[asterisk-commits] rmudgett: branch group/issue14292 r186870 - in /team/group/issue14292: ./ app...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Tue Apr 7 19:23:39 CDT 2009
Author: rmudgett
Date: Tue Apr 7 19:23:25 2009
New Revision: 186870
URL: http://svn.digium.com/svn-view/asterisk?view=rev&rev=186870
Log:
Merged revisions 186560,186581,186634,186665,186698,186736,186819 via svnmerge from
https://origsvn.digium.com/svn/asterisk/team/group/issue14068
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r186560 | rmudgett | 2009-04-03 19:58:11 -0500 (Fri, 03 Apr 2009) | 1 line
Rebase branch now that the base branch has been merged to trunk.
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r186581 | root | 2009-04-06 09:19:46 -0500 (Mon, 06 Apr 2009) | 25 lines
Merged revisions 186563,186566 via svnmerge from
file:///srv/subversion/repos/asterisk/trunk
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r186563 | file | 2009-04-06 08:23:12 -0500 (Mon, 06 Apr 2009) | 8 lines
Pass the correct value to sizeof when copying address information.
(issue #14827)
Reported by: pj
Patches:
14827.diff uploaded by file (license 11)
Tested by: pj
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r186566 | mmichelson | 2009-04-06 08:57:39 -0500 (Mon, 06 Apr 2009) | 8 lines
Blocked revisions 186565 via svnmerge
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r186565 | mmichelson | 2009-04-06 08:54:41 -0500 (Mon, 06 Apr 2009) | 3 lines
Revert commit 186445 because it causes the build to fail when IMAP_STORAGE is used.
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r186634 | root | 2009-04-06 11:19:17 -0500 (Mon, 06 Apr 2009) | 24 lines
Merged revisions 186620,186624 via svnmerge from
file:///srv/subversion/repos/asterisk/trunk
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r186620 | mmichelson | 2009-04-06 11:06:25 -0500 (Mon, 06 Apr 2009) | 3 lines
Silly svn. These files didn't get merged over in the merge of the issue8824 branch.
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r186624 | file | 2009-04-06 11:15:30 -0500 (Mon, 06 Apr 2009) | 13 lines
Add support for changing the outbound codec on a SIP call using
a dialplan variable.
This adds a dialplan variable (SIP_CODEC_OUTBOUND) which controls
the codec offered for an outgoing SIP call. This is much like the
SIP_CODEC dialplan variable and has the same restrictions. The codec
set must be one that is configured for the call.
(closes issue #13243)
Reported by: samdell3
Patches:
13243.diff uploaded by file (license 11)
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r186665 | root | 2009-04-06 12:19:49 -0500 (Mon, 06 Apr 2009) | 9 lines
Merged revisions 186653 via svnmerge from
file:///srv/subversion/repos/asterisk/trunk
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r186653 | file | 2009-04-06 12:03:07 -0500 (Mon, 06 Apr 2009) | 2 lines
Fix problem when authenticating a non-RTP dialog.
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r186698 | root | 2009-04-06 18:19:13 -0500 (Mon, 06 Apr 2009) | 9 lines
Merged revisions 186687 via svnmerge from
file:///srv/subversion/repos/asterisk/trunk
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r186687 | file | 2009-04-06 18:11:13 -0500 (Mon, 06 Apr 2009) | 2 lines
Fix a log message getting output when it should not have been.
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r186736 | root | 2009-04-07 16:20:29 -0500 (Tue, 07 Apr 2009) | 19 lines
Merged revisions 186720 via svnmerge from
file:///srv/subversion/repos/asterisk/trunk
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r186720 | mmichelson | 2009-04-07 15:46:18 -0500 (Tue, 07 Apr 2009) | 12 lines
Merged revisions 186719 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r186719 | mmichelson | 2009-04-07 15:43:49 -0500 (Tue, 07 Apr 2009) | 6 lines
Ensure that \r\n is printed after the ActionID in an OriginateResponse.
(closes issue #14847)
Reported by: kobaz
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r186819 | root | 2009-04-07 18:19:35 -0500 (Tue, 07 Apr 2009) | 17 lines
Merged revisions 186799 via svnmerge from
file:///srv/subversion/repos/asterisk/trunk
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r186799 | tilghman | 2009-04-07 17:23:46 -0500 (Tue, 07 Apr 2009) | 10 lines
Merged revisions 186775 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r186775 | tilghman | 2009-04-07 17:16:50 -0500 (Tue, 07 Apr 2009) | 3 lines
Fix Macro documentation to match current (and intended) behavior.
(See -dev mailing list)
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Modified:
team/group/issue14292/ (props changed)
team/group/issue14292/CHANGES
team/group/issue14292/apps/app_macro.c
team/group/issue14292/channels/chan_sip.c
team/group/issue14292/doc/tex/channelvariables.tex
team/group/issue14292/main/manager.c
team/group/issue14292/main/rtp_engine.c
team/group/issue14292/res/res_rtp_asterisk.c
Propchange: team/group/issue14292/
------------------------------------------------------------------------------
Binary property 'branch-1.4-blocked' - no diff available.
Propchange: team/group/issue14292/
------------------------------------------------------------------------------
Binary property 'branch-1.4-merged' - no diff available.
Propchange: team/group/issue14292/
('issue14068-integrated' removed)
Propchange: team/group/issue14292/
------------------------------------------------------------------------------
--- issue14292-integrated (original)
+++ issue14292-integrated Tue Apr 7 19:23:25 2009
@@ -1,1 +1,1 @@
-/team/group/issue14068:1-186531
+/team/group/issue14068:1-186846
Propchange: team/group/issue14292/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Tue Apr 7 19:23:25 2009
@@ -1,1 +1,1 @@
-/trunk:1-186519
+/trunk:1-186818
Modified: team/group/issue14292/CHANGES
URL: http://svn.digium.com/svn-view/asterisk/team/group/issue14292/CHANGES?view=diff&rev=186870&r1=186869&r2=186870
==============================================================================
--- team/group/issue14292/CHANGES (original)
+++ team/group/issue14292/CHANGES Tue Apr 7 19:23:25 2009
@@ -15,6 +15,9 @@
-----------
* Added preferred_codec_only option in sip.conf. This feature limits the joint
codecs sent in response to an INVITE to the single most preferred codec.
+ * Added SIP_CODEC_OUTBOUND dialplan variable which can be used to set the codec
+ to be used for the outgoing call. It must be one of the codecs configured
+ for the device.
Applications
------------
Modified: team/group/issue14292/apps/app_macro.c
URL: http://svn.digium.com/svn-view/asterisk/team/group/issue14292/apps/app_macro.c?view=diff&rev=186870&r1=186869&r2=186870
==============================================================================
--- team/group/issue14292/apps/app_macro.c (original)
+++ team/group/issue14292/apps/app_macro.c Tue Apr 7 19:23:25 2009
@@ -62,10 +62,6 @@
at the location of the Goto.</para>
<para>If <variable>MACRO_OFFSET</variable> is set at termination, Macro will attempt to continue
at priority MACRO_OFFSET + N + 1 if such a step exists, and N + 1 otherwise.</para>
- <para>Extensions: While a macro is being executed, it becomes the current context. This means that if
- a hangup occurs, for instance, that the macro will be searched for an <literal>h</literal> extension,
- NOT the context from which the macro was called. So, make sure to define all appropriate extensions
- in your macro! (Note: AEL does not use macros)</para>
<warning><para>Because of the way Macro is implemented (it executes the priorities contained within
it via sub-engine), and a fixed per-thread memory stack allowance, macros are limited to 7 levels
of nesting (macro calling macro calling macro, etc.); It may be possible that stack-intensive
Modified: team/group/issue14292/channels/chan_sip.c
URL: http://svn.digium.com/svn-view/asterisk/team/group/issue14292/channels/chan_sip.c?view=diff&rev=186870&r1=186869&r2=186870
==============================================================================
--- team/group/issue14292/channels/chan_sip.c (original)
+++ team/group/issue14292/channels/chan_sip.c Tue Apr 7 19:23:25 2009
@@ -5836,7 +5836,12 @@
int fmt;
const char *codec;
- codec = pbx_builtin_getvar_helper(p->owner, "SIP_CODEC");
+ if (p->outgoing_call) {
+ codec = pbx_builtin_getvar_helper(p->owner, "SIP_CODEC_OUTBOUND");
+ } else if (!(codec = pbx_builtin_getvar_helper(p->owner, "SIP_CODEC_INBOUND"))) {
+ codec = pbx_builtin_getvar_helper(p->owner, "SIP_CODEC");
+ }
+
if (!codec)
return;
@@ -9838,6 +9843,7 @@
if (p->do_history)
append_history(p, "ReInv", "Re-invite sent");
+ try_suggested_sip_codec(p);
if (t38version)
add_sdp(&req, p, oldsdp, FALSE, TRUE);
else
@@ -10199,8 +10205,10 @@
ast_udptl_offered_from_local(p->udptl, 1);
ast_debug(1, "T38 is in state %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>");
add_sdp(&req, p, FALSE, FALSE, TRUE);
- } else if (p->rtp)
+ } else if (p->rtp) {
+ try_suggested_sip_codec(p);
add_sdp(&req, p, FALSE, TRUE, FALSE);
+ }
} else {
if (!p->notify_headers) {
add_header_contentLength(&req, 0);
@@ -13450,8 +13458,10 @@
if (p->t38.peercapability)
p->t38.jointcapability &= p->t38.peercapability;
if (!dialog_initialize_rtp(p)) {
- ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(p->rtp), p->rtp, &peer->prefs);
- p->autoframing = peer->autoframing;
+ if (p->rtp) {
+ ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(p->rtp), p->rtp, &peer->prefs);
+ p->autoframing = peer->autoframing;
+ }
} else {
res = AUTH_RTP_FAILED;
}
Modified: team/group/issue14292/doc/tex/channelvariables.tex
URL: http://svn.digium.com/svn-view/asterisk/team/group/issue14292/doc/tex/channelvariables.tex?view=diff&rev=186870&r1=186869&r2=186870
==============================================================================
--- team/group/issue14292/doc/tex/channelvariables.tex (original)
+++ team/group/issue14292/doc/tex/channelvariables.tex Tue Apr 7 19:23:25 2009
@@ -925,7 +925,9 @@
${SIPFROMDOMAIN} Set SIP domain on outbound calls
${SIPUSERAGENT} * SIP user agent (deprecated)
${SIPURI} * SIP uri
-${SIP_CODEC} Set the SIP codec for a call
+${SIP_CODEC} Set the SIP codec for an inbound call
+${SIP_CODEC_INBOUND} Set the SIP codec for an inbound call
+${SIP_CODEC_OUTBOUND} Set the SIP codec for an outbound call
${SIP_URI_OPTIONS} * additional options to add to the URI for an outgoing call
${RTPAUDIOQOS} RTCP QoS report for the audio of this call
${RTPVIDEOQOS} RTCP QoS report for the video of this call
Modified: team/group/issue14292/main/manager.c
URL: http://svn.digium.com/svn-view/asterisk/team/group/issue14292/main/manager.c?view=diff&rev=186870&r1=186869&r2=186870
==============================================================================
--- team/group/issue14292/main/manager.c (original)
+++ team/group/issue14292/main/manager.c Tue Apr 7 19:23:25 2009
@@ -2344,7 +2344,7 @@
snprintf(requested_channel, AST_CHANNEL_NAME, "%s/%s", in->tech, in->data);
/* Tell the manager what happened with the channel */
manager_event(EVENT_FLAG_CALL, "OriginateResponse",
- "%s"
+ "%s%s"
"Response: %s\r\n"
"Channel: %s\r\n"
"Context: %s\r\n"
@@ -2353,7 +2353,8 @@
"Uniqueid: %s\r\n"
"CallerIDNum: %s\r\n"
"CallerIDName: %s\r\n",
- in->idtext, res ? "Failure" : "Success", chan ? chan->name : requested_channel, in->context, in->exten, reason,
+ in->idtext, ast_strlen_zero(in->idtext) ? "" : "\r\n", res ? "Failure" : "Success",
+ chan ? chan->name : requested_channel, in->context, in->exten, reason,
chan ? chan->uniqueid : "<null>",
S_OR(in->cid_num, "<unknown>"),
S_OR(in->cid_name, "<unknown>")
@@ -2451,7 +2452,7 @@
res = -1;
} else {
if (!ast_strlen_zero(id))
- snprintf(fast->idtext, sizeof(fast->idtext), "ActionID: %s\r\n", id);
+ snprintf(fast->idtext, sizeof(fast->idtext), "ActionID: %s", id);
ast_copy_string(fast->tech, tech, sizeof(fast->tech));
ast_copy_string(fast->data, data, sizeof(fast->data));
ast_copy_string(fast->app, app, sizeof(fast->app));
Modified: team/group/issue14292/main/rtp_engine.c
URL: http://svn.digium.com/svn-view/asterisk/team/group/issue14292/main/rtp_engine.c?view=diff&rev=186870&r1=186869&r2=186870
==============================================================================
--- team/group/issue14292/main/rtp_engine.c (original)
+++ team/group/issue14292/main/rtp_engine.c Tue Apr 7 19:23:25 2009
@@ -374,7 +374,7 @@
if ((address->sin_family != AF_INET) ||
(address->sin_port != instance->local_address.sin_port) ||
(address->sin_addr.s_addr != instance->local_address.sin_addr.s_addr)) {
- memcpy(address, &instance->local_address, sizeof(address));
+ memcpy(address, &instance->local_address, sizeof(*address));
return 1;
}
@@ -386,7 +386,7 @@
if ((address->sin_family != AF_INET) ||
(address->sin_port != instance->remote_address.sin_port) ||
(address->sin_addr.s_addr != instance->remote_address.sin_addr.s_addr)) {
- memcpy(address, &instance->remote_address, sizeof(address));
+ memcpy(address, &instance->remote_address, sizeof(*address));
return 1;
}
Modified: team/group/issue14292/res/res_rtp_asterisk.c
URL: http://svn.digium.com/svn-view/asterisk/team/group/issue14292/res/res_rtp_asterisk.c?view=diff&rev=186870&r1=186869&r2=186870
==============================================================================
--- team/group/issue14292/res/res_rtp_asterisk.c (original)
+++ team/group/issue14292/res/res_rtp_asterisk.c Tue Apr 7 19:23:25 2009
@@ -1308,8 +1308,10 @@
samples = ntohl(*((unsigned int *)(data)));
samples &= 0xFFFF;
- ast_verbose("Got RTP RFC2833 from %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u, mark %d, event %08x, end %d, duration %-5.5d) \n", ast_inet_ntoa(remote_address.sin_addr),
- ntohs(remote_address.sin_port), payloadtype, seqno, timestamp, len, (mark?1:0), event, ((event_end & 0x80)?1:0), samples);
+ if (rtp_debug_test_addr(&remote_address)) {
+ ast_verbose("Got RTP RFC2833 from %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u, mark %d, event %08x, end %d, duration %-5.5d) \n", ast_inet_ntoa(remote_address.sin_addr),
+ ntohs(remote_address.sin_port), payloadtype, seqno, timestamp, len, (mark?1:0), event, ((event_end & 0x80)?1:0), samples);
+ }
/* Print out debug if turned on */
if (rtpdebug || option_debug > 2)
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