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r66474 | oej | 2007-05-29 21:02:04 +0200 (Tue, 29 May 2007) | 2 lines
Don't issue hangup on hangup on hangup on hangup (for jcmoore)
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2007-05-29 19:17:49 +00:00
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r66414 | oej | 2007-05-29 18:07:44 +0200 (Tue, 29 May 2007) | 2 lines
Don't reset hangupcause if we already have one
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2007-05-29 16:19:53 +00:00
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r66404 | oej | 2007-05-29 18:02:50 +0200 (Tue, 29 May 2007) | 2 lines
Tracking down hanging channels, killing them one by one. Issue #9235 and related
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2007-05-29 16:17:17 +00:00
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r66363 | oej | 2007-05-29 11:41:40 +0200 (Tue, 29 May 2007) | 10 lines
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r66349 | oej | 2007-05-29 09:53:14 +0200 (Tue, 29 May 2007) | 2 lines
Issue #9802 - Change inuse counter on CANCEL
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2007-05-29 10:02:31 +00:00
4d03b4f268
Don't try to unregister a peer using the sip unregister CLI command if they are not registered. (issue #9811 reported by eliel)
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2007-05-28 23:28:52 +00:00
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Due to the way stringfields work the value of the url pointer will always be non-NULL so we have to use ast_strlen_zero to make sure it is not empty. (issue #9821 reported by pj)
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2007-05-28 23:24:04 +00:00
f371a4c756
more minor fixes
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2007-05-25 15:07:26 +00:00
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r66157 | kpfleming | 2007-05-25 10:28:46 -0400 (Fri, 25 May 2007) | 3 lines
handle the GNUTLS library properly in the configure script and build system
don't build in OSP support unless we have found and are allowed to use SSL support
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2007-05-25 14:37:55 +00:00
4573b36af1
use the OpenSSL AES implementation if it's available (unless configured not to)
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2007-05-24 22:07:50 +00:00
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r65965 | kpfleming | 2007-05-24 14:24:55 -0400 (Thu, 24 May 2007) | 2 lines
don't use uninitialized variables
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r65966 | kpfleming | 2007-05-24 14:25:21 -0400 (Thu, 24 May 2007) | 2 lines
don't reference GnuTLS headers and functions unless the configure script found it
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r65967 | kpfleming | 2007-05-24 14:28:48 -0400 (Thu, 24 May 2007) | 2 lines
oops, use #ifdef instead of #if
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2007-05-24 19:05:42 +00:00
4b3a3fb14c
Add a new API call for creating detached threads. Then, go replace all of the
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places in the code where the same block of code for creating detached threads
was replicated. (patch from bbryant)
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2007-05-24 18:30:19 +00:00
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r65901 | oej | 2007-05-24 17:26:10 +0200 (Thu, 24 May 2007) | 2 lines
Issue 7672 - fix by zandbelt - Asterisk core dump since the GnuTLS interface did not support multithreading correctly.
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2007-05-24 15:29:10 +00:00
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r65892 | oej | 2007-05-24 17:20:54 +0200 (Thu, 24 May 2007) | 2 lines
Issue 8193 - NAT issues with gtalk/STUN. Patch by phsultan. Thanks!
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2007-05-24 15:23:04 +00:00
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r65857 | oej | 2007-05-24 17:05:10 +0200 (Thu, 24 May 2007) | 2 lines
Issue 7686, fix by phsultan, NAT issues when calling from gtalk to SIP over nat.
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2007-05-24 15:21:39 +00:00
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r65841 | oej | 2007-05-24 16:48:55 +0200 (Thu, 24 May 2007) | 2 lines
Issue #8536 - Caller ID not set in CDR for jingle
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2007-05-24 14:52:01 +00:00
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r65839 | file | 2007-05-24 10:42:12 -0400 (Thu, 24 May 2007) | 10 lines
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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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2007-05-24 14:43:49 +00:00
9b95428cce
Issue #8409 and accidentally a fix to chan_sip that wasn't supposed to be there
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but is still ok... Sorry. Lack of Tea, really.
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2007-05-24 14:41:43 +00:00
0ec502099f
Yes Virginia, there is a reason why we have stringfields in the sip_pvt structure...
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2007-05-24 11:38:20 +00:00
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r65768 | crichter | 2007-05-24 11:37:32 +0200 (Do, 24 Mai 2007) | 9 lines
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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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2007-05-24 09:51:01 +00:00
d1ba4f90de
- Remove debug variable that was only used in one place
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- convert string handling to the ast_str API
- Convert strdup() to ast_strdup() and check the result
- Minor formatting changes
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2007-05-24 03:28:39 +00:00
04e45cfda3
Add SendURL support
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2007-05-24 02:23:08 +00:00
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r65685 | kpfleming | 2007-05-23 16:59:19 -0400 (Wed, 23 May 2007) | 2 lines
start the delayed PBX when receive voice or video full frames as well, and comment this delayed-PBX activity
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2007-05-23 21:01:18 +00:00
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r65683 | kpfleming | 2007-05-23 16:51:56 -0400 (Wed, 23 May 2007) | 10 lines
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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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r65679 | kpfleming | 2007-05-23 16:30:24 -0400 (Wed, 23 May 2007) | 2 lines
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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r65680 | kpfleming | 2007-05-23 16:35:50 -0400 (Wed, 23 May 2007) | 2 lines
clear the 'delay PBX' flag when we are ready to start the PBX
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r65677 | kpfleming | 2007-05-23 16:07:59 -0400 (Wed, 23 May 2007) | 10 lines
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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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2007-05-23 20:08:59 +00:00
cc016de696
Make sure we get the cause code in the REL
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r65589 | russell | 2007-05-23 08:07:13 -0500 (Wed, 23 May 2007) | 11 lines
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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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r65501 | russell | 2007-05-22 13:40:38 -0500 (Tue, 22 May 2007) | 3 lines
List res_smdi as a dependency for app_voicemail and chan_zap
(Thanks to mnicholson for pointing it out)
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2007-05-22 18:41:56 +00:00
a3e603cd0f
Use ast_strlen_zero where possible. (issue #9774 reported by eliel)
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2007-05-22 15:08:08 +00:00
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r65342 | crichter | 2007-05-22 10:12:20 +0200 (Di, 22 Mai 2007) | 9 lines
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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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2007-05-22 08:21:59 +00:00
a3f9350ec2
Related to issue #9235 btw.
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r65123 | oej | 2007-05-18 20:16:09 +0200 (Fri, 18 May 2007) | 10 lines
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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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2007-05-18 18:18:59 +00:00
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r65076 | oej | 2007-05-18 17:18:13 +0200 (Fri, 18 May 2007) | 13 lines
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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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r65039 | crichter | 2007-05-18 14:40:46 +0200 (Fr, 18 Mai 2007) | 9 lines
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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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2007-05-18 12:43:59 +00:00
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r64904 | crichter | 2007-05-18 10:58:51 +0200 (Fr, 18 Mai 2007) | 9 lines
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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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r64974 | oej | 2007-05-18 12:37:44 +0200 (Fri, 18 May 2007) | 2 lines
Issue 9487 - stop media flows at hangup of call
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2007-05-18 10:41:31 +00:00
ac343d43c8
Makeup, darling.
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r64515 | crichter | 2007-05-16 10:44:51 +0200 (Mi, 16 Mai 2007) | 9 lines
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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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r63534 | crichter | 2007-05-09 15:17:10 +0200 (Mi, 09 Mai 2007) | 17 lines
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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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r62912 | crichter | 2007-05-03 16:36:32 +0200 (Do, 03 Mai 2007) | 17 lines
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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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r59774 | crichter | 2007-04-03 09:20:27 +0200 (Di, 03 Apr 2007) | 17 lines
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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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2007-05-18 09:47:19 +00:00
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r59254 | crichter | 2007-03-27 17:00:10 +0200 (Di, 27 Mär 2007) | 9 lines
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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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2007-05-18 09:42:42 +00:00
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r59064 | crichter | 2007-03-20 14:16:06 +0100 (Di, 20 Mär 2007) | 21 lines
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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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2007-05-18 09:36:16 +00:00
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r58825 | crichter | 2007-03-12 13:43:24 +0100 (Mo, 12 Mär 2007) | 1 line
added UU transceiving and corect handling for rdnis
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r58826 | crichter | 2007-03-12 14:08:06 +0100 (Mo, 12 Mär 2007) | 21 lines
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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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2007-05-18 09:31:27 +00:00
f3ec447a23
Another fix for the support for recordings controlled by INFO-packets
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We still lack a setting to enable/disable this per peer
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2007-05-18 08:49:34 +00:00
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r64754 | file | 2007-05-17 12:10:12 -0400 (Thu, 17 May 2007) | 2 lines
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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2007-05-17 16:11:26 +00:00
d83dcae6b1
Below patches with some re-structuring for trunk
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r64602 | oej | 2007-05-16 12:38:18 +0200 (Wed, 16 May 2007) | 2 lines
Issue #9681 - Handle www-auth on BYE
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2007-05-16 10:58:22 +00:00
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r64578 | oej | 2007-05-16 12:05:47 +0200 (Wed, 16 May 2007) | 2 lines
Final part of issue #9483 - fixing transfer() of sip calls in the dial plan (twilson)
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2007-05-16 10:09:42 +00:00
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r64543 | oej | 2007-05-16 11:12:34 +0200 (Wed, 16 May 2007) | 10 lines
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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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2007-05-16 10:02:06 +00:00
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r64516 | oej | 2007-05-16 10:46:18 +0200 (Wed, 16 May 2007) | 17 lines
Merged following patch with a lot of changes for 1.4
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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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2007-05-16 08:51:39 +00:00
7532b0bc4b
Issue #9304 - Update help text to match functionality. Patch by kshumard with changes by oej
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2007-05-16 07:58:43 +00:00
90bad9d2f5
Issue #6789 - Marquis - Add option to support regexten removal when host becomes unreachable
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2007-05-16 07:35:56 +00:00
9b5454232c
XXX-XXX-XXX appears to be the standard ANSI pointcode format
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2007-05-15 20:45:20 +00:00
79f553a4ab
Only print the SS7 UP once. Not every time we get the test messages on the
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line.
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2007-05-14 22:25:56 +00:00
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r64324 | oej | 2007-05-14 21:26:50 +0200 (Mon, 14 May 2007) | 2 lines
Change -2 to XMIT_ERROR to clarify a bit more
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2007-05-14 19:35:58 +00:00
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r64306 | russell | 2007-05-14 14:13:00 -0500 (Mon, 14 May 2007) | 3 lines
Properly handle AST_CONTROL_PROGRESS by just ignoring it. An unknown indication
will trigger an error and cause sounds to stop, which in this case, is ringing.
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2007-05-14 19:21:31 +00:00
4fbb449444
If no port is specified in the outboundproxy setting then use the standard SIP port. (issue #9665 reported by tootai)
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2007-05-14 18:21:30 +00:00
a76ff39c0e
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r64193 | murf | 2007-05-14 07:58:42 -0600 (Mon, 14 May 2007) | 1 line
As per 9570, worrisome CDR warnings have been removed, that are either not helpful, or not relevant.
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2007-05-14 14:13:45 +00:00
fc057b1890
Improve handling network errors on transmission to hosts that don't reply or are unreachable
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With this code, the call will fail as soon as we get a network error. This may happen on
first xmit or a later one, so the retransmit code handles this too.
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2007-05-13 19:20:36 +00:00
b71e691b3e
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r64114 | file | 2007-05-12 18:27:04 -0400 (Sat, 12 May 2007) | 2 lines
This concludes my final adventure with bitmasks and the onhold flag. Would anyone care for some peanuts?
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2007-05-12 22:28:09 +00:00
38e951cfda
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r64086 | file | 2007-05-12 17:10:45 -0400 (Sat, 12 May 2007) | 2 lines
Tweak hold flags some more. They can be of three states when active: active, inactive, one direction.
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2007-05-12 21:12:18 +00:00
82a30356da
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r64044 | file | 2007-05-12 12:32:15 -0400 (Sat, 12 May 2007) | 2 lines
Ensure the onhold flag is set no matter what when being put on hold.
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2007-05-12 16:33:34 +00:00
60f6a72123
Add/fix support for Redial, Speeddial, and Messages buttons.
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Combined effort by DEA and mvanbaak.
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2007-05-11 22:52:36 +00:00
2768946cf5
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r63830 | qwell | 2007-05-10 18:15:37 -0500 (Thu, 10 May 2007) | 12 lines
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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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2007-05-10 23:16:33 +00:00
aa320037d2
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r63749 | oej | 2007-05-10 22:46:41 +0200 (Thu, 10 May 2007) | 12 lines
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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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2007-05-10 20:51:59 +00:00
e2ca869abd
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r63654 | mattf | 2007-05-09 12:25:21 -0500 (Wed, 09 May 2007) | 10 lines
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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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2007-05-09 17:35:54 +00:00
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r63611 | file | 2007-05-09 12:54:56 -0400 (Wed, 09 May 2007) | 10 lines
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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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2007-05-09 16:56:45 +00:00
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r63532 | oej | 2007-05-09 15:04:14 +0200 (Wed, 09 May 2007) | 2 lines
Don't retransmit 200 OK's on ignore status. (Reported on asterisk-users)
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2007-05-09 13:07:44 +00:00
314c874d7d
I noted this on the dev list but got no response, so I just did it myself.
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Lock the call features when being used in chan_sip.
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2007-05-08 16:41:35 +00:00
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Adding external referenses for doxygen
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See http://www.asterisk.org/doxygen/trunk/extref.html
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2007-05-07 18:25:56 +00:00
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Adding external reference
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2007-05-07 18:18:43 +00:00
238bcd6c45
Doxyfication... There's a shortage of comments in this file...
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2007-05-07 18:17:18 +00:00
28f4727e75
Lock iax2 pvt structure when passing off to the AMI function, and make sure it exists. (issue #9674 reported by arabe)
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2007-05-06 20:09:18 +00:00
d326d84ae0
- Adding some missing spaces
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- Correcting error messages
- Disabling code that doesn't do anything
- Making sure we always respond to this request, happily
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2007-05-05 08:05:38 +00:00
02337303ef
a small upgrade to the coding standard, and an update to the code that triggered the upgrade.
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2007-05-04 17:49:20 +00:00
3ee0077f04
Added a small bit of code to support the SNOM 360's Record button. Made the find_feature func in res_features.c public, so I could use it to find the automon dial sequence as configured by the user. When the INFO packet has a Record: header with on/off, the sequence is sent as consecutive DTMF frames on the phone's channel, triggering the automon functionality. The user has to configure the automon in features.conf, and set up his dialplan accordingly.
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2007-05-04 16:37:23 +00:00
1b15d8852d
Add the new ChannelUpdate event to inform manager clients about the PVT ID and some other channel driver data that
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is needed to follow the call through the PBX.
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2007-05-04 13:56:25 +00:00
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r62989 | file | 2007-05-03 13:44:00 -0300 (Thu, 03 May 2007) | 10 lines
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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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2007-05-03 16:45:39 +00:00
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r62692 | tilghman | 2007-05-02 12:43:48 -0500 (Wed, 02 May 2007) | 12 lines
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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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2007-05-02 17:49:36 +00:00
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r62689 | murf | 2007-05-02 11:10:50 -0600 (Wed, 02 May 2007) | 1 line
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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2007-05-02 17:24:03 +00:00
3d409eb793
Update the device state functionality of chan_local such that it will return
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NOT_INUSE or INUSE when Local channels are in use as opposed to just UNKNOWN.
It will still return INVALID if the extension doesn't exist at all.
(issue #8048 , patch from tim_ringenbach)
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2007-05-02 15:46:49 +00:00
e1ec3f917c
Add a small message that we're doing something. On my systems, there's a long
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dead period with a non-responsive CLI after I issue "load chan_sip.so"
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2007-05-02 12:12:02 +00:00
1d51b2e161
More username body parts to fix... If working, this needs to be backported to 1.2, 1.4.
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But first, some serious SIP testing :-)
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2007-05-02 12:00:03 +00:00
8fee67c83b
Handle sip:username;parameter=12345@example.com;parameter=1234 URI's properly
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2007-05-02 09:41:03 +00:00
daefa6a8b4
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r62624 | oej | 2007-05-02 08:15:43 +0200 (Wed, 02 May 2007) | 2 lines
Don't unlock a channel that we already know does not exist (propably isue 8228)
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2007-05-02 09:35:14 +00:00
b419fc1134
Add support for setting the CoS for VLAN traffic (802.1p) in Linux. The
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file doc/qos.tex has been updated to document the new functionality.
(issue #9540 , patch submitted by IgorG)
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2007-04-30 16:16:26 +00:00
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r62419 | russell | 2007-04-30 10:58:28 -0500 (Mon, 30 Apr 2007) | 12 lines
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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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2007-04-30 15:59:27 +00:00
5cb08adc7a
Don't crash when invalid arguments are provided to the CHANNEL() function
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for a SIP channel.
(issue #9619 , reported by jtodd, original patch by Corydon76, committed patch
slightly modified by me)
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2007-04-30 15:37:23 +00:00
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r62331 | russell | 2007-04-29 00:50:37 -0500 (Sun, 29 Apr 2007) | 3 lines
Fix a bug that made the "language" setting in zapata.conf not
functional. (issue #9626 , reported and fixed by sergee)
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2007-04-29 05:51:18 +00:00
7d2102c081
Reformat some of iax2.h and convert comments to doxygen format
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2007-04-28 21:26:00 +00:00
b6b1bf3213
Merge changes from team/russell/events
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This set of changes introduces a new generic event API for use within Asterisk.
I am still working on a way for events to be shared between servers, but this
part is ready and can already be used inside of Asterisk.
This set of changes introduces the first use of the API, as well. I have
restructured the way that MWI (message waiting indication) is handled. It is
now event based instead of polling based. For example, if there are a bunch
of SIP phones subscribed to mailboxes, then chan_sip will not have to
constantly poll the mailboxes for changes. app_voicemail will generate events
when changes occur.
See UPGRADE.txt and CHANGES for some more information on the effects of these
changes from the user perspective. For developer information, see the text in
include/asterisk/event.h.
As always, additional feedback is welcome on the asterisk-dev mailing list.
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r62218 | russell | 2007-04-27 16:10:51 -0500 (Fri, 27 Apr 2007) | 11 lines
Fix a weird problem where when a caller talking to someone sitting behind an
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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2007-04-27 21:11:46 +00:00
240bd841b0
Issue #9545 Autocomplete for "sip unregister" cli command. (eliel) Thanks!
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2007-04-27 14:40:28 +00:00
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r62137 | oej | 2007-04-27 16:04:07 +0200 (Fri, 27 Apr 2007) | 12 lines
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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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r62038 | file | 2007-04-26 12:33:52 -0400 (Thu, 26 Apr 2007) | 10 lines
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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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2007-04-26 16:35:14 +00:00
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r61914 | kpfleming | 2007-04-25 17:29:53 -0500 (Wed, 25 Apr 2007) | 10 lines
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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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2007-04-25 22:34:58 +00:00
dc7514a746
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r61870 | russell | 2007-04-25 16:59:07 -0500 (Wed, 25 Apr 2007) | 10 lines
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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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r61863 | russell | 2007-04-25 16:13:15 -0500 (Wed, 25 Apr 2007) | 10 lines
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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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2007-04-25 21:15:19 +00:00
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r61799 | russell | 2007-04-25 11:22:07 -0500 (Wed, 25 Apr 2007) | 11 lines
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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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2007-04-25 16:23:00 +00:00
f3ab33014a
removed #if 0 block from chan_zap restart_monitor()
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2007-04-24 19:08:28 +00:00
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r61772 | file | 2007-04-24 12:07:02 -0400 (Tue, 24 Apr 2007) | 10 lines
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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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2007-04-24 16:10:10 +00:00
b94378b5e6
Merge changes from team/russell/iax2_osp
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This set of changes adds OSP support to chan_iax2. However, I have modified
the patch a bit from what was submitted. You now use the CHANNEL() function
to get and set the OSP token for IAX2.
(issue #8531 , reported by and original patch by homesick, patch updated by me)
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2007-04-20 21:12:53 +00:00
49af71c100
Use the last line in the SDP, even if it has no CRLF. Remember Jon Postel :-)
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This code exists in 1.2 and 1.4 but was removed from trunk for some unknown reason.
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2007-04-20 08:41:24 +00:00
34469a8707
added CLI 'sip unregister <peer>' for issue 9326. thanks eliel
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2007-04-13 21:23:10 +00:00
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r61648 | file | 2007-04-13 13:19:53 -0400 (Fri, 13 Apr 2007) | 2 lines
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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2007-04-13 17:21:53 +00:00
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r61644 | murf | 2007-04-13 11:01:02 -0600 (Fri, 13 Apr 2007) | 1 line
A fix for chan_oss that resulted from the CDR changes; it helps to use the right info.
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2007-04-13 17:11:53 +00:00
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r61641 | file | 2007-04-13 12:32:03 -0400 (Fri, 13 Apr 2007) | 2 lines
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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2007-04-13 16:35:33 +00:00
c4c2def716
Don't treat a host lookup as failed if sipregs is not in use when doing a realtime lookup. (issue #9255 reported by sergee)
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2007-04-12 19:32:00 +00:00
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r61477 | russell | 2007-04-11 11:05:29 -0500 (Wed, 11 Apr 2007) | 13 lines
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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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r61342 | nadi | 2007-04-11 12:52:28 +0200 (Mi, 11 Apr 2007) | 2 lines
AOCD's are now exported to asterisk channel variables.
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r61372 | nadi | 2007-04-11 15:33:30 +0200 (Mi, 11 Apr 2007) | 2 lines
Ignore facility messages in case we don't have a corresponding channel object.
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r61373 | nadi | 2007-04-11 15:40:26 +0200 (Mi, 11 Apr 2007) | 2 lines
Export AOCD variables on misdn_hangup.
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r61443 | nadi | 2007-04-11 17:39:14 +0200 (Mi, 11 Apr 2007) | 2 lines
Don't export AOCD variables on misdn_hangup anymore, this was mainly a fix for trunk..
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2007-04-11 15:48:54 +00:00
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r61427 | russell | 2007-04-11 10:09:39 -0500 (Wed, 11 Apr 2007) | 14 lines
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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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r61377 | russell | 2007-04-11 09:04:44 -0500 (Wed, 11 Apr 2007) | 13 lines
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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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2007-04-11 14:13:08 +00:00
9fff461080
Remove duplicate prototype declaration. (issue #9517 reported by junky)
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2007-04-11 14:01:53 +00:00
7c9e46507b
Added zapata.conf parameter "cid_rxgain" to allow the user to adjust the gain bump used during CID acquisition.
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2007-04-10 22:02:53 +00:00
28b5fb02bd
updated ast_channel_alloc() call to include the 4 extra args everyone got. Not much info there, as the config file evidently does not allow amaflags, or accountcode settings; and the pvt's exten doesn't sound like what we need in the cdr, either.
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r61183 | nadi | 2007-04-10 14:43:40 +0200 (Di, 10 Apr 2007) | 10 lines
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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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2007-04-10 12:47:57 +00:00
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r60989 | murf | 2007-04-09 12:32:07 -0600 (Mon, 09 Apr 2007) | 1 line
This is a big improvement over the current CDR fixes. It may still need refinement, but this won't have as many folks bothered.
This also adds the mods from 1.4/r.61136;
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r61072 | oej | 2007-04-09 21:58:17 +0200 (Mon, 09 Apr 2007) | 11 lines
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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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2007-04-09 20:01:28 +00:00
4aef0155d6
use "ChannelType" in events to indicate which channel driver that generates the event. This replaces
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"ChannelDriver" and "Channel", previously used to indicate channel driver. ChannelType is more
in line with "core show channeltypes"
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2007-04-09 18:22:43 +00:00
a4bef3bb3a
Make RTP session ID and session version generation random. (issue #9456 reported by tjardick)
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2007-04-09 12:33:49 +00:00
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r60847 | tilghman | 2007-04-08 21:42:48 -0500 (Sun, 08 Apr 2007) | 10 lines
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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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2007-04-09 02:44:37 +00:00
ed75ded048
Add counter for sip show registry CLI command. (issue #9352 reported by junky)
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2007-04-09 00:47:06 +00:00
5bc2aa8ab1
Use the same parameter to the two "Registry" AMI events - ChannelDriver
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r60485 | mattf | 2007-04-06 13:21:52 -0500 (Fri, 06 Apr 2007) | 2 lines
Make sure we check the faxdetect option before doing fax processing
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r60459 | mattf | 2007-04-06 12:32:31 -0500 (Fri, 06 Apr 2007) | 10 lines
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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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r60214 | file | 2007-04-05 08:55:02 -0400 (Thu, 05 Apr 2007) | 10 lines
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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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2007-04-05 12:57:35 +00:00
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r60112 | russell | 2007-04-04 11:49:45 -0500 (Wed, 04 Apr 2007) | 3 lines
Add a Content-Length of 0 to the response built by transmit_response_with_unsupported().
(issue #9454 , reported by makoto, fixed by me)
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2007-04-04 16:50:31 +00:00
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r60088 | russell | 2007-04-04 11:39:04 -0500 (Wed, 04 Apr 2007) | 12 lines
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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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2007-04-04 16:40:01 +00:00
b2ddaaf033
Add support for RTP packetization in chan_jingle and chan_gtalk.
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(issue #9416 , phsultan)
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2007-04-03 22:33:03 +00:00
11ab6db24b
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r59939 | russell | 2007-04-03 14:16:53 -0500 (Tue, 03 Apr 2007) | 12 lines
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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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2007-04-03 19:17:55 +00:00
3f14c8b6fc
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r59936 | russell | 2007-04-03 13:55:57 -0500 (Tue, 03 Apr 2007) | 11 lines
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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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2007-04-03 18:57:52 +00:00
b908f9717a
Remove a duplicate function prototype. (issue #9444 , junky)
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2007-04-03 18:34:14 +00:00
4d1edcd4fd
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r59804 | nadi | 2007-04-03 13:02:46 +0200 (Di, 03 Apr 2007) | 15 lines
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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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2007-04-03 11:15:45 +00:00
592870b6ce
This hashing code is still causing some random crashes on my system, and
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probably others, too. I don't really have time to work on it at the moment,
so I am just going to revert it for now.
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2007-04-02 17:40:26 +00:00
d6e7b047f0
Update to support multiple CIC groups and DPCs per linkset.
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2007-03-30 20:54:42 +00:00
5d4e6a480a
Fix an issue with hashing iax2 pvt structures that caused random crashes on
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systems under heavy load such as IAXtel.
(possibly related to issue #9403 )
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2007-03-29 18:31:35 +00:00
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r59341 | russell | 2007-03-29 11:55:39 -0500 (Thu, 29 Mar 2007) | 8 lines
When the IAX2 read callback gets called, return NULL instead of a "null frame".
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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2007-03-29 16:56:35 +00:00
6626c31f35
added filtering options to 'zap show channels' to implement functionality described in issue 6520
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2007-03-28 00:09:25 +00:00
04f1955781
Use better english. Renegotiate! Repeat after me: renegotiate.
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2007-03-27 20:11:11 +00:00
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r59262 | russell | 2007-03-27 13:17:47 -0500 (Tue, 27 Mar 2007) | 3 lines
Fix the check that ensures that the CHANNEL function's first argument is "rtpqos".
Thanks, Corydon. :)
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2007-03-27 18:18:36 +00:00
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r59259 | russell | 2007-03-27 13:05:46 -0500 (Tue, 27 Mar 2007) | 12 lines
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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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2007-03-27 18:08:59 +00:00
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r59256 | russell | 2007-03-27 11:20:53 -0500 (Tue, 27 Mar 2007) | 4 lines
Convert the RTPQOS function to just be additional parameter of the CHANNEL
function. This way, it will be possible for other RTP based channel drivers
to expose this information in the future.
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2007-03-27 16:25:02 +00:00
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r59227 | tilghman | 2007-03-26 16:37:41 -0500 (Mon, 26 Mar 2007) | 2 lines
Change this to a single dp function to make oej happy.
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2007-03-26 21:44:59 +00:00
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r59209 | russell | 2007-03-26 12:53:07 -0500 (Mon, 26 Mar 2007) | 1 line
Rename the new dialplan functions to match the variable name
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2007-03-26 17:57:50 +00:00
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r59207 | russell | 2007-03-26 12:45:55 -0500 (Mon, 26 Mar 2007) | 7 lines
The AUDIORTPQOS and VIDEORTPQOS variables are not fully functional in some
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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2007-03-26 17:51:27 +00:00
e4a7d1b35f
Fix bug in which parameter type we are passing. This shouldn't be a problem
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since both types are the same underneath.
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2007-03-26 16:48:29 +00:00
df42d78569
Small API related SS7 updates.
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2007-03-26 16:46:07 +00:00
24d8595d00
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r59202 | nadi | 2007-03-26 17:25:53 +0200 (Mo, 26 Mär 2007) | 4 lines
* mISDN >= 1.2 provides a dsp pipeline for i.e. echo cancellation modules, make chan_misdn use it.
* 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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2007-03-26 15:59:56 +00:00
cc22f60f30
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r59195 | file | 2007-03-23 21:39:44 -0400 (Fri, 23 Mar 2007) | 10 lines
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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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2007-03-24 01:42:11 +00:00
e8e9e5e23c
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r59182 | kpfleming | 2007-03-22 16:40:01 -0700 (Thu, 22 Mar 2007) | 2 lines
don't allow string input to overrun the buffer to hold it (ASA-2007-010)
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2007-03-22 23:41:37 +00:00
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r59180 | kpfleming | 2007-03-22 16:34:22 -0700 (Thu, 22 Mar 2007) | 2 lines
remove variables that are no longer used (--enable-dev-mode is good, developers should be using it)
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2007-03-22 23:39:26 +00:00
af9c17025f
Minor tweak. Only queue up an unhold control frame if we are actually on hold. This would have shown itself when a call was initially being setup and the SDP data was being parsed in.
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2007-03-21 03:33:57 +00:00
1d5be2d1c7
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r59081 | file | 2007-03-20 23:25:48 -0400 (Tue, 20 Mar 2007) | 2 lines
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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2007-03-21 03:27:58 +00:00
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r59076 | file | 2007-03-20 16:42:46 -0400 (Tue, 20 Mar 2007) | 2 lines
Copy len variable as well, should fix remaining IAX2 DTMF issues.
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r59037 | oej | 2007-03-18 21:37:06 +0100 (Sun, 18 Mar 2007) | 3 lines
Issue #9313 , Asterisk crash on SIP return code 0 (reported by qwerty1979) (ASA-2007-011)
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r58923 | file | 2007-03-15 15:13:21 -0300 (Thu, 15 Mar 2007) | 2 lines
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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2007-03-15 18:21:03 +00:00
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r58906 | russell | 2007-03-14 14:18:08 -0500 (Wed, 14 Mar 2007) | 4 lines
Some people like to put "limitonpeer" instead of "limitonpeers" in their
configuration. While we're at it, support "limitonpeerz" and
"limitonpeerssssss". (inspired by issue #9172 )
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2007-03-14 19:19:00 +00:00
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r58848 | oej | 2007-03-13 12:49:35 +0100 (Tue, 13 Mar 2007) | 10 lines
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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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2007-03-14 16:59:35 +00:00
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r58845 | oej | 2007-03-13 11:03:03 +0100 (Tue, 13 Mar 2007) | 3 lines
Don't hangup the call on OK or errors on MESSAGE and INFO
inside of a dialog (like video update requests).
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2007-03-13 10:14:13 +00:00
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r58843 | oej | 2007-03-13 10:12:16 +0100 (Tue, 13 Mar 2007) | 2 lines
Issue #9251 - Clear From URI from user attributes (tgrman)
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2007-03-13 09:15:17 +00:00
d008232b27
Change URL to OpenH323 (thanks, Tzafrir!)
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2007-03-12 09:37:13 +00:00
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r58779 | file | 2007-03-11 20:51:16 -0400 (Sun, 11 Mar 2007) | 2 lines
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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2007-03-12 00:54:13 +00:00
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r58705 | russell | 2007-03-10 12:11:11 -0600 (Sat, 10 Mar 2007) | 6 lines
Fix a few more places in chan_iax2 where the ast_frame used for receiving a
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
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r58584 | file | 2007-03-09 15:49:47 -0500 (Fri, 09 Mar 2007) | 10 lines
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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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r58510 | russell | 2007-03-08 16:06:54 -0600 (Thu, 08 Mar 2007) | 3 lines
Add a missing break statement so that handling the above event does not
incorrectly destroy the channel. (issue #9242 , andrew)
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2007-03-08 22:08:28 +00:00
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r58320 | russell | 2007-03-07 19:01:46 -0600 (Wed, 07 Mar 2007) | 6 lines
If we receive ZT_EVENT_REMOVED, destroy the specified channel.
(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.
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2007-03-08 01:06:00 +00:00
f86aee41f7
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r58243 | russell | 2007-03-07 12:19:19 -0600 (Wed, 07 Mar 2007) | 17 lines
(This bug was reported to me by Kinsey Moore)
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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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2007-03-07 18:20:51 +00:00
e7da006562
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r58240 | file | 2007-03-07 12:52:58 -0500 (Wed, 07 Mar 2007) | 2 lines
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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2007-03-07 17:55:11 +00:00
5bb1d16f47
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r58121 | murf | 2007-03-06 16:10:14 -0700 (Tue, 06 Mar 2007) | 9 lines
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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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2007-03-06 23:19:59 +00:00
52750892b0
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r58053 | oej | 2007-03-06 21:37:07 +0100 (Tue, 06 Mar 2007) | 10 lines
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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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2007-03-06 20:45:29 +00:00
d7aa9b6e2d
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r58023 | russell | 2007-03-06 12:01:20 -0600 (Tue, 06 Mar 2007) | 3 lines
Return an error of transmit_response is called without a session.
(issue #9002 )
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2007-03-06 18:02:35 +00:00
b7e47198da
Add zap show version CLI command. This pulls the version/echo canceller in use directly using the ZT_GETVERSION ioctl. (issue #9094 reported by tootai)
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2007-03-05 20:13:51 +00:00
2adf4fa4e0
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r57914 | file | 2007-03-05 14:19:07 -0500 (Mon, 05 Mar 2007) | 2 lines
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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2007-03-05 19:20:45 +00:00
e23433e50e
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r57477 | file | 2007-03-02 12:06:52 -0500 (Fri, 02 Mar 2007) | 10 lines
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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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b3bc675402
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r57318 | file | 2007-03-01 17:21:44 -0500 (Thu, 01 Mar 2007) | 10 lines
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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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2007-03-01 22:23:26 +00:00
fd2db34f15
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r57093 | file | 2007-02-28 13:57:52 -0500 (Wed, 28 Feb 2007) | 10 lines
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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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2007-02-28 19:00:24 +00:00
a3da18c244
Issue 7789 - some telcos want the TON set based on the number, but without the NANP prefix removed
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2007-02-27 00:11:32 +00:00
b3776415aa
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r56888 | russell | 2007-02-26 14:42:21 -0600 (Mon, 26 Feb 2007) | 4 lines
Restore the behavior of Asterisk 1.2 where if a device was not specified in
alsa.conf, then we just use the system default, instead of creating our own
default of hw:0,0. (issue #9139 )
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2007-02-26 20:43:18 +00:00
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r56847 | russell | 2007-02-26 14:04:13 -0600 (Mon, 26 Feb 2007) | 2 lines
Fix a crash in my last change to iax2_indicate(). (issue #9150 )
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2007-02-26 20:04:53 +00:00
82179d1202
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r56785 | russell | 2007-02-26 10:51:18 -0600 (Mon, 26 Feb 2007) | 3 lines
Do more complete locking of the chan_iax2_pvt struct in the indicate callback.
(Problem brought up by Ben Smithurst on the asterisk-dev list)
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2007-02-26 16:54:17 +00:00
5873c9c90f
There is no need to look in the iaxs array for the pvt struct when we already
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have a pointer to it.
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2007-02-26 00:37:03 +00:00
75d387acbc
Doxygen additions, corrections
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2007-02-24 20:29:41 +00:00
e916cf45da
Doxygen updates and corrections
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2007-02-24 19:49:11 +00:00
bc01e39174
Creating new doxygen macro "\extref" to create page that lists
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external libraries and URLs to these. Please help me add these
references.
We might want to create a similar macro "\linuxpackage" to list
the needed Linux packages in popular distributions.
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2007-02-24 19:27:50 +00:00
97ab07a9e8
Allow a Skinny device to monitor a dialplan hint (w00t!).
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See skinny.conf.sample for configuration example.
Note: Some devices (seen on 12SP+/30VIP) will lock up if they monitor too many hints.
This seems to be a hardware limitation - there isn't anything we can do about it.
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2007-02-24 02:23:43 +00:00
a23096985b
Support devicestate requests.
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Now you should be able to subscribe to a Skinny device/line.
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2007-02-24 02:06:44 +00:00
639aee2ec7
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r56569 | qwell | 2007-02-23 20:02:53 -0600 (Fri, 23 Feb 2007) | 4 lines
Make sure to set a speeddials parent on creation.
Don't crash if hold is pressed when no call is active.
Don't return in places that we shouldn't..
Update softkey map when call is connected
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2007-02-24 02:04:34 +00:00
b85a5ffbe3
Make the hashing function calculate something that makes more sense.
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(Thanks to bmd on #asterisk-dev for pointing out my pointless math).
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2007-02-23 22:26:00 +00:00
9fb1404279
Merge team/russell/iax2_performance.
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There is not a large amount of code here and the changes are not very invasive.
However, they should significantly improve performance of chan_iax2 under load.
IAX2 media frames only carry the *source* call number. So, when one arrives,
the correct session that it is a part of has to be matched on IP address, port
number, and call number, instead of just a call number. Had these frames
carried the *destination* call number, this would not be an issue, because that
would be a unique identifier that would make it easy to immediately identify
the correct session.
The way that chan_iax2 did this matching was extremely inefficient. It starts
at the first available call number and traverses each call number sequentially,
locking and unlocking a mutex for each one, to try to match against it. It
would do this regardless of whether the call number was in use or not. So,
for a call with a local call number of 25000, every single incoming media
frame would require a traversal that required 25000 mutex lock and unlock
operations. (Note that the max call number is about 32k).
I have introduced a hash table of active IAX2 calls to improve this lookup
process. The hash is done on the IP address, port number, and call number.
So, for the previous example, a few lock/unlock operations may be done versus
25000 for each frame.
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2007-02-23 21:20:33 +00:00
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r56407 | russell | 2007-02-23 14:20:00 -0600 (Fri, 23 Feb 2007) | 12 lines
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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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2007-02-23 20:21:35 +00:00
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r56231 | file | 2007-02-22 13:49:39 -0500 (Thu, 22 Feb 2007) | 10 lines
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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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2007-02-22 18:53:22 +00:00
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r56125 | oej | 2007-02-22 11:33:55 +0100 (Thu, 22 Feb 2007) | 2 lines
Move message from verbose to debug
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2007-02-22 10:46:09 +00:00
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r56055 | russell | 2007-02-21 19:24:10 -0600 (Wed, 21 Feb 2007) | 3 lines
Restructure a little bit of code to reduce nesting. There is no functionality
change here.
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2007-02-22 01:26:22 +00:00
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r56011 | russell | 2007-02-21 18:57:36 -0600 (Wed, 21 Feb 2007) | 11 lines
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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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2007-02-22 00:59:17 +00:00
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r55954 | qwell | 2007-02-21 14:27:08 -0600 (Wed, 21 Feb 2007) | 4 lines
Fix locking issue, and accept "transport-accept" as a valid accept message.
This should solve issues 8970 and 8503.
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2007-02-21 20:30:54 +00:00
cdf9cab49d
Clarify in the doxygen docs abou RFC2833 compensation flag.
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2007-02-21 20:26:43 +00:00
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r55914 | file | 2007-02-21 12:18:19 -0500 (Wed, 21 Feb 2007) | 2 lines
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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2007-02-21 17:23:42 +00:00
4534fbb7bc
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r55834 | oej | 2007-02-21 09:32:34 +0100 (Wed, 21 Feb 2007) | 2 lines
Issue #8848 - Turn off lamp more quickly after transfer (decrement inuse early on transferer's call leg)
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2007-02-21 08:39:15 +00:00
8f28800765
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r55799 | qwell | 2007-02-20 20:01:36 -0600 (Tue, 20 Feb 2007) | 4 lines
Fix segfault when buddy couldn't be found.
Issue 7764, patch by sailer
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2007-02-21 02:04:10 +00:00
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r55717 | file | 2007-02-20 18:57:03 -0500 (Tue, 20 Feb 2007) | 2 lines
Return behavior I removed. I did not remember that you could just add a localnet entry to make it work.
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2007-02-21 00:00:11 +00:00
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r55688 | file | 2007-02-20 18:08:45 -0500 (Tue, 20 Feb 2007) | 2 lines
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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2007-02-20 23:12:55 +00:00
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r55670 | file | 2007-02-20 17:47:00 -0500 (Tue, 20 Feb 2007) | 10 lines
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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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2007-02-20 22:49:53 +00:00
ae47fc4541
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r55555 | qwell | 2007-02-20 10:53:45 -0600 (Tue, 20 Feb 2007) | 4 lines
No need to cast nor free with strdupa (thanks file)
55555!
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2007-02-20 16:56:58 +00:00
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r55397 | dbailey | 2007-02-19 08:52:59 -0600 (Mon, 19 Feb 2007) | 3 lines
Changed iax2 process thread to detached to correct memory leak due to left over thread context on thread exit.
Modified module unload process to avoid deadlocks on pthread cancels
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2007-02-19 15:24:46 +00:00
8ebf3659aa
Update chan_h323 to new set_rtp_peer definition.
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2007-02-18 19:09:29 +00:00
74f31a51bc
add -Wundef to the --enable-dev-mode flags, so that mistyped macro names in #if expressions will be caught
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convert various #if expressions to #ifdef for macros that may not be defined (and where the value is not important)
Note: two of these changes are in bison generated files which is going to be inconvenient when they are regenerated
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2007-02-18 15:03:42 +00:00
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r55217 | qwell | 2007-02-17 11:10:09 -0600 (Sat, 17 Feb 2007) | 4 lines
Fix an issue where callerid would not be displayed on some phones.
Issue 8995, initial patch and research done by wedhorn
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2007-02-17 17:11:07 +00:00
354c03c4e6
Update chan_jingle to new definition of set_rtp_peer.
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2007-02-17 01:37:29 +00:00
977fb01cdd
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r55086 | file | 2007-02-16 20:16:59 -0500 (Fri, 16 Feb 2007) | 10 lines
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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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2007-02-17 01:22:01 +00:00
4331761b5e
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r55002 | file | 2007-02-16 17:18:46 -0500 (Fri, 16 Feb 2007) | 10 lines
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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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2007-02-16 22:20:55 +00:00
cfb3f84979
Formatting, whitespace fixes
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2007-02-16 14:31:18 +00:00
ba32ee49d0
Adding Realtime Text support (T.140) to Asterisk
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T.140/RFC 2793 is a live communication channel, originally
created for IP based text phones for hearing impaired.
Feels very much like the old Unix talk application.
This code is developed and disclaimed by John Martin of Aupix, UK.
Tested for interoperability by myself and Omnitor in Sweden,
the company that wrote most of the specifications.
A big thank you to everyone involved in this.
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2007-02-16 13:35:44 +00:00
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r54787 | oej | 2007-02-16 13:06:23 +0100 (Fri, 16 Feb 2007) | 2 lines
Issue #7541 - Handle multipart attachments to SIP messages - even if boundary is quoted.
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2007-02-16 12:14:53 +00:00
fef74f1574
Add callgroup and pickupgroup to SIPPEER function. (thanks ramon)
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2007-02-15 15:52:35 +00:00
1f52d1cc81
Issue #7443 - amdtech - Optionally SIP registrations in another
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realtime family.
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2007-02-15 12:10:55 +00:00
daa5f504ec
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r54375 | mattf | 2007-02-14 10:56:40 -0600 (Wed, 14 Feb 2007) | 10 lines
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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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2007-02-14 17:14:42 +00:00
276b570c3e
Issue #9060 - host= parameter in sip.conf stopped working
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caused by outbound proxy patch.
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2007-02-14 16:20:48 +00:00
0653be0c33
Add port number to SIPPEER dialplan function
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2007-02-14 15:27:49 +00:00
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r54204 | russell | 2007-02-13 13:42:00 -0600 (Tue, 13 Feb 2007) | 5 lines
If we fail to create the SIP socket, then return -1 from reload_config() so
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.
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2007-02-13 21:57:31 +00:00
9e99a51802
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r54235 | russell | 2007-02-13 15:31:22 -0600 (Tue, 13 Feb 2007) | 2 lines
Remove a couple of leftover debug messages
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2007-02-13 21:33:03 +00:00
1295d40d77
Be careful with debug messages in trunk, they tend to stay around for release....
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2007-02-11 20:49:38 +00:00
6e139adc56
Small fix in outbound proxy support.
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2007-02-11 20:04:49 +00:00
32495f91f0
Add support for outbound proxy for peers and [general]
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This replaces the older, broken, implementation where a setting in
[general] did not do anything and the [peer] part was broken.
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2007-02-11 19:42:55 +00:00
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r53881 | pcadach | 2007-02-10 01:09:49 -0800 (Сбт, 10 Фев 2007) | 1 line
Fix VLDTMF reception
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2007-02-10 09:23:09 +00:00
5715b49c30
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r53810 | russell | 2007-02-09 18:35:09 -0600 (Fri, 09 Feb 2007) | 24 lines
Merge team/russell/sla_rewrite
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.
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2007-02-10 00:40:57 +00:00
ac4090fce0
add another dependency
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2007-02-10 00:20:57 +00:00
e758ef09b3
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r53779 | kpfleming | 2007-02-09 17:51:29 -0600 (Fri, 09 Feb 2007) | 2 lines
fix awk scripts to work when both MODULEINFO and MAKEOPTS are present in a source file
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r53780 | kpfleming | 2007-02-09 17:51:41 -0600 (Fri, 09 Feb 2007) | 2 lines
add some inter-module dependencies
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r53781 | kpfleming | 2007-02-09 17:52:44 -0600 (Fri, 09 Feb 2007) | 2 lines
another dependency
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2007-02-09 23:53:51 +00:00
cee4bd43dc
Rename this instance of "busy limit" to "busy level" as well
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2007-02-08 17:19:27 +00:00
44c6630e4d
rename busy-limit to busy-level, since it is not a limit
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actually parse the busy-limit option from sip.conf, instead of ignoring it
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2007-02-08 16:41:23 +00:00
bd41ac0ae8
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r53358 | file | 2007-02-07 10:43:39 -0500 (Wed, 07 Feb 2007) | 10 lines
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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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2007-02-07 15:46:30 +00:00
a96e62d9d2
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r53324 | crichter | 2007-02-07 10:22:44 +0100 (Mi, 07 Feb 2007) | 9 lines
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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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2007-02-07 09:51:06 +00:00
17af1bd4c8
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r53143 | oej | 2007-02-05 01:18:34 +0100 (Mon, 05 Feb 2007) | 3 lines
Add some comments on queue system behaviour and how it affects the
SIP channel
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2007-02-05 00:30:03 +00:00
014feba426
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r53138 | file | 2007-02-03 15:05:02 -0600 (Sat, 03 Feb 2007) | 2 lines
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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2007-02-03 21:06:36 +00:00
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r53131 | pcadach | 2007-02-03 02:02:55 -0800 (Сбт, 03 Фев 2007) | 1 line
Remove quote from H.323 vendor string because due to compatibilities with
Nortel Meridian CS1000 reported at www.voip-info.org
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2007-02-03 10:12:20 +00:00
ce8a7c3d9c
Add onHold value to sip show inuse as well.
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2007-02-02 18:21:46 +00:00
cfe66e6b26
Patch based on this patch with small changes for trunk...
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r53109 | oej | 2007-02-02 01:24:03 +0100 (Fri, 02 Feb 2007) | 4 lines
Disable the direct p2p RTP call setup in SIP. You can enable it in sip.conf, but it is now
considered experimental until we solve the AST_CONTROL_ANSWER with payload and videocaps
stuff.
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2007-02-02 00:26:25 +00:00
44a9af3576
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r53104 | file | 2007-02-01 16:24:32 -0600 (Thu, 01 Feb 2007) | 10 lines
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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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2007-02-01 22:26:11 +00:00
bf66c620c3
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r53097 | file | 2007-02-01 15:54:28 -0600 (Thu, 01 Feb 2007) | 10 lines
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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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2007-02-01 21:56:23 +00:00
544f414c0d
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r53085 | oej | 2007-02-01 22:05:34 +0100 (Thu, 01 Feb 2007) | 4 lines
- Clean INC_COUNT flag when we decrement call counter
- If it's still set at time of dialog destruction, make sure we decrement the device call counter properly
before we destroy the dialog
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2007-02-01 21:17:08 +00:00
0b84b386b9
Implementing "busy-limit".
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If you set call limit and busy limit, chan_sip will indicate BUSY for a device
that has reached the busy limit and allow calls up to the call limit, allowing
for call transfers (that generate a new call).
If you only set call limit, chan_sip will not indicate BUSY until that limit
is filled.
This affects SIP subscriptions, call queues and manager applications.
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2007-02-01 20:43:49 +00:00
34eaa61700
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r53079 | oej | 2007-02-01 21:28:54 +0100 (Thu, 01 Feb 2007) | 2 lines
Cleaning up the devicestate callback function
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2007-02-01 20:33:59 +00:00
38b87ec4b7
Signal HOLD status to phones that subscribe for status.
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2007-02-01 19:04:47 +00:00
e88dda8ca9
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r53064 | file | 2007-02-01 11:37:44 -0600 (Thu, 01 Feb 2007) | 2 lines
Fix silly logic. We really want to write UDPTL frames out when the call is up.
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2007-02-01 17:42:08 +00:00
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r53057 | pcadach | 2007-02-01 03:07:41 -0800 (Чтв, 01 Фев 2007) | 1 line
chan_h323 is very stable, so let it built by default
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2007-02-01 11:16:00 +00:00
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r53046 | russell | 2007-01-31 15:32:08 -0600 (Wed, 31 Jan 2007) | 11 lines
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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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2007-01-31 21:35:15 +00:00
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r52952 | russell | 2007-01-30 13:33:12 -0600 (Tue, 30 Jan 2007) | 5 lines
Only set the DTMF flag on the rtp structure if the DTMF mode is actually
RFC2833, not just that it is not INFO. This makes it get set for inband DTMF
as well, which is not valid.
(issue #8936 )
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2007-01-30 19:36:28 +00:00
1fc144435d
Use provided variable for name instead of one in the structure since the structure was just allocated and will be NULL. (issue #8938 reported by st41ker)
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2007-01-30 15:39:09 +00:00
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r52763 | russell | 2007-01-29 18:15:50 -0600 (Mon, 29 Jan 2007) | 13 lines
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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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2007-01-30 00:16:38 +00:00
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r52370 | file | 2007-01-26 19:08:18 -0500 (Fri, 26 Jan 2007) | 10 lines
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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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2007-01-27 00:10:19 +00:00
300f980223
Use atomic operation functions for use/ringing/hold manipulation.
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2007-01-25 18:10:18 +00:00
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r52210 | file | 2007-01-25 12:49:39 -0500 (Thu, 25 Jan 2007) | 2 lines
Drop out variables I accidentally put in.
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2007-01-25 17:51:35 +00:00
afb9151e19
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r52208 | file | 2007-01-25 12:14:53 -0500 (Thu, 25 Jan 2007) | 2 lines
Decrement onHold count if we are hung up on and still on hold. (issue #8909 reported by alexh42)
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2007-01-25 17:17:56 +00:00
2396e24e65
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r52016 | file | 2007-01-24 12:59:55 -0500 (Wed, 24 Jan 2007) | 2 lines
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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2007-01-24 18:04:47 +00:00
273f1d78c7
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r51931 | oej | 2007-01-24 10:30:21 +0100 (Wed, 24 Jan 2007) | 3 lines
Show capabilities *and* preference in general settings in "sip show settings"
(reported by Clona/Telio - Thanks!)
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2007-01-24 09:42:31 +00:00
ee3ab150f6
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r51788 | file | 2007-01-23 17:46:31 -0500 (Tue, 23 Jan 2007) | 2 lines
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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2007-01-23 22:59:55 +00:00
ef4db783c6
Issue #8817 - Registry corruption when packet retransmits fail. (tootai, patchy by oej)
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2007-01-23 15:36:01 +00:00
cd61f69a83
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r51615 | pcadach | 2007-01-22 22:51:51 -0800 (Пнд, 22 Янв 2007) | 1 line
Do not abort Asterisk startup if h323 configuration file not found (reported by mithraen)
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2007-01-23 06:56:26 +00:00
34df128519
Add SRV Lookup support on outbound calls to chan_iax2. It's listed in the RFC so we might want to support it and please don't hurt me Marko ... (issue #7812 reported by drorlb)
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2007-01-23 03:15:04 +00:00
5a25c156c6
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r51558 | file | 2007-01-22 22:00:12 -0500 (Mon, 22 Jan 2007) | 2 lines
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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2007-01-23 03:02:09 +00:00
1a5dfca2a1
Remove (to quote Rizzo) "useless" variable.
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2007-01-22 19:00:25 +00:00
3275357f20
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r51328 | russell | 2007-01-19 13:08:25 -0600 (Fri, 19 Jan 2007) | 5 lines
Fix VLDTMF support in chan_gtalk. AST_FRAME_DTMF and AST_FRAME_DTMF_END are
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.
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2007-01-19 19:09:04 +00:00
dcca8f345f
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r51311 | russell | 2007-01-19 11:49:38 -0600 (Fri, 19 Jan 2007) | 23 lines
Merge the changes from the /team/group/vldtmf_fixup branch.
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...)
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2007-01-19 18:06:03 +00:00
a52fa7e5ad
sizeof() is compatible with format %d so don't be too
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picky on printf formats.
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2007-01-19 17:45:46 +00:00
455ea1c99e
remove variable declaration in the middle of a block
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2007-01-19 17:38:28 +00:00
0a8d59d88a
whoops, fix a cut&paste error...
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2007-01-19 16:35:37 +00:00
03fc21712a
slight change to the initialization of a structure,
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also using '\0' to make it clear we need a (char)0
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2007-01-19 16:32:05 +00:00
3fefb9d516
include "asterisk/zapata.h" to get the zaptel headers.
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2007-01-19 16:28:24 +00:00
dc31eaa025
include "asterisk/zapata.h" instead of looking
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directly for the zaptel.h and tonezone.h
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2007-01-19 16:16:19 +00:00
18c23101a5
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r51274 | dhubbard | 2007-01-18 18:17:32 -0600 (Thu, 18 Jan 2007) | 3 lines
chan_zap compiles without libpri after committing 7877 patch
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2007-01-19 00:28:25 +00:00
33243c6624
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r51272 | dhubbard | 2007-01-18 17:56:49 -0600 (Thu, 18 Jan 2007) | 11 lines
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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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2007-01-19 00:06:35 +00:00
1a06a58250
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r51243 | file | 2007-01-18 13:36:35 -0500 (Thu, 18 Jan 2007) | 2 lines
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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2007-01-18 18:39:21 +00:00
ae286e1843
Update ast_append_ha() usage
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2007-01-18 06:59:22 +00:00
45bd0c4f3f
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r51204 | russell | 2007-01-17 16:09:52 -0600 (Wed, 17 Jan 2007) | 4 lines
Instead of dividing the offset by 2 directly, make it more clear that the
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)
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2007-01-17 23:31:40 +00:00
c56f9184e8
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r51198 | russell | 2007-01-17 15:18:35 -0600 (Wed, 17 Jan 2007) | 11 lines
Merged revisions 51197 via svnmerge from
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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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2007-01-17 21:20:22 +00:00
31224a6b52
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r51172 | file | 2007-01-16 19:46:29 -0500 (Tue, 16 Jan 2007) | 2 lines
Move rescheduling of lagrq/pings into the scheduler callback.
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2007-01-17 00:50:59 +00:00
04426fab2c
Add support for G729 passthrough with Sigma Designs boards. (issue #8829 reported by ywalther)
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2007-01-16 17:23:31 +00:00
3d91c0a0c0
IAX2 remote variables - Bug 7619
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2007-01-16 08:38:59 +00:00
bf5c607c69
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r51087 | file | 2007-01-16 00:55:23 -0500 (Tue, 16 Jan 2007) | 10 lines
Merged revisions 51085 via svnmerge from
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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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2007-01-16 05:56:42 +00:00
b3288f79fe
Feature: allow soundcard to be used in both modes (autoanswer and not),
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selectable by how it is called in the dialplan. This allows a speaker
system hooked up to the soundcard to be used for both ring notification,
as well as paging.
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2007-01-15 02:28:32 +00:00
91dfb27494
Get rid of unneeded code, fix a spelling mistake, and use registry state a bit more. (issue #8402 reported by rizzo)
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2007-01-13 04:56:25 +00:00
033d849bda
Drop trunkrealloc option and just have the maximum size be a configurable option. This is per Kevin's comments on -dev and my own thoughts after I put the previous option in.
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2007-01-13 04:04:04 +00:00
0d781f9553
Ensure error variable is set to 0 or else we might get false error messages. (issue #8798 reported by tootai, fix by anthonyl)
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2007-01-13 03:37:10 +00:00
c4b4615dcd
Merge in trunkrealloc option for chan_iax2. (issue #8267 reported by marcodmb, branch by anthonyl)
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2007-01-13 03:26:04 +00:00
469cf8ab2a
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r50468 | file | 2007-01-11 00:53:09 -0500 (Thu, 11 Jan 2007) | 2 lines
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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2007-01-11 05:54:14 +00:00
4942fd94d2
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r50466 | file | 2007-01-11 00:19:39 -0500 (Thu, 11 Jan 2007) | 2 lines
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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2007-01-11 05:21:03 +00:00
47c515768c
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r50377 | file | 2007-01-10 13:32:29 -0500 (Wed, 10 Jan 2007) | 2 lines
Fix chan_sip not working issue. Let's not prematurely return 0. (issue #8783 reported by st41ker)
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2007-01-10 18:34:45 +00:00
f215636ef3
Allow usedistinctiveringdetection and distinctiveringaftercid to be reset during a reload. (issue #8739 reported by tzafrir)
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2007-01-10 04:21:46 +00:00
444acad95b
Covert some spaces to tabs, and put a list of defines in an enum.
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2007-01-10 02:50:06 +00:00
3246ad09a4
Based on the following patch, changed for trunk...
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r50124 | oej | 2007-01-09 12:25:20 +0100 (Tue, 09 Jan 2007) | 3 lines
- handle re-invites properly in sip_hangup()
- Add some invitestate status changes just to be sure
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2007-01-09 12:25:33 +00:00
5d4f272a90
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r50006 | oej | 2007-01-08 15:26:14 +0100 (Mon, 08 Jan 2007) | 11 lines
Issue #8677 - Handle failure of T.38 re-invite
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.
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2007-01-08 14:31:16 +00:00
835e2ed684
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r49983 | oej | 2007-01-08 14:28:18 +0100 (Mon, 08 Jan 2007) | 3 lines
Issue #8524 , support multiple via header values (tardieu)
Thanks!
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2007-01-08 13:34:56 +00:00
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r49945 | oej | 2007-01-08 10:08:10 +0100 (Mon, 08 Jan 2007) | 2 lines
We only need one forward declaration
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2007-01-08 09:22:53 +00:00
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r49925 | oej | 2007-01-08 09:55:03 +0100 (Mon, 08 Jan 2007) | 2 lines
Issue 8735: Terminate state when extension is unavailable for subscription
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2007-01-08 08:57:35 +00:00
585e0e3e09
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r49890 | file | 2007-01-08 00:11:54 -0500 (Mon, 08 Jan 2007) | 10 lines
Merged revisions 49889 via svnmerge from
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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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2007-01-08 05:13:24 +00:00
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r49831 | tilghman | 2007-01-07 15:24:04 -0600 (Sun, 07 Jan 2007) | 2 lines
Second condition was a subset of the first, so hold was never decremented, thus hint stayed stuck (Issue 8747)
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2007-01-07 21:26:43 +00:00
37182c873e
finish const-ifying ast_func_read()
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2007-01-06 00:13:33 +00:00
cd73a483f1
const-ify some more APIs, and fix rev 49710 from branch-1.4 in a better way here
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2007-01-05 23:32:42 +00:00
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r49705 | qwell | 2007-01-05 17:16:16 -0600 (Fri, 05 Jan 2007) | 4 lines
Make codec_zap and chan_zap also depend on zaptel.
This fixes an issue (8727) with zaptel being in a different directory, using --with-zaptel.
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2007-01-05 23:17:45 +00:00
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r49676 | kpfleming | 2007-01-05 16:16:33 -0600 (Fri, 05 Jan 2007) | 2 lines
reduce stack consumption for AMI and AMI/HTTP requests by nearly 20K in most cases
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2007-01-05 22:43:18 +00:00
a8dc50812f
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r49636 | kpfleming | 2007-01-05 11:09:00 -0600 (Fri, 05 Jan 2007) | 10 lines
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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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2007-01-05 17:10:59 +00:00
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r49600 | kpfleming | 2007-01-04 18:01:40 -0600 (Thu, 04 Jan 2007) | 2 lines
revert the dynamic_list insertion change... that was not the right thing to do
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2007-01-05 00:02:48 +00:00
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r49581 | kpfleming | 2007-01-04 17:50:15 -0600 (Thu, 04 Jan 2007) | 3 lines
create the IAX2 processing threads as background threads so they will use smaller stacks
when we create a dynamic thread, put it on the dynamic_list right away so we don't lose track of it
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2007-01-04 23:54:09 +00:00
5a140436f3
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r49568 | file | 2007-01-04 18:00:50 -0500 (Thu, 04 Jan 2007) | 2 lines
It's possible for the iax2 pvt to disappear, so if it has... don't bother looking for dpentries.
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2007-01-04 23:02:32 +00:00
c5c17a1434
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r49536 | kpfleming | 2007-01-04 15:58:42 -0600 (Thu, 04 Jan 2007) | 2 lines
don't mark these allocations as 'cache' allocations when caching has been disabled
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2007-01-04 21:59:06 +00:00
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r49523 | kpfleming | 2007-01-04 15:06:02 -0600 (Thu, 04 Jan 2007) | 2 lines
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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2007-01-04 21:06:35 +00:00
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r49465 | kpfleming | 2007-01-04 12:31:55 -0600 (Thu, 04 Jan 2007) | 2 lines
only do IAX2 frame caching for voice and video frames
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2007-01-04 18:32:23 +00:00
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r49457 | kpfleming | 2007-01-04 12:05:47 -0600 (Thu, 04 Jan 2007) | 2 lines
make building of codec_gsm against the system GSM library actually work
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r49460 | kpfleming | 2007-01-04 12:16:40 -0600 (Thu, 04 Jan 2007) | 2 lines
don't define this type either if LOW_MEMORY is enabled
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r49461 | kpfleming | 2007-01-04 12:17:01 -0600 (Thu, 04 Jan 2007) | 2 lines
don't do frame header caching in the core if LOW_MEMORY is defined
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2007-01-04 18:19:55 +00:00
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r49458 | kpfleming | 2007-01-04 12:06:51 -0600 (Thu, 04 Jan 2007) | 2 lines
don't do frame caching in LOW_MEMORY mode
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2007-01-04 18:17:24 +00:00
1fe0e3d192
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r49313 | crichter | 2007-01-03 10:06:50 +0100 (Mi, 03 Jan 2007) | 41 lines
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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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r47989 | crichter | 2006-11-24 16:46:13 +0100 (Fr, 24 Nov 2006) | 9 lines
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r47968 | crichter | 2006-11-23 17:10:23 +0100 (Do, 23 Nov 2006) | 1 line
fixed a litle bug regarding HOLD/RETRIEVE. beatufied some logs, changed some loglevels. changed the default value of block_on_alarm
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2007-01-03 11:03:57 +00:00
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r49259 | file | 2007-01-02 20:19:53 -0500 (Tue, 02 Jan 2007) | 2 lines
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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2007-01-03 01:21:41 +00:00
f7ee0abcd2
Removing propably accidentally added debug messages sent to verbose channel
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2007-01-02 20:02:37 +00:00
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r49212 | oej | 2007-01-02 20:58:45 +0100 (Tue, 02 Jan 2007) | 2 lines
Small cleanup of add_t38sdp - it's always enabled at that point in the code
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2007-01-02 20:01:34 +00:00
29058da351
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r49165 | kpfleming | 2007-01-02 07:59:44 -0600 (Tue, 02 Jan 2007) | 2 lines
remove comment that is unrelated to this function
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2007-01-02 14:01:20 +00:00
cb97e0c353
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r49102 | kpfleming | 2007-01-01 17:34:35 -0600 (Mon, 01 Jan 2007) | 2 lines
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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2007-01-01 23:43:43 +00:00
9074555e37
- Add error handling to ast_parse_allow_disallow
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- Use this in chan_sip configuration parsing
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2007-01-01 19:48:31 +00:00
f83b845f08
- Implement error handling in ast_append_ha
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- Use this in chan_sip
- Document ha functions in acl.c
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2007-01-01 19:20:46 +00:00
7eb0f10f34
count is no longer used in the iaxq structure really so let's just make this a statically declared linked list.
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2006-12-31 19:15:23 +00:00
7674260e75
Add version number to useragent string - Issue #8700 , blanchet - THANKS!
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2006-12-31 09:31:23 +00:00
ae649fef27
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r49063 | file | 2006-12-29 22:37:22 -0500 (Fri, 29 Dec 2006) | 2 lines
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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2006-12-30 03:40:42 +00:00
1f8001e816
Convert various comments to doxygen format.
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2006-12-29 13:25:07 +00:00
fc86ea2eed
Removing extra output
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2006-12-29 11:02:28 +00:00
b54feaffec
Only include include files once (imported from 1.4)
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2006-12-27 21:27:57 +00:00
906a51a1af
Only set "rfc2833compensate" option once
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2006-12-27 20:30:03 +00:00
8794577966
Only handle T38 options once
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2006-12-27 20:26:30 +00:00
820d82470a
-Remove "localmask" setting (deprecated in earlier version)
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- Remove "musiconhold" and "musicclass" settings (also deprecated earlier)
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2006-12-27 19:57:06 +00:00
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r48988 | kpfleming | 2006-12-27 12:33:22 -0600 (Wed, 27 Dec 2006) | 2 lines
make the option actually match the documentation
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2006-12-27 18:34:24 +00:00
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r48987 | kpfleming | 2006-12-27 12:29:13 -0600 (Wed, 27 Dec 2006) | 2 lines
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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2006-12-27 18:33:44 +00:00
29ed493b40
Be politically correct
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2006-12-27 18:02:10 +00:00
da7a35a1cc
Add support for buggy Cisco MWI firmware > 8.0.3 (issue 8575 - flewid)
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2006-12-27 16:56:11 +00:00
3713ce7ed1
Cleanup of handle_common_options
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2006-12-27 16:35:41 +00:00
a12dc35eb3
Reset invitestate when sending new invite
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2006-12-27 16:26:02 +00:00
d69e9086cc
Issue #8600 - bogus SDP Content Length in T.38 re-invite
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2006-12-27 16:01:07 +00:00
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r48964 | file | 2006-12-25 23:31:58 -0500 (Mon, 25 Dec 2006) | 2 lines
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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2006-12-26 04:34:07 +00:00
09f75aa6dc
rename the structs struct tone_zone_sound and struct tone_zone
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defined in indications.h to ind_tone_zone_sound and ind_tone_zone,
to avoid conflicts with the structs with the same names
defined in tonezone.h
Hope i haven't missed any instance.
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2006-12-25 06:38:09 +00:00
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r48948 | russell | 2006-12-24 16:19:37 -0500 (Sun, 24 Dec 2006) | 3 lines
Fix a typo in an error message that indicated that the MGCP channel type could
not be registered, instead of the correct type, OSS.
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2006-12-24 21:21:21 +00:00
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r48944 | russell | 2006-12-24 02:25:38 -0500 (Sun, 24 Dec 2006) | 11 lines
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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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r48940 | russell | 2006-12-24 01:49:31 -0500 (Sun, 24 Dec 2006) | 11 lines
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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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2006-12-24 06:51:37 +00:00
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r48888 | qwell | 2006-12-22 15:40:20 -0600 (Fri, 22 Dec 2006) | 2 lines
Note to self: Run make before committing...
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2006-12-22 21:40:56 +00:00
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r48870 | qwell | 2006-12-22 14:43:05 -0600 (Fri, 22 Dec 2006) | 2 lines
Fix for issue 7774 - patch by alamantia
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2006-12-22 20:43:38 +00:00
7b338004bf
a quick fix to app_sms.c to get rid of cursed compiler warnings so I can compile under --enable-dev-mode
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2006-12-21 19:44:20 +00:00
9327720c37
As per bug 7978, this version introduces the jittertargetextra option in config files
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2006-12-21 00:24:08 +00:00
fa5931d0c3
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r48586 | kpfleming | 2006-12-19 15:28:04 -0600 (Tue, 19 Dec 2006) | 2 lines
suppress compiler warnings in this module until it can be improved
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2006-12-19 21:28:43 +00:00
381d49c9af
Clean up find_idle_thread function and use atomic operations for dynamic thread count.
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2006-12-18 17:30:53 +00:00
c17181aae3
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r48564 | file | 2006-12-18 12:15:49 -0500 (Mon, 18 Dec 2006) | 2 lines
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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2006-12-18 17:18:18 +00:00
d5eef300d8
define a mask SIP_INSECURE sam as for other sets of flags.
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2006-12-17 12:47:08 +00:00
6bdd1a3388
Merged revisions 48504 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48504 | file | 2006-12-15 14:38:51 -0500 (Fri, 15 Dec 2006) | 2 lines
Hold call structure lock in places where a qualify or peer action can destroy it.
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2006-12-15 19:40:29 +00:00
12565be69e
Merged revisions 48502 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48502 | file | 2006-12-15 14:24:15 -0500 (Fri, 15 Dec 2006) | 2 lines
Lock network retransmission queue in all places that it is used.
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2006-12-15 19:26:45 +00:00
7db228ac95
replace ast_safe_string_alloc() with asprintf()
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2006-12-15 15:40:57 +00:00
a736605bf4
replace ast_safe_string_alloc() with asprintf()
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2006-12-15 15:39:29 +00:00
168deb2366
Issue #8592 - treat 504 as congestion (imported from 1.2/1.4)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48488 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-12-15 10:58:36 +00:00
bf2172faac
Update to latest IANA specs
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48485 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-12-15 10:54:06 +00:00
315f8bb1b2
Merged revisions 48478 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48478 | file | 2006-12-15 01:28:05 -0500 (Fri, 15 Dec 2006) | 2 lines
Use a wakeup variable so that we don't wait on IO indefinitely if packets need to be retransmitted.
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2006-12-15 06:31:26 +00:00
52396dff26
revert check for a zaptel transcoder related definition that was done in the
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wrong module.
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2006-12-13 00:56:30 +00:00
666d526aad
Fix various spelling mistakes in comments.
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2006-12-12 22:32:20 +00:00
4490b5710d
Make chan_zap inform you that your version of zaptel is too old instead of
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just failing to compile.
It seems like the proper way to do this would be in the configure script.
However, that wouldn't help existing checkouts unless we forced the configure
script to be executed after any code was changed.
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2006-12-12 22:27:22 +00:00
23e606f70a
Merged revisions 48372 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48372 | murf | 2006-12-09 20:04:18 -0700 (Sat, 09 Dec 2006) | 9 lines
Merged revisions 48371 via svnmerge from
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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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2006-12-10 03:14:27 +00:00
a9415d4b8e
convert the thread IO state and type to use enums.
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2006-12-09 16:44:41 +00:00
86ad1435c1
Merged revisions 48363 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48363 | russell | 2006-12-09 10:59:42 -0500 (Sat, 09 Dec 2006) | 8 lines
Use locking when accessing the registrations list. This list is not actually
used very often, so the likelihood of there being a problem is pretty small,
but still possible. For example, if the CLI command to list the registrations
was called at the same time that a reload was occurring and the registrations
list was getting destroyed and rebuilt, a crash could occur.
In passing, go ahead and convert this list to use the linked list macros.
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2006-12-09 16:04:06 +00:00
62c5be4fda
chan_iax2 has an extremely large function, socket_process(), to handle incoming
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frames. The function, before this commit, was roughly 1400 lines long. So, I
am working on breaking this up into functions so that the code is easier to
follow and debug. Also, I will be committing these changes in chunks as I do
them to ease tracking down any potentially introduced problems.
Break out roughly 150 lines from socket_process() and introduce a new function,
socket_process_meta() which handles the parsing of an incoming meta frame.
Also, restructure some of this code a bit to reduce the deep nesting that was
in this code.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48360 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-12-09 07:10:55 +00:00
6592c25a69
- Fix a few spelling mistakes
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- Use sizeof() to pass an array size to a function
- Use a single bit for a variable in the chan_iax2_pvt stuct since that is all
it needs.
- Add some comments about the iaxs, iaxl, and lastused arrays.
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2006-12-09 05:04:38 +00:00
2b42422940
Handle multiple 487's correctly
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2006-12-06 16:54:27 +00:00
c39f887ac8
Don't send Contact in SIP Messages (imported from 1.2/1.4).
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Reported by Gunnar at Omnitor.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48318 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-12-06 12:34:58 +00:00
ec8591d04c
Constify a bunch of usage strings for CLI commands.
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2006-12-06 07:35:31 +00:00
a44e55a3f3
Constify a bunch of the usage strings for CLI commands.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48302 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-12-06 07:23:32 +00:00
b761fb83cd
Instead of creating an unused instance of an unnamed enum, give it a name.
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2006-12-06 07:16:09 +00:00
22142e1aa0
Cleaning up handle_response a bit. (Imported from 1.4)
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2006-12-02 22:03:14 +00:00
df8c773783
Send CANCEL to call with early media (PROGRESS INBAND).
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This is imported from branch "invitestate" and "invitestate-1.4"
***
***
*** IF YOU HAVE ISSUES WITH BYEs/CANCELs - PLEASE UPDATE AND TEST AGAIN!
*** Thank you!
***
***
/Olle
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2006-12-02 20:30:58 +00:00
fc6de9e982
Invitestate updates
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48213 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-12-02 19:11:02 +00:00
c23bc46089
- Disable RTP timeouts during T.38 transmission
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- Encapsulate RTP timers to the RTP structure, so we have one set for video and one for audio
- Document RTP keepalive configuration option
- Cleanup and document the monitor support function to hangup on RTP timeouts
- Add RTP keepalive to SIP show settings
Imported from 1.4 with modifications for trunk.
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2006-12-02 12:05:40 +00:00
4ce5b7c080
- Remove T.38 early media, since T.38 requires two way communication (imported from 1.4)
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- Small fixes to limitonpeer
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2006-12-01 18:16:16 +00:00
869101028b
Merged revisions 48168 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48168 | file | 2006-11-30 16:18:24 -0500 (Thu, 30 Nov 2006) | 2 lines
Do not do a partial bridge for Google Talk since we need to handle STUN. (issue #8448 reported by phsultan)
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2006-11-30 21:22:01 +00:00
813f8d26fe
Issue #8319 (imported from 1.2, 1.4) - Increment nonce-count properly (noriyuki)
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2006-11-30 20:55:43 +00:00
a77935b674
Merged revisions 48158 via svnmerge from
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r48158 | file | 2006-11-30 15:07:55 -0500 (Thu, 30 Nov 2006) | 10 lines
Merged revisions 48157 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r48157 | file | 2006-11-30 15:06:43 -0500 (Thu, 30 Nov 2006) | 2 lines
Only print out debug message if bridged channel is not NULL. (issue #8412 reported by jubilex)
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2006-11-30 20:09:46 +00:00
389ed67b41
Merging patch from 1.2/1.4. I think this was originally spotted by
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Luigi, but hit me in the back today.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48130 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-30 09:23:25 +00:00
bf9a5e1dee
I am pretty sure that oej only meant to change the variable name in the source, not the configuration option name so let's turn it back to srvlookup instead of global_srvlookup. (issue #8442 reported by jtodd)
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2006-11-30 03:29:42 +00:00
c16c606cff
Add a comment to note near some code that performs a very expensive operation
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that occurs for every incoming media frame.
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2006-11-28 23:13:37 +00:00
e021f5fbb9
Merged revisions 48088 via svnmerge from
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r48088 | file | 2006-11-28 11:57:16 -0500 (Tue, 28 Nov 2006) | 10 lines
Merged revisions 48087 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r48087 | file | 2006-11-28 11:56:01 -0500 (Tue, 28 Nov 2006) | 2 lines
According to the research I have done we never needed to include compiler.h in the first place so let's not! (issue #8430 reported by edguy3)
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2006-11-28 16:59:20 +00:00
42fb134171
Change error message (imported from 1.4)
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2006-11-27 06:44:40 +00:00
2102534748
Little fix so we use the right message
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2006-11-25 21:50:42 +00:00
94ec7877b5
Make compiler happier
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2006-11-25 21:46:58 +00:00
8b8187c6f8
bug fix
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2006-11-25 21:35:23 +00:00
1ded1fd55a
Make sure we don't send a group reset on a group larger than 32 CICs
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48011 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-25 21:32:33 +00:00
cb06f87d11
Add ss7 show linkset command
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2006-11-25 20:54:38 +00:00
4dfd954ac3
Updates to show linkset command
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48009 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-25 20:30:04 +00:00
e46e42ceec
Changing ERROR to lesser level. Imported from 1.2/1.4
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2006-11-25 09:48:30 +00:00
a4cc4d03c9
Remove unused memory allocation
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2006-11-23 11:04:25 +00:00
db08a0f346
Don't over-deprecate... :-)
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2006-11-22 11:28:12 +00:00
4221020191
Treat 101 as 100, not 183 session progress
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47893 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-21 15:25:38 +00:00
230376c7c0
Properly reset schedule items (rizzo)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47844 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-20 11:36:45 +00:00
f14aa70947
Merged revisions 47823 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r47823 | rizzo | 2006-11-18 18:59:35 +0100 (Sat, 18 Nov 2006) | 5 lines
fix bug 7450 - Parsing fails if From header contains angle brackets
(the bug was only in a corner case where the < was right after the
opening quote, and the fix is trivial).
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47824 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-18 18:03:42 +00:00
34ec231da7
prevent the sound thread from consuming all the available CPU
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doing busy-wait on the output audio device.
As it is set now, it tries to push a frame every 10ms,
which is still too frequent but avoids deep restructuring
of the code (which i should do, though).
Note, this is only for ring tones, regular audio coming
from the network is still delivered as soon as it is
available.
Eventually this could well end up in the 1.4 branch, but
since i am probably the only user of chan_oss there isn't
much urgency to do that.
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2006-11-18 08:19:41 +00:00
03ee553343
Add some sense of link state. Don't make calls if link is down. Only reset if we're bringing the
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link up for the first time.
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2006-11-17 18:26:33 +00:00
b55e30b075
remove an unused function
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47789 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-17 11:31:55 +00:00
779f4c0e0a
use the regexp cli support on some of the command
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47788 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-17 11:15:54 +00:00
57d4919427
Make sure we choose the last channel as the dchannel if it's not E1 (for BRI). ( #8077 ) Thanks Tzafrir.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47784 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-17 01:05:28 +00:00
24c241c5a9
convert two entries to new style
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2006-11-16 23:05:50 +00:00
65fb29bc03
Remove the interim variable for range modifications, and set it on the structure directly.
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Also move the default checking to where it gets set initially.
Fixes suggested by file.
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2006-11-16 22:51:00 +00:00
2c9643d49f
convert some handlers to new style.
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2006-11-16 22:44:29 +00:00
bfd630682e
Add ability to modify range for dring matching.
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Issue #8369 , patch by ssuehring, modified slightly by me.
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2006-11-16 22:32:23 +00:00
a63d971afc
fix indentation
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2006-11-16 22:03:27 +00:00
Olle Johansson
Joshua Colp
Kevin P. Fleming
Russell Bryant
Christian Richter
Mark Spencer
Matthew Fredrickson
Steve Murphy
Jason Parker
Tilghman Lesher
Dwayne M. Hubbard
Nadi Sarrar
Doug Bailey
Paul Cadach
Luigi Rizzo
Matt O'Gorman