolean value indicating if probation mode is active at the end of the function
*/
static int rtp_learning_rtp_seq_update(struct ast_rtp *rtp, uint16_t seq)
{
int probation = 1;
ast_debug(1, "%p -- probation = %d, seq = %d\n", rtp, rtp->learning_probation, seq);
if (seq == rtp->learning_max_seq + 1) {
/* packet is in sequence */
rtp->learning_probation--;
rtp->learning_max_seq = seq;
if (rtp->learning_probation == 0) {
probation = 0;
}
} else {
rtp->learning_probation = learning_min_sequential - 1;
rtp->learning_max_seq = seq;
}
return probation;
}
static int ast_rtp_new(struct ast_rtp_instance *instance,
struct ast_sched_context *sched, struct ast_sockaddr *addr,
void *data)
@@ -476,6 +530,9 @@ static int ast_rtp_new(struct ast_rtp_instance *instance,
rtp->ssrc = ast_random();
rtp->seqno = ast_random() & 0xffff;
rtp->strict_rtp_state = (strictrtp ? STRICT_RTP_LEARN : STRICT_RTP_OPEN);
if (strictrtp) {
rtp_learning_seq_init(rtp, (uint16_t)rtp->seqno);
}
/* Create a new socket for us to listen on and use */
if ((rtp->s =
@@ -2082,7 +2139,17 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
/* If strict RTP protection is enabled see if we need to learn the remote address or if we need to drop the packet */
if (rtp->strict_rtp_state == STRICT_RTP_LEARN) {
ast_debug(1, "%p -- start learning mode pass with addr = %s\n", rtp, ast_sockaddr_stringify(&addr));
/* For now, we always copy the address. */
ast_sockaddr_copy(&rtp->strict_rtp_address, &addr);
/* Send the rtp and the seqno from header to rtp_learning_rtp_seq_update to see whether we can exit or not*/
if (rtp_learning_rtp_seq_update(rtp, ntohl(rtpheader[0]))) {
ast_debug(1, "%p -- Condition for learning hasn't exited, so reject the frame.\n", rtp);
return &ast_null_frame;
}
ast_debug(1, "%p -- Probation Ended. Set strict_rtp_state to STRICT_RTP_CLOSED with address %s\n", rtp, ast_sockaddr_stringify(&addr));
rtp->strict_rtp_state = STRICT_RTP_CLOSED;
} else if (rtp->strict_rtp_state == STRICT_RTP_CLOSED) {
if (ast_sockaddr_cmp(&rtp->strict_rtp_address, &addr)) {
@@ -2497,6 +2564,7 @@ static void ast_rtp_remote_address_set(struct ast_rtp_instance *instance, struct
if (strictrtp) {
rtp->strict_rtp_state = STRICT_RTP_LEARN;
rtp_learning_seq_init(rtp, rtp->seqno);
}
return;
@@ -2884,6 +2952,7 @@ static int rtp_reload(int reload)
rtpend = DEFAULT_RTP_END;
dtmftimeout = DEFAULT_DTMF_TIMEOUT;
strictrtp = STRICT_RTP_CLOSED;
learning_min_sequential = DEFAULT_LEARNING_MIN_SEQUENTIAL;
if (cfg) {
if ((s = ast_variable_retrieve(cfg, "general", "rtpstart"))) {
rtpstart = atoi(s);
@@ -2927,6 +2996,12 @@ static int rtp_reload(int reload)
if ((s = ast_variable_retrieve(cfg, "general", "strictrtp"))) {
strictrtp = ast_true(s);
}
if ((s = ast_variable_retrieve(cfg, "general", "probation"))) {
if ((sscanf(s, "%d", &learning_min_sequential) <= 0) || learning_min_sequential <= 0) {
ast_log(LOG_WARNING, "Value for 'probation' could not be read, using default of '%d' instead\n",
DEFAULT_LEARNING_MIN_SEQUENTIAL);
}
}
ast_config_destroy(cfg);
}
if (rtpstart >= rtpend) {