[asterisk-commits] rmudgett: branch rmudgett/misdn_facility r175545 - in /team/rmudgett/misdn_fa...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Feb 13 09:45:33 CST 2009
Author: rmudgett
Date: Fri Feb 13 09:45:32 2009
New Revision: 175545
URL: http://svn.digium.com/svn-view/asterisk?view=rev&rev=175545
Log:
Resolved conflict and restarted automerge
Merged revisions 175404,175421,175485,175522 via svnmerge from
https://origsvn.digium.com/svn/asterisk/team/group/issue8824
................
r175404 | mmichelson | 2009-02-12 17:17:11 -0600 (Thu, 12 Feb 2009) | 3 lines
Resolve conflict and reset automerge
................
r175421 | root | 2009-02-12 18:19:50 -0600 (Thu, 12 Feb 2009) | 39 lines
Merged revisions 175408,175411 via svnmerge from
file:///srv/subversion/repos/asterisk/trunk
................
r175408 | mmichelson | 2009-02-12 17:23:47 -0600 (Thu, 12 Feb 2009) | 17 lines
Blocked revisions 175407 via svnmerge
........
r175407 | mmichelson | 2009-02-12 17:22:44 -0600 (Thu, 12 Feb 2009) | 12 lines
Fix a place where filestreams were not refcounted properly
This section was already present in trunk and other branches,
but did not exist in 1.4.
(closes issue #14395)
Reported by: ZX81
Patches:
14395.patch uploaded by putnopvut (license 60)
Tested by: ZX81
........
................
r175411 | dhubbard | 2009-02-12 18:13:38 -0600 (Thu, 12 Feb 2009) | 13 lines
Add dynamic fax buffer configuration option to chan_dahdi.conf
When the 'faxdetect' configuration option is used, one may also want to use
the 'faxbuffers' configuration option in chan_dahdi.conf. This option will
dynamically use the configured 'faxbuffers' buffer policy on a channel for
the life of the call following the detection of fax tones. The faxbuffers
buffer policy will be reverted during call teardown.
An example use of 'faxbuffers' is below. This example would switch to using
6 buffers with a full buffer policy.
faxbuffers=>6,full
................
................
r175485 | root | 2009-02-12 23:19:02 -0600 (Thu, 12 Feb 2009) | 9 lines
Merged revisions 175475 via svnmerge from
file:///srv/subversion/repos/asterisk/trunk
........
r175475 | dhubbard | 2009-02-12 22:22:35 -0600 (Thu, 12 Feb 2009) | 1 line
add 'faxbuffers' configuration option information to CHANGES
........
................
r175522 | root | 2009-02-13 08:19:12 -0600 (Fri, 13 Feb 2009) | 25 lines
Merged revisions 175508,175512 via svnmerge from
file:///srv/subversion/repos/asterisk/trunk
........
r175508 | kpfleming | 2009-02-13 07:35:24 -0600 (Fri, 13 Feb 2009) | 15 lines
Add basic (passthrough, playback, record) support for ITU G.722.1 and G.722.1C (also known as Siren7 and Siren14)
This patch adds passthrough, file recording and file playback support for the codecs listed above, with negotiation over SIP/SDP supported. Due to Asterisk's current limitation of treating a codec/bitrate combination as a unique codec, only G.722.1 at 32 kbps and G.722.1C at 48 kbps are supported.
Along the way, some related work was done:
1) The rtpPayloadType structure definition, used as a return result for an API call in rtp.h, was moved from rtp.c to rtp.h so that the API call was actually usable. The only previous used of the API all was chan_h323.c, which had a duplicate of the structure definition instead of doing it the right way.
2) The hardcoded SDP sample rates for various codecs in chan_sip.c were removed, in favor of storing these sample rates in rtp.c along with the codec definitions there. A new API call was added to allow retrieval of the sample rate for a given codec.
3) Some basic 'a=fmtp' parsing for SDP was added to chan_sip, because chan_sip *must* decline any media streams offered for these codecs that are not at the bitrates that we support (otherwise Bad Things (TM) would result).
Review: http://reviewboard.digium.com/r/158/
........
r175512 | kpfleming | 2009-02-13 07:41:52 -0600 (Fri, 13 Feb 2009) | 3 lines
document G.722.1/.1C support
........
................
Added:
team/rmudgett/misdn_facility/formats/format_siren14.c
- copied unchanged from r175522, team/group/issue8824/formats/format_siren14.c
team/rmudgett/misdn_facility/formats/format_siren7.c
- copied unchanged from r175522, team/group/issue8824/formats/format_siren7.c
Modified:
team/rmudgett/misdn_facility/ (props changed)
team/rmudgett/misdn_facility/CHANGES
team/rmudgett/misdn_facility/channels/chan_dahdi.c
team/rmudgett/misdn_facility/channels/chan_h323.c
team/rmudgett/misdn_facility/channels/chan_iax2.c
team/rmudgett/misdn_facility/channels/chan_sip.c
team/rmudgett/misdn_facility/configs/chan_dahdi.conf.sample
team/rmudgett/misdn_facility/configs/iax.conf.sample
team/rmudgett/misdn_facility/include/asterisk/frame.h
team/rmudgett/misdn_facility/include/asterisk/rtp.h
team/rmudgett/misdn_facility/main/frame.c
team/rmudgett/misdn_facility/main/rtp.c
team/rmudgett/misdn_facility/main/udptl.c
Propchange: team/rmudgett/misdn_facility/
------------------------------------------------------------------------------
automerge = *
Propchange: team/rmudgett/misdn_facility/
------------------------------------------------------------------------------
Binary property 'branch-1.4-blocked' - no diff available.
Propchange: team/rmudgett/misdn_facility/
------------------------------------------------------------------------------
Binary property 'branch-1.4-merged' - no diff available.
Propchange: team/rmudgett/misdn_facility/
------------------------------------------------------------------------------
--- misdn-facility (original)
+++ misdn-facility Fri Feb 13 09:45:32 2009
@@ -1,1 +1,1 @@
-/team/group/issue8824:1-175392
+/team/group/issue8824:1-175544
Propchange: team/rmudgett/misdn_facility/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Fri Feb 13 09:45:32 2009
@@ -1,1 +1,1 @@
-/trunk:1-175311
+/trunk:1-175521
Modified: team/rmudgett/misdn_facility/CHANGES
URL: http://svn.digium.com/svn-view/asterisk/team/rmudgett/misdn_facility/CHANGES?view=diff&rev=175545&r1=175544&r2=175545
==============================================================================
--- team/rmudgett/misdn_facility/CHANGES (original)
+++ team/rmudgett/misdn_facility/CHANGES Fri Feb 13 09:45:32 2009
@@ -116,6 +116,7 @@
first INVITE is generated - SIPRemoveHeader()
* Channel variables set with setvar= in a device configuration is now
set both for inbound and outbound calls.
+ * Added support for ITU G.722.1 and G.722.1C (Siren7 and Siren14) media streams.
Skinny Changes
--------------
@@ -127,6 +128,13 @@
-------------
* The UK option waitfordialtone has been added for use with BT analog
lines.
+ * Added a 'faxbuffers' configuration option to chan_dahdi.conf. This option
+ is used in conjunction with the 'faxdetect' configuration option. When
+ 'faxbuffers' is used and fax tones are detected, the channel will dynamically
+ switch to the configured faxbuffers policy. For example, to use 6 buffers
+ and a 'full' buffer policy for a fax transmission, add:
+ faxbuffers=>6,full
+ The faxbuffers configuration will be in affect until the call is torn down.
Dialplan Functions
------------------
@@ -196,6 +204,8 @@
* The contrib/scripts/ directory now has a script called sip_nat_settings that will
give you the correct output for an asterisk box behind nat. It will give you the
externhost and localnet settings.
+ * The Asterisk core now supports ITU G.722.1 and G.722.1C media streams, and
+ can connect calls in passthrough mode, as well as record and play back files.
Asterisk Manager Interface
--------------------------
@@ -614,6 +624,7 @@
* Added support for OSP. The token is set and retrieved through the CHANNEL()
dialplan function.
* Added immediate option to iax.conf
+ * Added forceencryption option to iax.conf
XMPP Google Talk/Jingle changes
-------------------------------
Modified: team/rmudgett/misdn_facility/channels/chan_dahdi.c
URL: http://svn.digium.com/svn-view/asterisk/team/rmudgett/misdn_facility/channels/chan_dahdi.c?view=diff&rev=175545&r1=175544&r2=175545
==============================================================================
--- team/rmudgett/misdn_facility/channels/chan_dahdi.c (original)
+++ team/rmudgett/misdn_facility/channels/chan_dahdi.c Fri Feb 13 09:45:32 2009
@@ -583,8 +583,11 @@
struct dahdi_pvt *master; /*!< Master to us (we follow their conferencing) */
int inconference; /*!< If our real should be in the conference */
+ int bufsize; /*!< Size of the buffers */
int buf_no; /*!< Number of buffers */
int buf_policy; /*!< Buffer policy */
+ int faxbuf_no; /*!< Number of Fax buffers */
+ int faxbuf_policy; /*!< Fax buffer policy */
int sig; /*!< Signalling style */
/*!
* \brief Nonzero if the signaling type is sent over a radio.
@@ -681,6 +684,10 @@
unsigned int echocanon:1;
/*! \brief TRUE if a fax tone has already been handled. */
unsigned int faxhandled:1;
+ /*! \brief TRUE if dynamic faxbuffers are configured for use, default is OFF */
+ unsigned int usefaxbuffers:1;
+ /*! \brief TRUE while dynamic faxbuffers are in use */
+ unsigned int faxbuffersinuse:1;
/*! \brief TRUE if over a radio and dahdi_read() has been called. */
unsigned int firstradio:1;
/*!
@@ -1233,7 +1240,10 @@
.sendcalleridafter = DEFAULT_CIDRINGS,
.buf_policy = DAHDI_POLICY_IMMEDIATE,
- .buf_no = numbufs
+ .buf_no = numbufs,
+ .usefaxbuffers = 0,
+ .faxbuf_policy = DAHDI_POLICY_IMMEDIATE,
+ .faxbuf_no = numbufs,
},
.timing = {
.prewinktime = -1,
@@ -3715,6 +3725,22 @@
p->dsp = NULL;
}
+ if (p->faxbuffersinuse) {
+ /* faxbuffers are in use, revert them */
+ struct dahdi_bufferinfo bi = {
+ .txbufpolicy = p->buf_policy,
+ .rxbufpolicy = p->buf_policy,
+ .bufsize = p->bufsize,
+ .numbufs = p->buf_no
+ };
+ int bpres;
+
+ if ((bpres = ioctl(p->subs[SUB_REAL].dfd, DAHDI_SET_BUFINFO, &bi)) < 0) {
+ ast_log(LOG_WARNING, "Channel '%s' unable to revert faxbuffer policy: %s\n", ast->name, strerror(errno));
+ }
+ p->faxbuffersinuse = 0;
+ }
+
law = DAHDI_LAW_DEFAULT;
res = ioctl(p->subs[SUB_REAL].dfd, DAHDI_SETLAW, &law);
if (res < 0)
@@ -4803,6 +4829,21 @@
} else if (f->subclass == 'f') {
/* Fax tone -- Handle and return NULL */
if ((p->callprogress & CALLPROGRESS_FAX) && !p->faxhandled) {
+ /* If faxbuffers are configured, use them for the fax transmission */
+ if (p->usefaxbuffers && !p->faxbuffersinuse) {
+ struct dahdi_bufferinfo bi = {
+ .txbufpolicy = p->faxbuf_policy,
+ .bufsize = p->bufsize,
+ .numbufs = p->faxbuf_no
+ };
+ int res;
+
+ if ((res = ioctl(p->subs[idx].dfd, DAHDI_SET_BUFINFO, &bi)) < 0) {
+ ast_log(LOG_WARNING, "Channel '%s' unable to set faxbuffer policy, reason: %s\n", ast->name, strerror(errno));
+ } else {
+ p->faxbuffersinuse = 1;
+ }
+ }
p->faxhandled = 1;
if (strcmp(ast->exten, "fax")) {
const char *target_context = S_OR(ast->macrocontext, ast->context);
@@ -9276,8 +9317,19 @@
if (res < 0) {
ast_log(LOG_WARNING, "Unable to set buffer policy on channel %d: %s\n", channel, strerror(errno));
}
- } else
+ } else {
ast_log(LOG_WARNING, "Unable to check buffer policy on channel %d: %s\n", channel, strerror(errno));
+ }
+ tmp->buf_policy = conf->chan.buf_policy;
+ tmp->buf_no = conf->chan.buf_no;
+ tmp->usefaxbuffers = conf->chan.usefaxbuffers;
+ tmp->faxbuf_policy = conf->chan.faxbuf_policy;
+ tmp->faxbuf_no = conf->chan.faxbuf_no;
+ /* This is not as gnarly as it may first appear. If the ioctl above failed, we'd be setting
+ * tmp->bufsize to zero which would cause subsequent faxbuffer-related ioctl calls to fail.
+ * The reason the ioctl call above failed should to be determined before worrying about the
+ * faxbuffer-related ioctl calls */
+ tmp->bufsize = bi.bufsize;
}
#endif
tmp->immediate = conf->chan.immediate;
@@ -14489,6 +14541,27 @@
#endif
} else {
ast_log(LOG_WARNING, "Invalid policy name given (%s).\n", policy);
+ }
+ } else if (!strcasecmp(v->name, "faxbuffers")) {
+ int res;
+ char policy[21] = "";
+
+ res = sscanf(v->value, "%d,%20s", &confp->chan.faxbuf_no, policy);
+ if (res != 2) {
+ ast_log(LOG_WARNING, "Parsing faxbuffers option data failed, using defaults.\n");
+ confp->chan.faxbuf_no = numbufs;
+ continue;
+ }
+ confp->chan.usefaxbuffers = 1;
+ if (confp->chan.faxbuf_no < 0)
+ confp->chan.faxbuf_no = numbufs;
+ if (!strcasecmp(policy, "full")) {
+ confp->chan.faxbuf_policy = DAHDI_POLICY_WHEN_FULL;
+ } else if (!strcasecmp(policy, "immediate")) {
+ confp->chan.faxbuf_policy = DAHDI_POLICY_IMMEDIATE;
+ } else {
+ ast_log(LOG_WARNING, "Invalid policy name given (%s).\n", policy);
+ confp->chan.usefaxbuffers = 0;
}
} else if (!strcasecmp(v->name, "dahdichan")) {
ast_copy_string(dahdichan, v->value, sizeof(dahdichan));
Modified: team/rmudgett/misdn_facility/channels/chan_h323.c
URL: http://svn.digium.com/svn-view/asterisk/team/rmudgett/misdn_facility/channels/chan_h323.c?view=diff&rev=175545&r1=175544&r2=175545
==============================================================================
--- team/rmudgett/misdn_facility/channels/chan_h323.c (original)
+++ team/rmudgett/misdn_facility/channels/chan_h323.c Fri Feb 13 09:45:32 2009
@@ -1922,15 +1922,6 @@
return info;
}
-/*
- * Definition taken from rtp.c for rtpPayloadType because we need it here.
- */
-
-struct rtpPayloadType {
- int isAstFormat; /* whether the following code is an AST_FORMAT */
- int code;
-};
-
/*! \brief
* Call-back function passing remote ip/port information from H.323 to asterisk
*
Modified: team/rmudgett/misdn_facility/channels/chan_iax2.c
URL: http://svn.digium.com/svn-view/asterisk/team/rmudgett/misdn_facility/channels/chan_iax2.c?view=diff&rev=175545&r1=175544&r2=175545
==============================================================================
--- team/rmudgett/misdn_facility/channels/chan_iax2.c (original)
+++ team/rmudgett/misdn_facility/channels/chan_iax2.c Fri Feb 13 09:45:32 2009
@@ -379,6 +379,7 @@
IAX_IMMEDIATE = (1 << 28), /*!< Allow immediate off-hook to extension s */
IAX_SENDCONNECTEDLINE = (1 << 28), /*!< Allow sending of connected line updates */
IAX_RECVCONNECTEDLINE = (1 << 29), /*!< Allow receiving of connected line updates */
+ IAX_FORCE_ENCRYPT = (1 << 30), /*!< Forces call encryption, if encryption not possible hangup */
};
static int global_rtautoclear = 120;
@@ -1941,8 +1942,7 @@
iaxs[x]->pingid = iax2_sched_add(sched, ping_time * 1000, send_ping, (void *)(long)x);
iaxs[x]->lagid = iax2_sched_add(sched, lagrq_time * 1000, send_lagrq, (void *)(long)x);
iaxs[x]->amaflags = amaflags;
- ast_copy_flags(iaxs[x], &globalflags, IAX_NOTRANSFER | IAX_TRANSFERMEDIA | IAX_USEJITTERBUF | IAX_FORCEJITTERBUF | IAX_NOKEYROTATE | IAX_SENDCONNECTEDLINE | IAX_RECVCONNECTEDLINE);
-
+ ast_copy_flags(iaxs[x], &globalflags, IAX_NOTRANSFER | IAX_TRANSFERMEDIA | IAX_USEJITTERBUF | IAX_FORCEJITTERBUF | IAX_NOKEYROTATE | IAX_SENDCONNECTEDLINE | IAX_RECVCONNECTEDLINE | IAX_FORCE_ENCRYPT);
ast_string_field_set(iaxs[x], accountcode, accountcode);
ast_string_field_set(iaxs[x], mohinterpret, mohinterpret);
ast_string_field_set(iaxs[x], mohsuggest, mohsuggest);
@@ -3560,7 +3560,7 @@
if (peer->maxms && ((peer->lastms > peer->maxms) || (peer->lastms < 0)))
goto return_unref;
- ast_copy_flags(cai, peer, IAX_SENDANI | IAX_TRUNK | IAX_NOTRANSFER | IAX_TRANSFERMEDIA | IAX_USEJITTERBUF | IAX_FORCEJITTERBUF | IAX_NOKEYROTATE | IAX_SENDCONNECTEDLINE | IAX_RECVCONNECTEDLINE);
+ ast_copy_flags(cai, peer, IAX_SENDANI | IAX_TRUNK | IAX_NOTRANSFER | IAX_TRANSFERMEDIA | IAX_USEJITTERBUF | IAX_FORCEJITTERBUF | IAX_NOKEYROTATE | IAX_SENDCONNECTEDLINE | IAX_RECVCONNECTEDLINE | IAX_FORCE_ENCRYPT);
cai->maxtime = peer->maxms;
cai->capability = peer->capability;
cai->encmethods = peer->encmethods;
@@ -3762,16 +3762,17 @@
ast_log(LOG_WARNING, "No peer provided in the IAX2 dial string '%s'\n", dest);
return -1;
}
-
if (!pds.exten) {
pds.exten = defaultrdest;
}
-
if (create_addr(pds.peer, c, &sin, &cai)) {
ast_log(LOG_WARNING, "No address associated with '%s'\n", pds.peer);
return -1;
}
-
+ if (ast_strlen_zero(cai.secret) && ast_test_flag(iaxs[callno], IAX_FORCE_ENCRYPT)) {
+ ast_log(LOG_WARNING, "Call terminated. No secret given and force encrypt enabled\n");
+ return -1;
+ }
if (!pds.username && !ast_strlen_zero(cai.username))
pds.username = cai.username;
if (!pds.password && !ast_strlen_zero(cai.secret))
@@ -6232,11 +6233,7 @@
if (user->maxauthreq > 0)
ast_set_flag(iaxs[callno], IAX_MAXAUTHREQ);
iaxs[callno]->prefs = user->prefs;
- ast_copy_flags(iaxs[callno], user, IAX_CODEC_USER_FIRST);
- ast_copy_flags(iaxs[callno], user, IAX_IMMEDIATE);
- ast_copy_flags(iaxs[callno], user, IAX_CODEC_NOPREFS);
- ast_copy_flags(iaxs[callno], user, IAX_CODEC_NOCAP);
- ast_copy_flags(iaxs[callno], user, IAX_NOKEYROTATE);
+ ast_copy_flags(iaxs[callno], user, IAX_CODEC_USER_FIRST | IAX_IMMEDIATE | IAX_CODEC_NOPREFS | IAX_CODEC_NOCAP | IAX_NOKEYROTATE | IAX_FORCE_ENCRYPT);
iaxs[callno]->encmethods = user->encmethods;
/* Store the requested username if not specified */
if (ast_strlen_zero(iaxs[callno]->username))
@@ -6414,7 +6411,10 @@
ast_string_field_set(p, host, user->name);
user = user_unref(user);
}
-
+ if (ast_test_flag(p, IAX_FORCE_ENCRYPT) && !p->encmethods) {
+ ast_log(LOG_NOTICE, "Call Terminated, Incomming call is unencrypted while force encrypt is enabled.");
+ return res;
+ }
if (!ast_test_flag(&p->state, IAX_STATE_AUTHENTICATED))
return res;
if (ies->password)
@@ -6734,8 +6734,13 @@
}
}
}
- if (ies->encmethods)
+
+ if (ies->encmethods) {
ast_set_flag(p, IAX_ENCRYPTED | IAX_KEYPOPULATED);
+ } else if (ast_test_flag(iaxs[callno], IAX_FORCE_ENCRYPT)) {
+ ast_log(LOG_NOTICE, "Call initiated without encryption while forceencryption=yes option is set");
+ return -1; /* if force encryption is yes, and no encryption methods, then return -1 to hangup */
+ }
if (!res) {
struct ast_datastore *variablestore;
struct ast_variable *var, *prev = NULL;
@@ -8850,6 +8855,11 @@
auth_fail(fr->callno, IAX_COMMAND_REJECT);
if (authdebug)
ast_log(LOG_NOTICE, "Rejected connect attempt from %s, who was trying to reach '%s@%s'\n", ast_inet_ntoa(sin.sin_addr), iaxs[fr->callno]->exten, iaxs[fr->callno]->context);
+ break;
+ }
+ if (ast_strlen_zero(iaxs[fr->callno]->secret) && ast_test_flag(iaxs[fr->callno], IAX_FORCE_ENCRYPT)) {
+ auth_fail(fr->callno, IAX_COMMAND_REJECT);
+ ast_log(LOG_WARNING, "Rejected connect attempt. No secret present while force encrypt enabled.\n");
break;
}
if (strcasecmp(iaxs[fr->callno]->exten, "TBD")) {
@@ -10695,7 +10705,7 @@
if (ast_test_flag(&globalflags, IAX_NOKEYROTATE)) {
ast_copy_flags(peer, &globalflags, IAX_NOKEYROTATE);
}
- ast_copy_flags(peer, &globalflags, IAX_USEJITTERBUF | IAX_FORCEJITTERBUF | IAX_SENDCONNECTEDLINE | IAX_RECVCONNECTEDLINE);
+ ast_copy_flags(peer, &globalflags, IAX_USEJITTERBUF | IAX_FORCEJITTERBUF | IAX_SENDCONNECTEDLINE | IAX_RECVCONNECTEDLINE | IAX_FORCE_ENCRYPT);
peer->encmethods = iax2_encryption;
peer->adsi = adsi;
ast_string_field_set(peer,secret,"");
@@ -10745,6 +10755,18 @@
peer->authmethods = get_auth_methods(v->value);
} else if (!strcasecmp(v->name, "encryption")) {
peer->encmethods = get_encrypt_methods(v->value);
+ if (!peer->encmethods) {
+ ast_clear_flag(peer, IAX_FORCE_ENCRYPT);
+ }
+ } else if (!strcasecmp(v->name, "forceencryption")) {
+ if (ast_false(v->value)) {
+ ast_clear_flag(peer, IAX_FORCE_ENCRYPT);
+ } else {
+ peer->encmethods = get_encrypt_methods(v->value);
+ if (peer->encmethods) {
+ ast_set_flag(peer, IAX_FORCE_ENCRYPT);
+ }
+ }
} else if (!strcasecmp(v->name, "keyrotate")) {
if (ast_false(v->value))
ast_set_flag(peer, IAX_NOKEYROTATE);
@@ -10973,7 +10995,7 @@
user->adsi = adsi;
ast_string_field_set(user, name, name);
ast_string_field_set(user, language, language);
- ast_copy_flags(user, &globalflags, IAX_USEJITTERBUF | IAX_FORCEJITTERBUF | IAX_CODEC_USER_FIRST | IAX_CODEC_NOPREFS | IAX_CODEC_NOCAP | IAX_NOKEYROTATE | IAX_SENDCONNECTEDLINE | IAX_RECVCONNECTEDLINE);
+ ast_copy_flags(user, &globalflags, IAX_USEJITTERBUF | IAX_FORCEJITTERBUF | IAX_CODEC_USER_FIRST | IAX_CODEC_NOPREFS | IAX_CODEC_NOCAP | IAX_NOKEYROTATE | IAX_SENDCONNECTEDLINE | IAX_RECVCONNECTEDLINE | IAX_FORCE_ENCRYPT);
ast_clear_flag(user, IAX_HASCALLERID);
ast_string_field_set(user, cid_name, "");
ast_string_field_set(user, cid_num, "");
@@ -11019,6 +11041,18 @@
user->authmethods = get_auth_methods(v->value);
} else if (!strcasecmp(v->name, "encryption")) {
user->encmethods = get_encrypt_methods(v->value);
+ if (!user->encmethods) {
+ ast_clear_flag(user, IAX_FORCE_ENCRYPT);
+ }
+ } else if (!strcasecmp(v->name, "forceencryption")) {
+ if (ast_false(v->value)) {
+ ast_clear_flag(user, IAX_FORCE_ENCRYPT);
+ } else {
+ user->encmethods = get_encrypt_methods(v->value);
+ if (user->encmethods) {
+ ast_set_flag(user, IAX_FORCE_ENCRYPT);
+ }
+ }
} else if (!strcasecmp(v->name, "keyrotate")) {
if (ast_false(v->value))
ast_set_flag(user, IAX_NOKEYROTATE);
@@ -11410,11 +11444,23 @@
ast_netsock_unref(ns);
}
}
- } else if (!strcasecmp(v->name, "authdebug"))
+ } else if (!strcasecmp(v->name, "authdebug")) {
authdebug = ast_true(v->value);
- else if (!strcasecmp(v->name, "encryption"))
- iax2_encryption = get_encrypt_methods(v->value);
- else if (!strcasecmp(v->name, "keyrotate")) {
+ } else if (!strcasecmp(v->name, "encryption")) {
+ iax2_encryption = get_encrypt_methods(v->value);
+ if (!iax2_encryption) {
+ ast_clear_flag((&globalflags), IAX_FORCE_ENCRYPT);
+ }
+ } else if (!strcasecmp(v->name, "forceencryption")) {
+ if (ast_false(v->value)) {
+ ast_clear_flag((&globalflags), IAX_FORCE_ENCRYPT);
+ } else {
+ iax2_encryption = get_encrypt_methods(v->value);
+ if (iax2_encryption) {
+ ast_set_flag((&globalflags), IAX_FORCE_ENCRYPT);
+ }
+ }
+ } else if (!strcasecmp(v->name, "keyrotate")) {
if (ast_false(v->value))
ast_set_flag((&globalflags), IAX_NOKEYROTATE);
else
Modified: team/rmudgett/misdn_facility/channels/chan_sip.c
URL: http://svn.digium.com/svn-view/asterisk/team/rmudgett/misdn_facility/channels/chan_sip.c?view=diff&rev=175545&r1=175544&r2=175545
==============================================================================
--- team/rmudgett/misdn_facility/channels/chan_sip.c (original)
+++ team/rmudgett/misdn_facility/channels/chan_sip.c Fri Feb 13 09:45:32 2009
@@ -2338,10 +2338,10 @@
static const char *get_sdp(struct sip_request *req, const char *name);
static int find_sdp(struct sip_request *req);
static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action);
-static void add_codec_to_sdp(const struct sip_pvt *p, int codec, int sample_rate,
+static void add_codec_to_sdp(const struct sip_pvt *p, int codec,
struct ast_str **m_buf, struct ast_str **a_buf,
int debug, int *min_packet_size);
-static void add_noncodec_to_sdp(const struct sip_pvt *p, int format, int sample_rate,
+static void add_noncodec_to_sdp(const struct sip_pvt *p, int format,
struct ast_str **m_buf, struct ast_str **a_buf,
int debug);
static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int oldsdp);
@@ -7734,22 +7734,16 @@
iterator = req->sdp_start;
while ((a = get_sdp_iterate(&iterator, req, "a"))[0] != '\0') {
char mimeSubtype[128];
- ast_copy_string(mimeSubtype, a, sizeof(mimeSubtype));
+ char fmtp_string[64];
+ unsigned int sample_rate;
+
if (option_debug > 1) {
int breakout = FALSE;
-
+
/* If we're debugging, check for unsupported sdp options */
if (!strncasecmp(a, "rtcp:", (size_t) 5)) {
if (debug)
ast_verbose("Got unsupported a:rtcp in SDP offer \n");
- breakout = TRUE;
- } else if (!strncasecmp(a, "fmtp:", (size_t) 5)) {
- /* Format parameters: Not supported */
- /* Note: This is used for codec parameters, like bitrate for
- G722 and video formats for H263 and H264
- See RFC2327 for an example */
- if (debug)
- ast_verbose("Got unsupported a:fmtp in SDP offer \n");
breakout = TRUE;
} else if (!strncasecmp(a, "framerate:", (size_t) 10)) {
/* Video stuff: Not supported */
@@ -7770,21 +7764,29 @@
if (breakout) /* We have a match, skip to next header */
continue;
}
+
if (!strcasecmp(a, "sendonly")) {
if (sendonly == -1)
sendonly = 1;
continue;
- } else if (!strcasecmp(a, "inactive")) {
+ }
+
+ if (!strcasecmp(a, "inactive")) {
if (sendonly == -1)
sendonly = 2;
continue;
- } else if (!strcasecmp(a, "sendrecv")) {
+ }
+
+ if (!strcasecmp(a, "sendrecv")) {
if (sendonly == -1)
sendonly = 0;
continue;
- } else if (strlen(a) > 5 && !strncasecmp(a, "ptime", 5)) {
+ }
+
+ if (!strncasecmp(a, "ptime", 5)) {
char *tmp = strrchr(a, ':');
long int framing = 0;
+
if (tmp) {
tmp++;
framing = strtol(tmp, NULL, 10);
@@ -7808,8 +7810,9 @@
ast_rtp_codec_setpref(p->rtp, pref);
}
continue;
-
- } else if (!strncmp(a, red_fmtp, strlen(red_fmtp))) {
+ }
+
+ if (!strncmp(a, red_fmtp, strlen(red_fmtp))) {
/* count numbers of generations in fmtp */
red_cp = &red_fmtp[strlen(red_fmtp)];
strncpy(red_fmtp, a, 100);
@@ -7821,15 +7824,59 @@
red_cp = strtok(NULL, "/");
}
red_cp = red_fmtp;
-
- } else if (sscanf(a, "rtpmap: %u %[^/]/", &codec, mimeSubtype) == 2) {
+ continue;
+ }
+
+ if (sscanf(a, "fmtp: %u %63s", &codec, fmtp_string) == 2) {
+ struct rtpPayloadType payload;
+ unsigned int handled = 0;
+
+ payload = ast_rtp_lookup_pt(newaudiortp, codec);
+ if (!payload.code) {
+ /* it wasn't found, try the video rtp */
+ payload = ast_rtp_lookup_pt(newvideortp, codec);
+ }
+ if (payload.code && payload.isAstFormat) {
+ unsigned int bit_rate;
+
+ switch (payload.code) {
+ case AST_FORMAT_SIREN7:
+ if (sscanf(fmtp_string, "bitrate=%u", &bit_rate) == 1) {
+ if (bit_rate != 32000) {
+ ast_log(LOG_WARNING, "Got Siren7 offer at %d bps, but only 32000 bps supported; ignoring.\n", bit_rate);
+ ast_rtp_unset_m_type(newaudiortp, codec);
+ } else {
+ handled = 1;
+ }
+ }
+ break;
+ case AST_FORMAT_SIREN14:
+ if (sscanf(fmtp_string, "bitrate=%u", &bit_rate) == 1) {
+ if (bit_rate != 48000) {
+ ast_log(LOG_WARNING, "Got Siren14 offer at %d bps, but only 48000 bps supported; ignoring.\n", bit_rate);
+ ast_rtp_unset_m_type(newaudiortp, codec);
+ } else {
+ handled = 1;
+ }
+ }
+ break;
+ }
+ }
+
+ if (!handled) {
+ ast_debug(1, "Got unsupported a:%s in SDP offer\n", a);
+ }
+ continue;
+ }
+
+ if (sscanf(a, "rtpmap: %u %127[^/]/%u", &codec, mimeSubtype, &sample_rate) == 3) {
/* We have a rtpmap to handle */
if (last_rtpmap_codec < SDP_MAX_RTPMAP_CODECS) {
- /* Note: should really look at the 'freq' and '#chans' params too */
+ /* Note: should really look at the '#chans' params too */
/* Note: This should all be done in the context of the m= above */
if (!strncasecmp(mimeSubtype, "H26", 3) || !strncasecmp(mimeSubtype, "MP4", 3)) { /* Video */
- if(ast_rtp_set_rtpmap_type(newvideortp, codec, "video", mimeSubtype, 0) != -1) {
+ if (ast_rtp_set_rtpmap_type_rate(newvideortp, codec, "video", mimeSubtype, 0, sample_rate) != -1) {
if (debug)
ast_verbose("Found video description format %s for ID %d\n", mimeSubtype, codec);
found_rtpmap_codecs[last_rtpmap_codec] = codec;
@@ -7851,11 +7898,12 @@
sprintf(red_fmtp, "fmtp:%d ", red_pt);
if (debug)
- ast_verbose("Red submimetype has payload type: %d\n", red_pt);
+ ast_verbose("RED submimetype has payload type: %d\n", red_pt);
}
} else { /* Must be audio?? */
- if(ast_rtp_set_rtpmap_type(newaudiortp, codec, "audio", mimeSubtype,
- ast_test_flag(&p->flags[0], SIP_G726_NONSTANDARD) ? AST_RTP_OPT_G726_NONSTANDARD : 0) != -1) {
+ if (ast_rtp_set_rtpmap_type_rate(newaudiortp, codec, "audio", mimeSubtype,
+ ast_test_flag(&p->flags[0], SIP_G726_NONSTANDARD) ? AST_RTP_OPT_G726_NONSTANDARD : 0,
+ sample_rate) != -1) {
if (debug)
ast_verbose("Found audio description format %s for ID %d\n", mimeSubtype, codec);
found_rtpmap_codecs[last_rtpmap_codec] = codec;
@@ -9042,7 +9090,7 @@
}
/*! \brief Add codec offer to SDP offer/answer body in INVITE or 200 OK */
-static void add_codec_to_sdp(const struct sip_pvt *p, int codec, int sample_rate,
+static void add_codec_to_sdp(const struct sip_pvt *p, int codec,
struct ast_str **m_buf, struct ast_str **a_buf,
int debug, int *min_packet_size)
{
@@ -9062,18 +9110,31 @@
return;
ast_str_append(m_buf, 0, " %d", rtp_code);
ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d\r\n", rtp_code,
- ast_rtp_lookup_mime_subtype(1, codec,
- ast_test_flag(&p->flags[0], SIP_G726_NONSTANDARD) ? AST_RTP_OPT_G726_NONSTANDARD : 0),
- sample_rate);
- if (codec == AST_FORMAT_G729A) {
+ ast_rtp_lookup_mime_subtype(1, codec,
+ ast_test_flag(&p->flags[0], SIP_G726_NONSTANDARD) ? AST_RTP_OPT_G726_NONSTANDARD : 0),
+ ast_rtp_lookup_sample_rate(1, codec));
+
+ switch (codec) {
+ case AST_FORMAT_G729A:
/* Indicate that we don't support VAD (G.729 annex B) */
ast_str_append(a_buf, 0, "a=fmtp:%d annexb=no\r\n", rtp_code);
- } else if (codec == AST_FORMAT_G723_1) {
+ break;
+ case AST_FORMAT_G723_1:
/* Indicate that we don't support VAD (G.723.1 annex A) */
ast_str_append(a_buf, 0, "a=fmtp:%d annexa=no\r\n", rtp_code);
- } else if (codec == AST_FORMAT_ILBC) {
+ break;
+ case AST_FORMAT_ILBC:
/* Add information about us using only 20/30 ms packetization */
ast_str_append(a_buf, 0, "a=fmtp:%d mode=%d\r\n", rtp_code, fmt.cur_ms);
+ break;
+ case AST_FORMAT_SIREN7:
+ /* Indicate that we only expect 32Kbps */
+ ast_str_append(a_buf, 0, "a=fmtp:%d bitrate=32000\r\n", rtp_code);
+ break;
+ case AST_FORMAT_SIREN14:
+ /* Indicate that we only expect 48Kbps */
+ ast_str_append(a_buf, 0, "a=fmtp:%d bitrate=48000\r\n", rtp_code);
+ break;
}
if (fmt.cur_ms && (fmt.cur_ms < *min_packet_size))
@@ -9086,7 +9147,7 @@
/*! \brief Add video codec offer to SDP offer/answer body in INVITE or 200 OK */
/* This is different to the audio one now so we can add more caps later */
-static void add_vcodec_to_sdp(const struct sip_pvt *p, int codec, int sample_rate,
+static void add_vcodec_to_sdp(const struct sip_pvt *p, int codec,
struct ast_str **m_buf, struct ast_str **a_buf,
int debug, int *min_packet_size)
{
@@ -9103,12 +9164,13 @@
ast_str_append(m_buf, 0, " %d", rtp_code);
ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d\r\n", rtp_code,
- ast_rtp_lookup_mime_subtype(1, codec, 0), sample_rate);
+ ast_rtp_lookup_mime_subtype(1, codec, 0),
+ ast_rtp_lookup_sample_rate(1, codec));
/* Add fmtp code here */
}
/*! \brief Add text codec offer to SDP offer/answer body in INVITE or 200 OK */
-static void add_tcodec_to_sdp(const struct sip_pvt *p, int codec, int sample_rate,
+static void add_tcodec_to_sdp(const struct sip_pvt *p, int codec,
struct ast_str **m_buf, struct ast_str **a_buf,
int debug, int *min_packet_size)
{
@@ -9125,11 +9187,12 @@
ast_str_append(m_buf, 0, " %d", rtp_code);
ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d\r\n", rtp_code,
- ast_rtp_lookup_mime_subtype(1, codec, 0), sample_rate);
+ ast_rtp_lookup_mime_subtype(1, codec, 0),
+ ast_rtp_lookup_sample_rate(1, codec));
/* Add fmtp code here */
if (codec == AST_FORMAT_T140RED) {
- ast_str_append(a_buf, 0, "a=fmtp:%d %d/%d/%d\r\n", rtp_code,
+ ast_str_append(a_buf, 0, "a=fmtp:%d %d/%d/%d\r\n", rtp_code,
ast_rtp_lookup_code(p->trtp, 1, AST_FORMAT_T140),
ast_rtp_lookup_code(p->trtp, 1, AST_FORMAT_T140),
ast_rtp_lookup_code(p->trtp, 1, AST_FORMAT_T140));
@@ -9254,7 +9317,7 @@
/*! \brief Add RFC 2833 DTMF offer to SDP */
-static void add_noncodec_to_sdp(const struct sip_pvt *p, int format, int sample_rate,
+static void add_noncodec_to_sdp(const struct sip_pvt *p, int format,
struct ast_str **m_buf, struct ast_str **a_buf,
int debug)
{
@@ -9267,8 +9330,8 @@
ast_str_append(m_buf, 0, " %d", rtp_code);
ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d\r\n", rtp_code,
- ast_rtp_lookup_mime_subtype(0, format, 0),
- sample_rate);
+ ast_rtp_lookup_mime_subtype(0, format, 0),
+ ast_rtp_lookup_sample_rate(0, format));
if (format == AST_RTP_DTMF) /* Indicate we support DTMF and FLASH... */
ast_str_append(a_buf, 0, "a=fmtp:%d 0-16\r\n", rtp_code);
}
@@ -9308,13 +9371,6 @@
}
}
-
-/*!
- * \note G.722 actually is supposed to specified as 8 kHz, even though it is
- * really 16 kHz. Update this macro for other formats as they are added in
- * the future.
- */
-#define SDP_SAMPLE_RATE(x) 8000
/*! \brief Add Session Description Protocol message
@@ -9487,9 +9543,7 @@
if (capability & p->prefcodec) {
int codec = p->prefcodec & AST_FORMAT_AUDIO_MASK;
- add_codec_to_sdp(p, codec, SDP_SAMPLE_RATE(codec),
- &m_audio, &a_audio,
- debug, &min_audio_packet_size);
+ add_codec_to_sdp(p, codec, &m_audio, &a_audio, debug, &min_audio_packet_size);
alreadysent |= codec;
}
@@ -9506,9 +9560,7 @@
if (alreadysent & codec)
continue;
- add_codec_to_sdp(p, codec, SDP_SAMPLE_RATE(codec),
- &m_audio, &a_audio,
- debug, &min_audio_packet_size);
+ add_codec_to_sdp(p, codec, &m_audio, &a_audio, debug, &min_audio_packet_size);
alreadysent |= codec;
}
@@ -9521,14 +9573,11 @@
continue;
if (x & AST_FORMAT_AUDIO_MASK)
- add_codec_to_sdp(p, x, SDP_SAMPLE_RATE(x),
- &m_audio, &a_audio, debug, &min_audio_packet_size);
- else if (x & AST_FORMAT_VIDEO_MASK)
- add_vcodec_to_sdp(p, x, 90000,
- &m_video, &a_video, debug, &min_video_packet_size);
+ add_codec_to_sdp(p, x, &m_audio, &a_audio, debug, &min_audio_packet_size);
+ else if (x & AST_FORMAT_VIDEO_MASK)
+ add_vcodec_to_sdp(p, x, &m_video, &a_video, debug, &min_video_packet_size);
else if (x & AST_FORMAT_TEXT_MASK)
- add_tcodec_to_sdp(p, x, 1000,
- &m_text, &a_text, debug, &min_text_packet_size);
+ add_tcodec_to_sdp(p, x, &m_text, &a_text, debug, &min_text_packet_size);
}
/* Now add DTMF RFC2833 telephony-event as a codec */
@@ -9536,7 +9585,7 @@
if (!(p->jointnoncodeccapability & x))
continue;
- add_noncodec_to_sdp(p, x, 8000, &m_audio, &a_audio, debug);
+ add_noncodec_to_sdp(p, x, &m_audio, &a_audio, debug);
}
ast_debug(3, "-- Done with adding codecs to SDP\n");
Modified: team/rmudgett/misdn_facility/configs/chan_dahdi.conf.sample
URL: http://svn.digium.com/svn-view/asterisk/team/rmudgett/misdn_facility/configs/chan_dahdi.conf.sample?view=diff&rev=175545&r1=175544&r2=175545
==============================================================================
--- team/rmudgett/misdn_facility/configs/chan_dahdi.conf.sample (original)
+++ team/rmudgett/misdn_facility/configs/chan_dahdi.conf.sample Fri Feb 13 09:45:32 2009
@@ -735,6 +735,15 @@
;faxdetect=outgoing
;faxdetect=no
;
+; When 'faxdetect' is used, one could use 'faxbuffers' to configure the DAHDI
+; transmit buffer policy. The default is *OFF*. When this configuration
+; option is used, the faxbuffer policy will be used for the life of the call
+; after a fax tone is detected. The faxbuffer policy is reverted after the
+; call is torn down. The sample below will result in 6 buffers and a full
+; buffer policy.
+;
+;faxbuffers=>6,full
+;
; This option specifies a preference for which music on hold class this channel
; should listen to when put on hold if the music class has not been set on the
; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer
Modified: team/rmudgett/misdn_facility/configs/iax.conf.sample
URL: http://svn.digium.com/svn-view/asterisk/team/rmudgett/misdn_facility/configs/iax.conf.sample?view=diff&rev=175545&r1=175544&r2=175545
==============================================================================
--- team/rmudgett/misdn_facility/configs/iax.conf.sample (original)
+++ team/rmudgett/misdn_facility/configs/iax.conf.sample Fri Feb 13 09:45:32 2009
@@ -175,6 +175,11 @@
; Enable IAX2 encryption. The default is no.
;
; encryption = yes
+;
+; Force encryption insures no connection is established unless both sides support
+; encryption. By turning this option on, encryption is automatically turned on as well.
+;
+; forceencryption = yes
;
; This is a compatibility option for older versions of IAX2 that do not support
; key rotation with encryption. This option will disable the IAX_COMMAND_RTENC message.
Modified: team/rmudgett/misdn_facility/include/asterisk/frame.h
URL: http://svn.digium.com/svn-view/asterisk/team/rmudgett/misdn_facility/include/asterisk/frame.h?view=diff&rev=175545&r1=175544&r2=175545
==============================================================================
--- team/rmudgett/misdn_facility/include/asterisk/frame.h (original)
+++ team/rmudgett/misdn_facility/include/asterisk/frame.h Fri Feb 13 09:45:32 2009
@@ -261,6 +261,10 @@
#define AST_FORMAT_G726 (1 << 11)
/*! G.722 */
#define AST_FORMAT_G722 (1 << 12)
+/*! G.722.1 (also known as Siren7, 32kbps assumed) */
+#define AST_FORMAT_SIREN7 (1 << 13)
+/*! G.722.1 Annex C (also known as Siren14, 48kbps assumed) */
+#define AST_FORMAT_SIREN14 (1 << 14)
/*! Raw 16-bit Signed Linear (16000 Hz) PCM */
#define AST_FORMAT_SLINEAR16 (1 << 15)
/*! Maximum audio mask */
@@ -527,8 +531,8 @@
#endif
/*@} Doxygen marker */
-struct ast_format_list *ast_get_format_list_index(int index);
-struct ast_format_list *ast_get_format_list(size_t *size);
+const struct ast_format_list *ast_get_format_list_index(int index);
+const struct ast_format_list *ast_get_format_list(size_t *size);
void ast_frame_dump(const char *name, struct ast_frame *f, char *prefix);
/*! \page AudioCodecPref Audio Codec Preferences
@@ -634,10 +638,16 @@
*/
static force_inline int ast_format_rate(int format)
{
- if (format == AST_FORMAT_G722 || format == AST_FORMAT_SLINEAR16)
+ switch (format) {
+ case AST_FORMAT_G722:
+ case AST_FORMAT_SLINEAR16:
+ case AST_FORMAT_SIREN7:
return 16000;
-
- return 8000;
+ case AST_FORMAT_SIREN14:
+ return 32000;
+ default:
+ return 8000;
+ }
}
#if defined(__cplusplus) || defined(c_plusplus)
Modified: team/rmudgett/misdn_facility/include/asterisk/rtp.h
URL: http://svn.digium.com/svn-view/asterisk/team/rmudgett/misdn_facility/include/asterisk/rtp.h?view=diff&rev=175545&r1=175544&r2=175545
==============================================================================
--- team/rmudgett/misdn_facility/include/asterisk/rtp.h (original)
+++ team/rmudgett/misdn_facility/include/asterisk/rtp.h Fri Feb 13 09:45:32 2009
@@ -83,6 +83,12 @@
struct ast_rtp;
/*! T.140 Redundancy structure*/
struct rtp_red;
+
[... 620 lines stripped ...]
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