5f9897ba56
handle the GNUTLS library properly in the configure script and build system
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don't build in OSP support unless we have found and are allowed to use SSL support
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18 years ago
0b14e93d00
oops, use #ifdef instead of #if
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18 years ago
e49f14e6cb
don't reference GnuTLS headers and functions unless the configure script found it
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18 years ago
4eab565b25
Issue 7672 - fix by zandbelt - Asterisk core dump since the GnuTLS interface did not support multithreading correctly.
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18 years ago
e8addde3f1
Issue 8193 - NAT issues with gtalk/STUN. Patch by phsultan. Thanks!
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18 years ago
abc8683ce7
Issue 7686, fix by phsultan, NAT issues when calling from gtalk to SIP over nat.
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18 years ago
e4f9cc15e6
Issue #8536 - Caller ID not set in CDR for jingle
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18 years ago
0df2a42f96
Merged revisions 65837 via svnmerge from
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r65837 | file | 2007-05-24 10:40:38 -0400 (Thu, 24 May 2007) | 2 lines
Allow RFC2833 to be negotiated when an INVITE comes in without SDP and is not matched to a user or peer. (issue #9546 reported by mcrawford)
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18 years ago
6cfe6a550e
Issue 8409 - phsultan - Fix "login" as component to jabber server.
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...and, by accident, fix a bug in chan_sip for stopping a loop on retransmits
of BYE requests.
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18 years ago
9f54cd55af
Merged revisions 65767 via svnmerge from
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r65767 | crichter | 2007-05-24 11:19:58 +0200 (Do, 24 Mai 2007) | 1 line
we should only activate the generator in chan_misdn, when asterisk hask not yet taken the call (WAITING4DIGS state). Alerting audio will be generated fomr asterisk for example.
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18 years ago
e1518f42ae
start the delayed PBX when receive voice or video full frames as well, and comment this delayed-PBX activity
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18 years ago
ca6b421be4
Merged revisions 65682 via svnmerge from
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r65682 | kpfleming | 2007-05-23 16:46:22 -0400 (Wed, 23 May 2007) | 2 lines
ensure that variables are set on a newly created channel before we start a PBX on it
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18 years ago
b89faf596b
clear the 'delay PBX' flag when we are ready to start the PBX
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18 years ago
09dc4253d8
don't start a PBX on a new incoming IAX2 channel until we have some sort of response to our ACCEPT (ACK or anything else)
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18 years ago
f608d90283
Merged revisions 65676 via svnmerge from
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r65676 | kpfleming | 2007-05-23 16:06:13 -0400 (Wed, 23 May 2007) | 2 lines
if we are going to set variables on a newly created channel, it should be done *before* we start the PBX on it
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18 years ago
7398856cea
Merged revisions 65588 via svnmerge from
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r65588 | russell | 2007-05-23 08:06:17 -0500 (Wed, 23 May 2007) | 3 lines
Revert revision 62417 as someone reported problems with it to Mark. This was
related to issue #9588 .
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18 years ago
c556cc222c
List res_smdi as a dependency for app_voicemail and chan_zap
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(Thanks to mnicholson for pointing it out)
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18 years ago
e7355ec53b
Merged revisions 65328 via svnmerge from
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r65328 | crichter | 2007-05-22 09:46:39 +0200 (Di, 22 Mai 2007) | 1 line
we stop the tones only when we're in the pre-call phase, otherwise e.g. when in CONNECTED state we should not stop tones when we receive an Information Message
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18 years ago
4483fa12e8
Merged revisions 65122 via svnmerge from
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r65122 | oej | 2007-05-18 20:10:46 +0200 (Fri, 18 May 2007) | 2 lines
Not getting an ACK to a 200 OK in the initial invite is critical to the call.
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18 years ago
7fe3608300
Merged revisions 65075 via svnmerge from
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r65075 | oej | 2007-05-18 17:12:09 +0200 (Fri, 18 May 2007) | 5 lines
Issue 9235 - part of the problem, maybe not all. Please retry with this patch (and no
other patch) if you have problems with hanging SIP channels. Thank you.
A special Thank You to WeBRainstorm that gave me access to his system.
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18 years ago
50f79ba4b2
- Adding support for putting calls OFF hold with a re-invite with blank SDP. This was a bug found while doing tests at SIPit in Antwerp.
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- In order to not duplicate code, I restructured some of the code for putting calls on/off hold.
Thanks DEA for reminding me. This fix has been asleep in the videocaps branch until now.
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18 years ago
012fe116fa
Merged revisions 65007 via svnmerge from
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r65007 | crichter | 2007-05-18 13:23:11 +0200 (Fr, 18 Mai 2007) | 1 line
fixed a warning regarding Keypad encoding. encode the IE sending_complete at the right position.
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18 years ago
73d0ba053b
Issue 9487 - stop media flows at hangup of call
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18 years ago
d682a74e26
Merged revisions 64902 via svnmerge from
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r64902 | crichter | 2007-05-18 10:24:08 +0200 (Fr, 18 Mai 2007) | 1 line
we *need* to send a PROCEEDING when sending_complete is set, even if need_more_infos is requested.
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18 years ago
7a8ca54257
Even more direct RTP setup fixes! Don't allow a codec that isn't supported to creep into the SDP of either side. (issue #9446 reported by marcelbarbulescu)
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18 years ago
4ae20ba8e4
Fix auth on BYE. (Different patch than for 1.2)
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18 years ago
5acc63d688
Issue #9681 - Handle www-auth on BYE
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18 years ago
374b52f717
Final part of issue #9483 - fixing transfer() of sip calls in the dial plan (twilson)
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18 years ago
70ad556544
Issue #9439 - properly handle username parameters in SIP uri.
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18 years ago
bf1a15b9bf
Merged revisions 64535 via svnmerge from
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r64535 | oej | 2007-05-16 11:08:22 +0200 (Wed, 16 May 2007) | 2 lines
Support SIP uri's starting with SIP: and sip: (reported by Tony Mountfield on the mailing list. Thanks!)
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18 years ago
56af259505
Merged following patch with a lot of changes for 1.4
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Merged revisions 64514 via svnmerge from
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r64514 | oej | 2007-05-16 10:25:56 +0200 (Wed, 16 May 2007) | 6 lines
Issue #9726 - rlister - Better logging for ACL denials
While at it, also added better logging and handling of peers that are not supposed to register.
My patch, stole the issue report from Russell. My apologies, Russell :-)
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18 years ago
d17174cfc1
Merged revisions 64513 via svnmerge from
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r64513 | crichter | 2007-05-16 10:23:42 +0200 (Mi, 16 Mai 2007) | 1 line
in the case immediate=yes, we directly jump into the dialplan, where people can use PlayTones to indicate a Dialtone, so we don't need to to that by ourself. also we should not do a dialtone_indicate for incoming calls on a TE port in overlapdialmode.
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18 years ago
1f2afa0ff1
Change -2 to XMIT_ERROR to clarify a bit more
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18 years ago
b340fcd7d0
Properly handle AST_CONTROL_PROGRESS by just ignoring it. An unknown indication
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will trigger an error and cause sounds to stop, which in this case, is ringing.
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18 years ago
e041d57175
Handle network errors, like host or network unreachable, in a better way. This means that
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calls to hosts or qualify (OPTION) messages will fail quicker if the TCP/IP stack tells us
that there is an issue.
Since this is an unconnected UDP socket, we will not get error messages directly
in most cases, but maybe on the second and third try.
This is already implemented in trunk.
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18 years ago
8c635fb2c1
As per 9570, worrisome CDR warnings have been removed, that are either not helpful, or not relevant.
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18 years ago
9a73c07fb1
This concludes my final adventure with bitmasks and the onhold flag. Would anyone care for some peanuts?
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18 years ago
699aa6ad74
Tweak hold flags some more. They can be of three states when active: active, inactive, one direction.
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18 years ago
026000bae8
Ensure the onhold flag is set no matter what when being put on hold.
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18 years ago
c2c91c7fef
Merged revisions 63828 via svnmerge from
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r63828 | qwell | 2007-05-10 18:14:55 -0500 (Thu, 10 May 2007) | 4 lines
Fix an issue with trying to kill a thread before it gets created.
Issue 9709, patch by nic_bellamy.
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18 years ago
ca1ae5e81a
Merged revisions 63748 via svnmerge from
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r63748 | oej | 2007-05-10 22:38:54 +0200 (Thu, 10 May 2007) | 4 lines
Do not allocate SIP pvt's for PEERs we can not reach.
This was seen as a lot of dialogs being created then immediately destroyed at reload/restart of the SIP channel.
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18 years ago
67a0bbaa83
Do not prematurely go on hold if sendonly was not actually set.
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18 years ago
a181c6a777
Merged revisions 63653 via svnmerge from
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r63653 | mattf | 2007-05-09 12:20:20 -0500 (Wed, 09 May 2007) | 2 lines
Make sure we only create a DSP if it's requested on SUB_REAL
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18 years ago
c83c0072bb
Merged revisions 63610 via svnmerge from
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r63610 | file | 2007-05-09 12:51:03 -0400 (Wed, 09 May 2007) | 2 lines
Properly handle hints that point to multiple devices in chan_sip. Why chan_sip is even doing this I have no idea but I would rather not go into a rant. (issue #9536 reported by rlister)
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18 years ago
3e63c9d542
Merged revisions 62945,63402,63519 via svnmerge from
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r62945 | crichter | 2007-05-03 17:39:21 +0200 (Do, 03 Mai 2007) | 1 line
when we're in state WAITING4DIGS, we use the asterisk tone-generator which prods us, so we can't just return -1 in misdn_write in this case. Added a MISDN_KEYPAD channel variable, and fixed the sending of keypad. this enables us to modify the call forward parameters in the switch.
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r63402 | crichter | 2007-05-08 17:07:37 +0200 (Di, 08 Mai 2007) | 1 line
added application misdn_check_l2l1 which tries to pull up the L1/L2 on all ports that have the layers down in a group. It waits then for a timeout. This helps for scenarios where multiple PMP BRIs are grouped together, or where a provider has a faulty PTP Implementation, that looses the L2 after a while.
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r63519 | crichter | 2007-05-09 13:26:16 +0200 (Mi, 09 Mai 2007) | 1 line
release_chan frees ch, so we should never touch ch after release_chan, this may cause segfaults.
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18 years ago
d30faa1dc2
Don't retransmit 200 OK's on ignore status. (Reported on asterisk-users)
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18 years ago
b50b92ab77
Minor backport of revision 59083 in trunk. Don't queue an unhold frame up if the call was never on hold to begin with.
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18 years ago
95de3fbf0c
Merged revisions 62987 via svnmerge from
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r62987 | file | 2007-05-03 13:42:19 -0300 (Thu, 03 May 2007) | 2 lines
When a peer is seeded or built tell the devicestate core to update it's status. This is easier then having chan_sip load before pbx_config. (issue #9658 reported by dlynes)
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18 years ago
460e677ea6
Merged revisions 61357,61770,62885 via svnmerge from
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r61357 | crichter | 2007-04-11 14:05:57 +0200 (Mi, 11 Apr 2007) | 1 line
some fixes for PMP Hold/Retrieve, it should work now, when briding=no
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r61770 | crichter | 2007-04-24 15:50:05 +0200 (Di, 24 Apr 2007) | 1 line
added lock for sending messages to avoid double sending. shuffled some empty_chans after the cb_event calls, this avoids that a release_complete from a quite different call releases a fresh created setup by accident.
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r62885 | crichter | 2007-05-03 15:59:00 +0200 (Do, 03 Mai 2007) | 1 line
fixed the problem that misdn_write did not return -1 when called with 0 samples in a frame this resultet in a deadlock in some circumstances, when the call ended because of a busy extension. added encoding of keypad.
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18 years ago
8bcfcfca88
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r62691 | tilghman | 2007-05-02 12:38:16 -0500 (Wed, 02 May 2007) | 4 lines
Issue 9638 - if a text frame is sent with no terminating NULL through a bridged
IAX connection, the remote end will receive garbage characters tacked onto the
end.
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18 years ago
55f4eb3e3d
a)In chan_zap, set the clid, src fields in channel_alloc call. b)in the channel_alloc func, set the cid_num and name fields from the arglist[blush]. c) don't update the channel app & app data fields if you are in the 'h' extension. d)the load_module func in cdr_radius needs to return DECLINE, SUCCESS.
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18 years ago
d5fda03428
Don't unlock a channel that we already know does not exist (propably isue 8228)
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18 years ago
3595c5fed1
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r62417 | russell | 2007-04-30 10:57:26 -0500 (Mon, 30 Apr 2007) | 4 lines
This patch fixes an issue where depending on the cause code, when the network
sends a PRI disconnect, the call may not be properly hung up.
(issue #9588 , reported and patched by softins)
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18 years ago
8936804377
Fix a bug that made the "language" setting in zapata.conf not
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functional. (issue #9626 , reported and fixed by sergee)
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18 years ago
6b76a95f09
Fix a weird problem where when a caller talking to someone sitting behind an
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agent channel sent a digit, the digit would be played to the agent for forever.
This is because chan_agent always returned -1 from its send_digit_begin and _end
callbacks. This non-zero return value indicates to the Asterisk core that it
would like an inband DTMF generator put on the channel. However, this is the
wrong thing to do. It should *always* return 0, instead. When the digit begin
and end functions are called on the proxied channel, the underlying channel
will indicate whether inband DTMF is needed or not, and the generator will be
put on that one, and not the Agent channel.
(issue #9615 , #9616 , reported by jiddings and BigJimmy, and fixed by me)
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18 years ago
34f9e0c4dd
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r62126 | oej | 2007-04-27 15:57:45 +0200 (Fri, 27 Apr 2007) | 4 lines
Issue #7351 - SIP Cancel fails due to the wrong contact uri. Reported by PPYY, failed to fix by OEJ
final fix by wojtekka - THANKS!!!! THis was a hard one to catch.
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18 years ago
b0121ca8a3
Issue #9608 - fix some annoying DEBUG messages not controlled by option_debug (DEA). Thanks!
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18 years ago
0525347958
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r62037 | file | 2007-04-26 12:30:57 -0400 (Thu, 26 Apr 2007) | 2 lines
Revert previous fix for when the IAX2 channel goes funky (that's the technical term). This is causing legit calls to be prematurely hung up. (issue #9600 reported by justdave)
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18 years ago
0e525813a8
Merged revisions 61913 via svnmerge from
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r61913 | kpfleming | 2007-04-25 17:24:59 -0500 (Wed, 25 Apr 2007) | 2 lines
handle a very bizarre race condition with channels being redirected before a simple switch can be started on them (issue #9286 )
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18 years ago
5574da0c12
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r61866 | russell | 2007-04-25 16:55:23 -0500 (Wed, 25 Apr 2007) | 2 lines
If the callerid= option is specified, but empty, clear any previous data.
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18 years ago
e6aa1145ed
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r61862 | russell | 2007-04-25 16:06:22 -0500 (Wed, 25 Apr 2007) | 2 lines
Ensure that callerid settings are reset on a reload.
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18 years ago
77c52fd8bf
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r61798 | russell | 2007-04-25 11:20:38 -0500 (Wed, 25 Apr 2007) | 3 lines
Fix a typo where cid_num got copied instead of cid_ani.
(issue #9587 , reported and patched by xrg)
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18 years ago
db55615090
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r61777 | dhubbard | 2007-04-24 13:20:31 -0500 (Tue, 24 Apr 2007) | 1 line
removed #if 0 block from chan_phone, chan_zap, and chan_modem restart_monitor()
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18 years ago
b271179161
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r61771 | file | 2007-04-24 12:05:06 -0400 (Tue, 24 Apr 2007) | 2 lines
Allow RFC2833 to be sent in the response SDP when an INVITE comes in without SDP. (issue #9546 reported by mcrawford)
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18 years ago
32be6621f6
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r61685 | oej | 2007-04-19 09:56:21 +0200 (Thu, 19 Apr 2007) | 3 lines
Send NOTIFY to Contact: in SUBSCRIBE
- as reported by Intertex and Citel. Fixed during SIPit 20 in Antwerp.
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18 years ago
dc6d309bdb
#9483 , half of patch by twilson to solve 302 redirect issues
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18 years ago
eb631c00a3
For those very verbose SIP implementations that attach tons of info to the Contact header... let's increase our variable sizes. (issue #9535 reported by jeffg)
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18 years ago
8a386a9816
A fix for chan_oss that resulted from the CDR changes; it helps to use the right info.
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18 years ago
c06e063e1a
Don't assume the callid of a dialog will be set, as in some circumstances it may not. (issue #9534 reported by tecnoxarxa)
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18 years ago
5858778f40
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r61476 | russell | 2007-04-11 11:01:25 -0500 (Wed, 11 Apr 2007) | 5 lines
If someone sets the "useragent" option in sip.conf to be empty, then don't add
the User-Agent header at all. It is an optional header, anyway. Also, the bug
report says that some of Japan's SIP providers don't allow it for some weird
reason. (issue #9488 , reported by makoto, fixed by me)
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18 years ago
3bd81c31fb
Don't export AOCD variables on misdn_hangup anymore, this was mainly a fix for trunk..
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18 years ago
a0d75effd5
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r61426 | russell | 2007-04-11 10:05:36 -0500 (Wed, 11 Apr 2007) | 6 lines
Fix a bug with switching between host=dynamic and using specific hosts for
peers. The code would only reset the peer's address when it is dynamic if
it was a new peer structure. Now, it will also reset the address if it was
already in the peer list, but before the reload, it was not dynamic.
(issue #9515 , reported by caio1982, fixed by me)
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18 years ago
9ce476a89c
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r61376 | russell | 2007-04-11 09:02:54 -0500 (Wed, 11 Apr 2007) | 5 lines
Remove the attempt at reporting configuration errors in sip.conf. This can
cause a bunch of improper messages when using realtime. I give up. As oej
tried to convince me when I put this in, there is just no easy way to do it.
(inspired by a message on the -dev list)
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18 years ago
4e2ab7de64
Export AOCD variables on misdn_hangup.
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18 years ago
b31fe9be83
Ignore facility messages in case we don't have a corresponding channel object.
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18 years ago
5e08746b28
AOCD's are now exported to asterisk channel variables.
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18 years ago
50d606bbf8
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r61170 | nadi | 2007-04-10 14:31:45 +0200 (Di, 10 Apr 2007) | 2 lines
msns config parameter defaults to '*'
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18 years ago
3e5c59f235
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r61038 | oej | 2007-04-09 21:38:59 +0200 (Mon, 09 Apr 2007) | 3 lines
- Don't send ActionID before Response: header.
- Don't use a blank in an AMI header
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18 years ago
7d5a79a0b9
This is a big improvement over the current CDR fixes. It may still need refinement, but this won't have as many folks bothered.
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18 years ago
1d3fbc214a
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r60846 | tilghman | 2007-04-08 21:37:18 -0500 (Sun, 08 Apr 2007) | 2 lines
Bug 9505 - If the return value for local_queue_frame is set, then p->lock is no longer valid.
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18 years ago
dc54d451dc
Off by one error, resulting in a crash (Issue 9500)
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18 years ago
e7e3929740
Make sure we check the faxdetect option before doing fax processing
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18 years ago
96a8eadf29
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r60456 | mattf | 2007-04-06 12:03:15 -0500 (Fri, 06 Apr 2007) | 2 lines
There should only be one code path for doing DTMF conditionals on channels. This fixes it.
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18 years ago
a85156a737
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r60213 | file | 2007-04-05 08:52:50 -0400 (Thu, 05 Apr 2007) | 2 lines
Only unlock our pvt and net locks if we are actually going to try to lock the owner again. (issue #9472 reported by zoa)
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18 years ago
0b94c7c27c
Add a Content-Length of 0 to the response built by transmit_response_with_unsupported().
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(issue #9454 , reported by makoto, fixed by me)
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18 years ago
3e058dab84
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r60083 | russell | 2007-04-04 11:37:04 -0500 (Wed, 04 Apr 2007) | 4 lines
Fix the return value of handle_common_options() so that it always properly
indicates whether it handled the option or not.
(issue #9455 , reported by Netview, fixed by me)
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18 years ago
dff65e38f4
Merged revisions 59938 via svnmerge from
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r59938 | russell | 2007-04-03 14:15:04 -0500 (Tue, 03 Apr 2007) | 4 lines
Don't attempt to report configuration errors in build_user(). oej pointed out
that for a "friend" entry, this won't work, because all user options are valid
for peers, but not the other way around.
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18 years ago
f8773a1e80
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r59916 | russell | 2007-04-03 13:43:54 -0500 (Tue, 03 Apr 2007) | 3 lines
Make chan_sip report when it encounters an unknown option.
(issue #9440 , reported by nightcrawler)
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18 years ago
911a8f501d
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r59788 | nadi | 2007-04-03 11:37:00 +0200 (Di, 03 Apr 2007) | 2 lines
Use the new sysfs way of mISDN 1.2 to check if a port is NT or not.
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r59803 | nadi | 2007-04-03 12:40:58 +0200 (Di, 03 Apr 2007) | 2 lines
ptp is the 5th bit, not the 4th.
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18 years ago
30ace959bb
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r59623 | crichter | 2007-04-02 09:12:24 +0200 (Mo, 02 Apr 2007) | 1 line
we can now make 30 channels on a PRI (before we forgot chan 31..)
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r59624 | crichter | 2007-04-02 09:25:54 +0200 (Mo, 02 Apr 2007) | 1 line
don't be verbose if no need
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r59639 | crichter | 2007-04-02 14:08:12 +0200 (Mo, 02 Apr 2007) | 1 line
added option which allows us to accept incoming SETUP Messages without automatically sending Proceeding or Setup Acknowledge, this is useful with some broken switches and if you want to Release incoming calls without previously having acknowledged them. The new option is noautorespond_on_setup=yes|no default is no, so we don't break the existing behaviour
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18 years ago
8acd1e8349
When the IAX2 read callback gets called, return NULL instead of a "null frame".
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This will cause Asterisk to hangup the call instead of keep trying whatever it
was doing. Under normal conditions, this function would *never* be called.
However, the author of this patch says an error will occur that will cause it
to get called every 100 thousand calls or so. When this does happen, it puts
the channel in a loop that eventually brings down the system. So, hangup up
the call is certainly a better alternative. (issue #8286 , john)
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18 years ago
a2fc7b9e36
Fix the check that ensures that the CHANNEL function's first argument is "rtpqos".
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Thanks, Corydon. :)
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18 years ago
8fdf6e3c47
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r59258 | russell | 2007-03-27 13:04:02 -0500 (Tue, 27 Mar 2007) | 4 lines
Fix the use of the "sourceaddress" option when "bindaddr" is set to 0.0.0.0
instead of having each interface explicitly listed.
(issue #7874 , patch by stevens)
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18 years ago
c0aca26f30
Convert the RTPQOS function to just be additional parameter of the CHANNEL
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function. This way, it will be possible for other RTP based channel drivers
to expose this information in the future.
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18 years ago
10760fce2c
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r59252 | crichter | 2007-03-27 15:56:15 +0200 (Di, 27 Mär 2007) | 1 line
fixed #9355
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18 years ago
0220349ca4
Oops, this should be case insensitive
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18 years ago
ec028bd1b2
Change this to a single dp function to make oej happy.
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18 years ago
cd3238d543
Rename the new dialplan functions to match the variable name
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18 years ago
fa97f6c381
The AUDIORTPQOS and VIDEORTPQOS variables are not fully functional in some
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because they get set in sip_hangup. So, there are common situations where
the variables will not be available in the dialplan at all. So, this patch
provides an alternate method for getting to this information by introducing
AUDIORTPQOS and VIDEORTPQOS dialplan functions.
(issue #9370 , patch by Corydon76, with some testing by blitzrage)
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18 years ago
980b0bc785
* mISDN >= 1.2 provides a dsp pipeline for i.e. echo cancellation modules, make chan_misdn use it.
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* add a check for linux/mISDNdsp.h to configure.ac and update the autogenerated files: 'configure', 'autoconfig.h.in'
(the 'configure' script was not in sync with the latest configure.ac, so the diff is a bit bigger than expected).
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18 years ago
3b6f376425
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r59194 | file | 2007-03-23 21:35:49 -0400 (Fri, 23 Mar 2007) | 2 lines
Only try to handle a response if it has a response code. (ASA-2007-011)
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18 years ago
cf58b2ba46
don't allow string input to overrun the buffer to hold it (ASA-2007-010)
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18 years ago
34f55a9bf6
remove variables that are no longer used (--enable-dev-mode is good, developers should be using it)
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18 years ago
067876afbc
Until we can do media level parsing for sendrecv/etc just use the first value found. This crept up when a phone was offered audio+video and returned an inactive video stream. chan_sip thought the phone said to put the person on hold but that was totally wrong. (issue #9319 reported by benbrown)
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18 years ago
bfeb3bcf18
Copy len variable as well, should fix remaining IAX2 DTMF issues.
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18 years ago
30165b1bf2
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r58849 | crichter | 2007-03-13 12:58:16 +0100 (Di, 13 Mär 2007) | 1 line
added method standard_dec for dialing out on groups, to avoid conflicts, which caused issues with some ISDN providers
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r58850 | crichter | 2007-03-13 13:58:32 +0100 (Di, 13 Mär 2007) | 1 line
fixed the crypt_keys stuff
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r59062 | crichter | 2007-03-20 10:18:06 +0100 (Di, 20 Mär 2007) | 1 line
avoid sending a disconnect when we already received one.
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r59063 | crichter | 2007-03-20 10:23:22 +0100 (Di, 20 Mär 2007) | 1 line
modified a loglevel
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18 years ago
156c178143
Issue #9313 , Asterisk crash on SIP return code 0 (reported by qwerty1979) (ASA-2007-011)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59037 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
6548086af0
Don't assume that the pvt structure will still exist after calling schedule_delivery as it may not. (issue #9278 reported by fmachado)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58923 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
4fc7d081d3
Some people like to put "limitonpeer" instead of "limitonpeers" in their
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configuration. While we're at it, support "limitonpeerz" and
"limitonpeerssssss". (inspired by issue #9172 )
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58906 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
0eda92609d
Merged revisions 58847 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r58847 | oej | 2007-03-13 12:45:52 +0100 (Tue, 13 Mar 2007) | 2 lines
Issue #9229 - No port in request URI on register to non default SIP ports (neelakantan)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58848 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
f546453573
Don't hangup the call on OK or errors on MESSAGE and INFO
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inside of a dialog (like video update requests).
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58845 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
83bb6718fc
Issue #9251 - Clear From URI from user attributes (tgrman)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58843 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
5923e48d7f
Merged revisions 57034,57523,57753,58558 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r57034 | crichter | 2007-02-28 17:09:27 +0100 (Mi, 28 Feb 2007) | 1 line
fixed bugs.digium.com bugs: #9157 and bugs.beronet.com bugs: #302 , #303 , #304
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r57523 | crichter | 2007-03-02 19:32:51 +0100 (Fr, 02 Mar 2007) | 1 line
fixed typo
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r57753 | crichter | 2007-03-04 11:39:50 +0100 (So, 04 Mar 2007) | 1 line
fixed another place where the out_cause was hardcoded to 16
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r58558 | crichter | 2007-03-09 15:43:58 +0100 (Fr, 09 Mar 2007) | 1 line
we can free channel 31 as well, since we can occupy it
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58826 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
160cfaf1c6
added UU transceiving and corect handling for rdnis
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58825 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
fa866efb5c
Add matchexterniplocally setting which only substitutes your externip/externhost setting if it matches the localnet setting. I know of at least two people who need opposite settings, so I made it an option! (issue #8821 reported by kokoskarokoska)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58779 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
8ae4e264d1
Fix a few more places in chan_iax2 where the ast_frame used for receiving a
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frame was not properly initialized.
- Interpolating a frame when the jitterbuffer is in use
- decrypting a frame when IAX2 encryption is on
- frames in an IAX2 trunk
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58705 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
b9124234e9
Merged revisions 58579 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r58579 | file | 2007-03-09 15:46:43 -0500 (Fri, 09 Mar 2007) | 2 lines
If we are unable to lookup the host in a c line we have to abort, otherwise the previous data is gone and we will (potentially) have no data when all is said and done.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58584 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
2f54661cf8
Add a missing break statement so that handling the above event does not
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incorrectly destroy the channel. (issue #9242 , andrew)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58510 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
5603ca69eb
If we receive ZT_EVENT_REMOVED, destroy the specified channel.
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(issue #7256 , tzafrir)
Also, update the configure script to make sure that we don't try to build
chan_zap if the installed version of zaptel does not include ZT_EVENT_REMOVED.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58320 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
f34c30860e
(This bug was reported to me by Kinsey Moore)
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Merged revisions 58242 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r58242 | russell | 2007-03-07 12:17:07 -0600 (Wed, 07 Mar 2007) | 7 lines
Fix a problem where the Asterisk channel name could be that of the wrong IAX2
user for a call. This is because the first step of choosing this name is to
look for an IAX2 peer that happens to have the same IP/port number that this
call is coming from and assuming that is it. However, this is not always
correct. So, I have made it change this name after authentication happens
since at that point, we have an exact match.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58243 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
240355d78b
Ensure we have (or should have) at least one matching codec before attempting early bridge SDP seeding. (issue #9221 reported by marcelbarbulescu)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58240 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
81ef9a4f18
Merged revisions 58115 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r58115 | murf | 2007-03-06 15:52:52 -0700 (Tue, 06 Mar 2007) | 1 line
Fix for 9220: Eyebeam cannot renew subscriptions for presence info. Reason: re-SUBSCRIBE requests don't include Accept headers, which the rfc says are optional (to put it tersely), (it uses MAY), and luckily, the sip_pvt struct has the format info stored, so we simply leave it if the format is set, and the accept header null.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58121 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
275abf4e08
Merged revisions 58052 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r58052 | oej | 2007-03-06 21:33:21 +0100 (Tue, 06 Mar 2007) | 2 lines
Change error message to proper message
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58053 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
905238839b
Return an error of transmit_response is called without a session.
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(issue #9002 )
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58023 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
94c9dcda60
Since chan_iax2 does not support reception of DTMF with duration ensure that it is set to 0 on the frame. (issue #8521 reported by gdhgdh)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@57914 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
53b9bc89c0
Merged revisions 57475 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r57475 | file | 2007-03-02 12:02:46 -0500 (Fri, 02 Mar 2007) | 2 lines
If a SIP message comes in and goes to a method handler that requires additional values that may not be present then send back an error.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@57477 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
219f4b4913
Merged revisions 57317 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r57317 | file | 2007-03-01 17:19:32 -0500 (Thu, 01 Mar 2007) | 2 lines
Don't even attempt to optimize things when a proxy channel is involved. It will just explode in weird and unexplaineable ways. (issue #9175 reported by clegall_proformatique)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@57318 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
43f841cadf
Merged revisions 57092 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r57092 | file | 2007-02-28 13:55:45 -0500 (Wed, 28 Feb 2007) | 2 lines
Fix a few more issues with the agent logoff CLI command. (issue #9123 reported by arbrandes)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@57093 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
83ce3720bc
Restore the behavior of Asterisk 1.2 where if a device was not specified in
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alsa.conf, then we just use the system default, instead of creating our own
default of hw:0,0. (issue #9139 )
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@56888 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
dd6b58edcd
Fix a crash in my last change to iax2_indicate(). (issue #9150 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@56847 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
268fc8a9cc
Do more complete locking of the chan_iax2_pvt struct in the indicate callback.
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(Problem brought up by Ben Smithurst on the asterisk-dev list)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@56785 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
a9e4e0eefa
Make sure to set a speeddials parent on creation.
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Don't crash if hold is pressed when no call is active.
Don't return in places that we shouldn't..
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@56569 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
76a643978e
Merged revisions 56406 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r56406 | russell | 2007-02-23 14:17:56 -0600 (Fri, 23 Feb 2007) | 4 lines
Don't destroy mutexes before unregistering all of the entry points from the core.
Also, fix a potential memory leak from not destroying the locks for all of the
possible call numbers (about 32k of them).
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@56407 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
4565c1483c
Merged revisions 56230 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r56230 | file | 2007-02-22 13:44:24 -0500 (Thu, 22 Feb 2007) | 2 lines
Only change the original or clone channel if it's the channel behind the proxy channel, not if it's just a regular bridged channel.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@56231 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
d67ef52159
Move message from verbose to debug
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@56125 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
9a7d1cc182
Restructure a little bit of code to reduce nesting. There is no functionality
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change here.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@56055 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
71921a8329
Merged revisions 56010 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r56010 | russell | 2007-02-21 18:53:25 -0600 (Wed, 21 Feb 2007) | 3 lines
If we receive a frame that is not in any of the negotiated formats, then drop
it. (potentially issue #8781 and SPD-12)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@56011 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
b937d7e8bc
Fix locking issue, and accept "transport-accept" as a valid accept message.
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This should solve issues 8970 and 8503.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@55954 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
3d95841eb3
Add a flag that indicates whether a SIP dialog is an outgoing call or not. SIP_OUTGOING originally did it but it was repurposed to the direction of the last transaction, which can cause update_call_counter to falsely decrease the wrong counters. (please don't hurt me oej) (issue #8943 reported by mdu113)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@55914 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
c0f5102378
Issue #8848 - Turn off lamp more quickly after transfer (decrement inuse early on transferer's call leg)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@55834 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
67421802de
Fix segfault when buddy couldn't be found.
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Issue 7764, patch by sailer
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@55799 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
30d30fef76
Return behavior I removed. I did not remember that you could just add a localnet entry to make it work.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@55717 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
49749e44a2
Don't test our own address against the localnet settings. At least one person has had issues as a result of this from #7051 so I'm reversing it. (issue #8821 reported by kokoskarokoska)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@55688 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
4f5689f95f
Merged revisions 55669 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r55669 | file | 2007-02-20 17:39:14 -0500 (Tue, 20 Feb 2007) | 2 lines
Defer clearing callback information if channels are up until they are hung up. This ensures the hangup process goes smoothly and no channels get hung in limbo. (issue #8088 reported by kebl0155)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@55670 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
e0b88aba12
No need to cast nor free with strdupa (thanks file)
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55555!
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@55555 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
c967342095
Changed iax2 process thread to detached to correct memory leak due to left over thread context on thread exit.
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Modified module unload process to avoid deadlocks on pthread cancels
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@55397 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
0ade592435
Fix an issue where callerid would not be displayed on some phones.
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Issue 8995, initial patch and research done by wedhorn
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@55217 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
7a4bed883e
Merged revisions 55073 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r55073 | file | 2007-02-16 20:09:50 -0500 (Fri, 16 Feb 2007) | 2 lines
Allow chan_sip to handle attended transfers from a SIP phone that is sitting behind chan_agent. Yes folks, all it took was one line of code. (issue #8784 reported by pzieba)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@55086 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
b0df14b32c
Merged revisions 54999 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r54999 | file | 2007-02-16 17:13:45 -0500 (Fri, 16 Feb 2007) | 2 lines
Do not send indications through ast_indicate in chan_agent but instead go directly to the technology. This way when indications are emulated they happen on the Agent channel and do not screw up formats on the channels. (issue #8439 reported by punkgode)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@55002 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
3ca445e34c
Issue #7541 - Handle multipart attachments to SIP messages - even if boundary is quoted.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@54787 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
91e22d1df2
Merged revisions 54373 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r54373 | mattf | 2007-02-14 10:25:49 -0600 (Wed, 14 Feb 2007) | 2 lines
When handling glare on a PRI, move the requested channel rather than hang up the old one. Fix for 8957 and 9011.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@54375 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
7bcf1d913a
Remove a couple of leftover debug messages
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@54235 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
b100b69703
If we fail to create the SIP socket, then return -1 from reload_config() so
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that load_module() will return AST_MODULE_LOAD_DECLINE. Otherwise, the console
will just get spammed with error messages every time chan_sip tries to send a
message.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@54204 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
f5590445cc
Fix VLDTMF reception
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53881 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
93fcd4a354
Change some text to properly state "On Hold", which was already done in trunk.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53818 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
7ee02f585d
Merge team/russell/sla_rewrite
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This is a completely new implementation of the SLA functionality introduced in
Asterisk 1.4. It is now functional and ready for testing. However, I will be
adding some additional features over the next week, as well.
For information on how to set this up, see configs/sla.conf.sample
and doc/sla.txt.
In addition to the changes in app_meetme.c for the SLA implementation itself,
this merge brings in various other changes:
chan_sip:
- Add the ability to indicate HOLD state in NOTIFY messages.
- Queue HOLD and UNHOLD control frames even if the channel is not bridged to
another channel.
linkedlists.h:
- Add support for rwlock based linked lists.
dial.c:
- Add the ability to run ast_dial_start() without a reference channel to
inherit information from.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53810 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
ff17d402de
another dependency
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53781 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
2f6312a37e
Merged revisions 53357 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r53357 | file | 2007-02-07 10:38:48 -0500 (Wed, 07 Feb 2007) | 2 lines
Fix a few potential memory leaks with realtime users and peers. (issue #8999 reported by bsmithurst)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53358 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
1c90803367
Merged revisions 52843 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r52843 | crichter | 2007-01-30 15:38:08 +0100 (Di, 30 Jan 2007) | 1 line
fixed some possible segfaults. also fixed an very important bug which occurs on high load (when calls are very fast generated)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53324 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
e7a0e86756
Formatting
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53222 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
8e07358edf
Add some comments on queue system behaviour and how it affects the
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SIP channel
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53143 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
910898b7be
Make SIPDtmfMode application work with recent capability changes, and also fix an RTP stack issue when the auto option was used. (issue #8972 reported by mdu113)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53138 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
10ad9bb6d0
Remove quote from H.323 vendor string because due to compatibilities with CS1000 reported at www.voip-info.org
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53131 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
90a4b844a9
Disable the direct p2p RTP call setup in SIP. You can enable it in sip.conf, but it is now
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considered experimental until we solve the AST_CONTROL_ANSWER with payload and videocaps
stuff.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53109 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
57fe6882ac
Merged revisions 53103 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r53103 | file | 2007-02-01 16:21:56 -0600 (Thu, 01 Feb 2007) | 2 lines
Copy noncodeccapability over to the joint variable so that telephone-event will get transmitted in the sent INVITE.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53104 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
09844a7f1a
Merged revisions 53095 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r53095 | file | 2007-02-01 15:47:11 -0600 (Thu, 01 Feb 2007) | 2 lines
Don't negotiate RFC2833 when not configured to do so. (issue #8799 reported by mdu113)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53097 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
97efd0be22
- Clean INC_COUNT flag when we decrement call counter
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- If it's still set at time of dialog destruction, make sure we decrement the device call counter properly
before we destroy the dialog
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53085 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
6bb6bba6a3
Cleaning up the devicestate callback function
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53079 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
e86275c11c
Fix silly logic. We really want to write UDPTL frames out when the call is up.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53064 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
60ad44b8e9
we update the name on any first reply of our setup
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19 years ago
6fd3418e40
chan_h323 is very stable, so let it built by default
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53057 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
9aab046002
Merged revisions 53045 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r53045 | russell | 2007-01-31 15:25:11 -0600 (Wed, 31 Jan 2007) | 3 lines
Fix a bunch of places where pthread_attr_init() was called, but
pthread_attr_destroy() was not.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53046 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
29b7393d84
Only set the DTMF flag on the rtp structure if the DTMF mode is actually
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RFC2833, not just that it is not INFO. This makes it get set for inband DTMF
as well, which is not valid.
(issue #8936 )
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@52952 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
362abb9deb
Drop the deprecated show commands since the original ones were changed back. (issue #8937 reported by PCadach)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@52856 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
a175571e39
Revert reprecation of h.323 gk cycle command from pre-1.4 version instead of duplicated h323 cycle gk
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@52809 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
a46e6d74df
Merged revisions 52762 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r52762 | russell | 2007-01-29 18:15:06 -0600 (Mon, 29 Jan 2007) | 5 lines
Fix the extraction of the timestamp from video frames. It was using the
mapping for a mini-frame instead of a video-frame, which caused it to
get invalid data.
(issue #8795 , mihai)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@52763 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
6b462c95c9
Merged revisions 52360 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r52360 | file | 2007-01-26 19:03:23 -0500 (Fri, 26 Jan 2007) | 2 lines
Make the last context entry read in the dominant one. (issue #8918 reported by pj)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@52370 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
ed48c69f06
Drop out variables I accidentally put in.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@52210 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
6b08afd05d
Decrement onHold count if we are hung up on and still on hold. (issue #8909 reported by alexh42)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@52208 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
5ebd1ecf63
Fix changing channel formats when joint capability changes and there are no audio formats... I didn't break it originally! (issue #8535 reported by ivoc)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@52016 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
9352eda642
added fix from #8899
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51989 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
6d2afb98b4
Merged revisions 51966 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r51966 | crichter | 2007-01-24 11:48:09 +0100 (Mi, 24 Jan 2007) | 1 line
fixed the busy problem (dialstatus was not busy when we called a busy extension)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51979 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
a207a31a97
Show capabilities *and* preference in general settings in "sip show settings"
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(reported by Clona/Telio - Thanks!)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51931 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
8f7ddbef0d
Update channel drivers to use module referencing so that unloading them while in use will not result in crashes. (issue #8897 reported by junky)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51788 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
e65aeb27b5
Merged revisions 50495,50506 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r50495 | crichter | 2007-01-11 14:27:52 +0100 (Do, 11 Jan 2007) | 6 lines
* more additions to make the RESTART message work
* added fix for misdn_call to allow SETUPs with empty
extensions, replaced the strtok_r functions with strsep for that
(inspired by Sandro Cappellazzo, thanks)
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r50506 | crichter | 2007-01-11 15:45:38 +0100 (Do, 11 Jan 2007) | 1 line
when we get L2 UP, the L1 is UP definitely too, so we set the L1 state up as well.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51649 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
19fdc9b14e
manually merged r49922 and r50335, because of conflicts. this commint includes addition of the ISDN RESTART Message
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51648 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
964d2d2e42
Do not abort Asterisk startup if h323 configuration file not found (reported by mithraen)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51615 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
5a3acb0511
Only change audio formats on the channel if we have an audio format to change to. (issue #8535 reported by ivoc)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51558 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
0e37d936da
Fix VLDTMF support in chan_gtalk. AST_FRAME_DTMF and AST_FRAME_DTMF_END are
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actually the same thing. So, a digit would have been interpreted incorrectly
here. Since the channel driver will always have the begin and end callbacks
called for a digit, only support the button-down and button-up messages.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51328 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
33235b40d6
Merge the changes from the /team/group/vldtmf_fixup branch.
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The main bug being addressed here is a problem introduced when two SIP
channels using SIP INFO dtmf have their media directly bridged. So, when a
DTMF END frame comes into Asterisk from an incoming INFO message, Asterisk
would try to emulate a digit of some length by first sending a DTMF BEGIN
frame and sending a DTMF END later timed off of incoming audio. However,
since there was no audio coming in, the DTMF_END was never generated. This
caused DTMF based features to no longer work.
To fix this, the core now knows when a channel doesn't care about DTMF BEGIN
frames (such as a SIP channel sending INFO dtmf). If this is the case, then
Asterisk will not emulate a digit of some length, and will instead just pass
through the single DTMF END event.
Channel drivers also now get passed the length of the digit to their digit_end
callback. This improves SIP INFO support even further by enabling us to put
the real digit duration in the INFO message instead of a hard coded 250ms.
Also, for an incoming INFO message, the duration is read from the frame and
passed into the core instead of just getting ignored.
(issue #8597 , maybe others...)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51311 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
b62be97f7a
chan_zap compiles without libpri after committing 7877 patch
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51274 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
a4317ce870
Merged revisions 51271 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r51271 | dhubbard | 2007-01-18 17:47:10 -0600 (Thu, 18 Jan 2007) | 3 lines
issue 7877: chan_zap module reload does not use default/initialized values on subsequent loads. Reset configuration variables to default values prior to parsing configuration file.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51272 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
1e3557c636
Copy MOH settings when calling a peer so that if they put someone on hold or get put on hold themselves they get the right music class. (issue #8840 reported by mdu113)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51243 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
2b2f8d22f7
Instead of dividing the offset by 2 directly, make it more clear that the
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offset is being scaled by the size of the elements in the buffer.
(Inspired by a discussing on the asterisk-dev list about this code)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51204 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
4244459e31
Merged revisions 51197 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r51197 | russell | 2007-01-17 15:17:21 -0600 (Wed, 17 Jan 2007) | 3 lines
Move the check for a failure of ast_channel_alloc() to before locking the
pvt structure again. Otherwise, on a failure, this will cause a deadlock.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51198 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
f986c4f2c3
Move rescheduling of lagrq/pings into the scheduler callback.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51172 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
531d008066
Merged revisions 51085 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r51085 | file | 2007-01-16 00:53:31 -0500 (Tue, 16 Jan 2007) | 2 lines
Add none as a valid callgroup/pickupgroup option. I consider it a bug that it would inherit it all the way down and not have any way to reset it to nothing - so that's why it is in 1.2. (issue #8296 reported by gkloepfer)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51087 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
915f9315e1
Remove check for channel state as it can definitely be something other then ring, and also clean up the code a bit. This should solve the parking issues and maybe some attended transfer issues people have been seeing.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@50468 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
240ca25bea
Add support to see whether NAT was detected (yay symmetric RTP) and also add a check in chan_sip so that if NAT has been detected and the reinvite behind nat option has been turned off, then just do partial bridge. (issue #8655 reported by mnicholson)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@50466 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
9aca2b2a54
Fix chan_sip not working issue. Let's not prematurely return 0. (issue #8783 reported by st41ker)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@50377 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
1a33c38a15
- handle re-invites properly in sip_hangup()
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- Add some invitestate status changes just to be sure
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@50124 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
3394598f93
Issue #8677 - Handle failure of T.38 re-invite
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This is not a fix, but adding an error message to tell the admin that
we have a bad configuration. We should not send T.38 re-invites to devices
that can't handle it (with the current architecture where you have to
hard-code t.38 support per device).
To really fix this, we need to figure out a way to tell the incoming
call that the re-invite failed, so we can signal failure on that
end and go back to the original call.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@50006 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
0f96f73768
Issue #8524 , support multiple via header values (tardieu)
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Thanks!
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
be32fad9b8
We only need one forward declaration
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19 years ago
484add6554
Issue 8735: Terminate state when extension is unavailable for subscription
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49925 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
0a51642588
Merged revisions 49889 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r49889 | file | 2007-01-08 00:10:07 -0500 (Mon, 08 Jan 2007) | 2 lines
Ensure we use the default refresh value of 60 if the remote server does not send one. (issue #8746 reported by maethor)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49890 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
dcbf36432e
Second condition was a subset of the first, so hold was never decremented, thus hint stayed stuck (Issue 8747)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49831 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
9803b9a357
Make codec_zap and chan_zap also depend on zaptel.
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This fixes an issue (8727) with zaptel being in a different directory, using --with-zaptel.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49705 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
444adcb477
reduce stack consumption for AMI and AMI/HTTP requests by nearly 20K in most cases
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49676 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
fb010e49aa
Merged revisions 49635 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r49635 | kpfleming | 2007-01-05 10:56:40 -0600 (Fri, 05 Jan 2007) | 2 lines
ensure that threads which are supposed to be detached (because we aren't going to wait on them) are created properly
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49636 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
ab4da71374
revert the dynamic_list insertion change... that was not the right thing to do
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49600 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
aed1467281
create the IAX2 processing threads as background threads so they will use smaller stacks
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when we create a dynamic thread, put it on the dynamic_list right away so we don't lose track of it
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49581 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
62b8b03294
It's possible for the iax2 pvt to disappear, so if it has... don't bother looking for dpentries.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49568 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
24bb7d865e
don't mark these allocations as 'cache' allocations when caching has been disabled
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49536 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
6b1817c0d1
if we're going to decrement the frame count when we free a frame, we should inrement it when we create one :-)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49523 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
497dd42772
only do IAX2 frame caching for voice and video frames
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49465 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
f7a47b38eb
don't define this type either if LOW_MEMORY is enabled
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19 years ago
74258ab85e
don't do frame caching in LOW_MEMORY mode
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49458 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
fb52698667
Merged revisions 48319,48321,48467,48552,48576,49135,49303 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r48319 | crichter | 2006-12-06 15:35:25 +0100 (Mi, 06 Dez 2006) | 1 line
changed a few debugs to higher debug levels
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r48321 | crichter | 2006-12-06 16:48:45 +0100 (Mi, 06 Dez 2006) | 1 line
added the export and import of the MISDN_ADDRESS_COMPLETE Variable to inidcate wether the extension is already completely dialed or if there might come additional digits by information elements. also added some docs for that.
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r48467 | crichter | 2006-12-14 14:03:49 +0100 (Do, 14 Dez 2006) | 1 line
removed FIXUP state. added check for channel allocation conflict when we create a setup while the other site creates a setup on the same channel, besides the check we resolve this conflict.
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r48552 | crichter | 2006-12-18 11:19:39 +0100 (Mo, 18 Dez 2006) | 1 line
when our PTP Partner sends us a SETUP with a preselected channel we just accept it, even when we're NT. added some checks for segfaults.
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r48576 | crichter | 2006-12-19 14:08:51 +0100 (Di, 19 Dez 2006) | 1 line
when we reject a channel, because it's in use already, we shouldn't process the setup anymore. made the channel allocation a bit easier and more understandable, removed a few unused lines
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r49135 | crichter | 2007-01-02 11:07:22 +0100 (Di, 02 Jan 2007) | 1 line
added check for channel ranges in the set/empty channel functions. set pmp_l1_check default to no. added misdn restart pid cli command. added cleaning of channel when we send a RELEASE_COMPLETE.
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r49303 | crichter | 2007-01-03 09:24:00 +0100 (Mi, 03 Jan 2007) | 9 lines
* Added check for bridging in misdn_call to avoid setting echocancellation
when 2 mISDN channels are involved and when bridging is set. That lead
to a kernel panic before under different situations, because we switched
about 2 times between hardware bridging and echocancelation
* readded MISDN_URATE variable which got lost before, this should make app_v110
work again
* fixed typo
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49313 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
40707f097a
Check pvt structure presence before passing to send_command. This gets rid of the irritating message about a packet without pvt structure. This happens because the scheduled item is getting cancelled at almost the exact moment it is getting executed.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49259 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
5edb7fa173
Small cleanup of add_t38sdp - it's always enabled at that point in the code
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49212 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
95758b4920
remove comment that is unrelated to this function
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49165 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
f195c5d622
check specifically for VLDTMF and transcoding support in the system's Zaptel installation, and make only the modules that need those features dependent on them (this will allow building the other Zaptel-using parts of Asterisk against older versions of Zaptel or those on other platforms that haven't caught up yet to the Linux version)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49102 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
7db2ca152c
remove incomplete implementation of dnsmgr. Let's fix this in trunk.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49096 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
05716471b5
Use asprintf to build the channel names instead of custom function. I believe the custom function is doing some things that are not portable across all implementations. (issue #8570 reported by hterag & issue #8692 reported by nicolasg)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49070 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
37e5a39d76
Initialize the packet queue in load_module instead of just declaring the list with the default value. (issue #8695 reported by ssokol)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49063 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
1e9c141c2d
Only include acl.h and lock.h once
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48997 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
4f157be79e
Only set rfc2833compensate flag once (handle_common_options)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48995 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
f1f8bbaabe
- Remove checking for T38 options twice. Keeping them in handle_common_options
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48993 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
0f5aae9688
make the option actually match the documentation
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48988 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
b2c8abbc6d
allow 'show memory' and 'show memory summary' to distinguish memory allocations that were done for caching purposes, so they don't look like memory leaks
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48987 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
d2b7e8b247
Be a bit more politically correct
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48985 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
bfe4bb0f1e
Issue #8575 - Buggy cisco MWI support.
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Normally we try not to change our software for bugs in other devices. But in
this case, the Cisco phones are so widespread so we try to implement a fix while
waiting for a bugfix from Cisco.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48982 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
e25756dfac
- Make sure handle_common_options return 1 when we found a common option
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- Move uncommon (only global) option away from handle_common_options
Reported by rizzo. Thanks!
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48980 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
4ea530f2dc
Issue 8599 (rizzo) Change invitestate before re-sending invite with auth.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48977 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
8a42650605
Fix bogus content-length in t38 sdp. (rizzo, #8600 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48975 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
9cc04e026d
Add an API call that initializes an RTP structure. We need this because chan_sip is cheeky and uses a temporary RTP structure for codec purposes, and the API calls that are used rely on the lock. (Pointed out on asterisk-dev by Andy Wang)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48964 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
a1eee8d2d9
Fix a typo in an error message that indicated that the MGCP channel type could
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not be registered, instead of the correct type, OSS.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48948 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
53115ff688
Merged revisions 48943 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r48943 | russell | 2006-12-24 02:23:07 -0500 (Sun, 24 Dec 2006) | 3 lines
Check for the proper return value on an error in a call to mmap().
This was reported by Andy Wang on the asterisk-dev list. Thanks!
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48944 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
1208869c00
Merged revisions 48939 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r48939 | russell | 2006-12-24 01:47:29 -0500 (Sun, 24 Dec 2006) | 3 lines
Remove a couple of misplaced dots in log messages. This was reported by
Andrea Spadaccini on the asterisk-dev mailing list.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48940 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
66783cb46f
Note to self: Run make before committing...
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48888 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
ef2f05a518
Fix for issue 7774 - patch by alamantia
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48870 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
2703d88576
suppress compiler warnings in this module until it can be improved
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48586 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
11e61ea68a
Put thread into proper list if we abort handling due to an error, and also hold the lock while putting it back into the proper idle list so we don't prematurely get a signal. (issue #8604 reported by arkadia)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48564 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
a0c21d30ac
Hold call structure lock in places where a qualify or peer action can destroy it.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48504 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
195c66b4a1
Lock network retransmission queue in all places that it is used.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48502 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
c1f1cf48ac
Issue #8592 - treat 504 as 503 (imported from 1.2)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48487 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
a8fe77b95b
Update to latest IANA spec
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48481 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
3706ecbbc5
Use a wakeup variable so that we don't wait on IO indefinitely if packets need to be retransmitted.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48478 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
f6cf1458ad
Merged revisions 48371 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r48371 | murf | 2006-12-09 19:14:13 -0700 (Sat, 09 Dec 2006) | 1 line
This version applies the patch suggested by stevens in bug 7836 (make inbound channel RINGING state consistent with other channels).
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48372 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Kevin P. Fleming
Olle Johansson
Joshua Colp
Christian Richter
Russell Bryant
Steve Murphy
Jason Parker
Matthew Fredrickson
Tilghman Lesher
Dwayne M. Hubbard
Nadi Sarrar
Doug Bailey
Paul Cadach