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Summary:ASTERISK-05953: Call-limit does not function while using queue
Reporter:Kenneth Holm (saitech)Labels:
Date Opened:2006-01-02 11:54:52.000-0600Date Closed:2008-01-15 16:28:31.000-0600
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) call_limit.cyrax.patch
( 1) call_limit.debug
Description:I am having some problems with limiting my sip calls. I want to eliminate the call waiting for X-Lite/Eyebeam, so im using the call-limit function to ensure that only 1 sip call is allowed pr user.

If the asterisk receive a sip call and put it into a queue. The first attempt to call the user is failed, if he already have a call. But the after the first announce in the queue, the call will not fail, and starts as a call waiting for the user.

I have also tried the deprecated funtions incominglimit and outgoinglimit both sat to 1. I have used thes setup before with asterisk 1.0.7 where it did function pretty good.
Comments:By: Olle Johansson (oej) 2006-01-02 11:58:30.000-0600

Please always attach SIP debug output when reporting SIP bugs. Set debug and verbose to 4 and turn on SIP debugging. Trace all information for the first and the second call. Thank you.

By: Jason Parker (jparker) 2006-01-02 12:34:41.000-0600

Reopening per request.

I forgot to add a note why this was closed, and for that, I apologize.

I'll close 6110 instead.

By: Jason Parker (jparker) 2006-01-02 12:43:13.000-0600

Changing the version back to 1.2.1...  I'll stop touching this issue after this...

Sorry. :(

By: wes (whoiswes) 2006-01-14 11:44:55.000-0600

[this is my first time posting to a bug list - be kind]

I can provide more information on this bug, as I am experiencing the same issue (on both 1.0.7 and 1.2.1).  This does not just affect queue calls, but incoming and internal calls as well - the exact problem seems to be related to the device state being changed to 1 when a device still is on an active call.

I have posted a lengthy thread on the digium forums and will paste relevant content here as well:

http://forums.digium.com/viewtopic.php?t=3716

I have tried to output the debug logs and have attached them.  Call-limit for all three lines is set to 1.  x998 calls x999 and the call is answered.  x2001 calls x999 and is rejected due to a usage limit of 1.  However, the devicestate is changed to not in use, even though there is still an active call on the line.  

A subsequent call from x2001 to x999 (with x998 still active) allows 2001 to ring through because the line is showing a usage limit of 0.  I can confirm that the CLI is showing the inuse variable as 0.

I'm not sure if I have provided the correct information, and my apologies if I have not.  I am fairly new to Asterisk and am not a programmer by trade.

Thanks, and please let me know if I can help further.

Wes



By: Daniele Martini (cyrax) 2006-01-15 05:14:13.000-0600

Hello.. this is my first post too.

I think that this is what's happening:

Let's suppose the "inuse" variable is equal to call_limit.
When someone try to call the extension the function sip_call
does a:
res=update_call_counter(p, INC_CALL_LIMIT) of the connection,

the update_call_counter sees that inuse is equal to call_limit and so returns
-1 (without increasing the inuse variabile)

So sip_call returns -1, and then the asterisk core runs sip_hangup.

sip_hanghup runs a update_call_counter(p,DEC_CALL_LIMIT) so it decreases the inuse var...

In a few word the problem is that sip_hanhup decrease the counter also if it was not incremented.

This could be a quick fix:
the update_call_counter should increase the inuse variable also when it is greater than call_limit, so the next hangup will set it down to normal...

By: wes (whoiswes) 2006-01-16 09:45:17.000-0600

SUCCESS!!!

After playing with this for an hour this morning, I have fixed the issue.

On lines 2223-2253 of chan_sip.c, we have a switch statement for either incrementing or decrementing the call counter.  The original code is here:

case INC_CALL_LIMIT:
if (*call_limit > 0 ) {
if (*inuse >= *call_limit) {
ast_log(LOG_ERROR, "Call %s %s '%s' rejected due to usage limit of %d\n", outgoing ? "to" : "from", u ? "user":"peer", name, *call_limit);
if (u)
ASTOBJ_UNREF(u,sip_destroy_user);
else
ASTOBJ_UNREF(p,sip_destroy_peer);
return -1;
}
}
(*inuse)++;
if (option_debug > 1 || sipdebug) {
ast_log(LOG_DEBUG, "Call %s %s '%s' is %d out of %d\n", outgoing ? "to" : "from", u ? "user":"peer", name, *inuse, *call_limit);
}
break;

Note how the (*inuse)++ is located AFTER the checking of the call-limit.  I moved that line up to the very top of the case statement, like this:

case INC_CALL_LIMIT:
(*inuse)++;
if (*call_limit > 0 ) {
if (*inuse > *call_limit) {
ast_log(LOG_ERROR, "Call %s %s '%s' rejected due to usage limit of %d\n", outgoing ? "to" : "from", u ? "user":"peer", name, *call_limit);
if (u)
ASTOBJ_UNREF(u,sip_destroy_user);
else
ASTOBJ_UNREF(p,sip_destroy_peer);
return -1;
}
}
if (option_debug > 1 || sipdebug) {
ast_log(LOG_DEBUG, "Call %s %s '%s' is %d out of %d\n", outgoing ? "to" : "from", u ? "user":"peer", name, *inuse, *call_limit);
}
break;

I also changed
if (*inuse >= *call_limit) {
to
if (*inuse > *call_limit) {

This allows for the inuse counter to be increased BEFORE the call-limit check is made.  This way, when the call is hungup, the counter can be decremented to the proper value.

We've checked both SIP-to-SIP and queue calls - both work as they should now.  The call counters seem to be keeping the correct values and calls are not ringing through anymore.

We did notice one thing - one of our macros uses ChanIsAvail - this seems to open a channel to check on it's status - it is doing the same thing, where the call counter is not incremented, so when ChanIsAvail hangs up, it's decrementing the call counter incorrectly, but that is a separate issue for a separate bug.

In any case, the changes I made above are the only ones required, to my knowledge.

Thanks for the input cyrax - I was looking at that part of the code yesterday and the way you worded it put it in much better context for me.

I'm welcome to everyone's input on this - like I said, this is the first time I've done anything like this, so I might have done or said something stupid, and you all have my apologies if that is the case.

Thanks!

Wes

By: BJ Weschke (bweschke) 2006-01-16 10:43:28.000-0600

whoiswes: do you have a disclaimer on file with Digium?

By: wes (whoiswes) 2006-01-16 11:38:40.000-0600

no, and if i need one, then i will edit my previous post.

EDIT - I read the guidelines - I will get a form filled out and faxed in.

I wish I could just say "use the code!" -  I only changed one line...



By: BJ Weschke (bweschke) 2006-01-16 11:44:24.000-0600

I don't think one is absolutely necessary for something so small as this, but it would be great for you to get one on file if you're going to become a regular contributor.

By: BJ Weschke (bweschke) 2006-01-16 12:46:28.000-0600

if you are incrementing inuse now before the check, shouldn't we decrement it again in the case where you are going to return -1 and not allow the call? otherwise, it seems like this inuse is going to get artificially inflated when we're denying calls based on limit, no?

By: Daniele Martini (cyrax) 2006-01-16 14:15:23.000-0600

bweschke: if the call is rejected it will be hangupped, and the function sip_hangup() will decrease the counter. (The problem was that when a call is rejected sip_hangup() decrease the counter also if it was not incremented)

whoiswes: I think that you should put the incrementing statement just AFTER the if(*call_limit > 0). If a call limit is not set (call_limit equal to 0) there is no need to increment the inuse variable..

(Sorry for the bad english)

By: wes (whoiswes) 2006-01-16 15:39:57.000-0600

Cyrax - you are exactly right on the placement of the increment counter - I have made that change on my test box.

Unfortunately, after playing around with this some more, it seems as though my fix is not totally viable.  

After several calls back and forth between three extensions, the inuse variable is getting incremented but not decremented on outbound calls.  so the inbound issue is fixed, but now the outbound counter is broken.

I still think we're on the right path, but don't want to go any further till someone with more experience has had a chance to look at this.

I am open to doing any testing and/or bug hunting if someone wants give me a bit of direction.

Thanks for the input everyone.

By: BJ Weschke (bweschke) 2006-01-16 15:58:01.000-0600

Assigned to request feedback from oej 'the SIP man' for Asterisk so he can provide some more feedback on this issue. :)

By: Daniele Martini (cyrax) 2006-01-16 17:15:44.000-0600

I've uploaded a patch that maybe will fix the problem (sorry i didn't test it..).

I've added a new flag SIP_INC_COUNT for use with struct sip_pvt ( this struct is used to keep info for a call and is passed to update_call_counter).

When update_call_counter increase the inuse counter, it also set the flag SIP_INC_COUNT on the connection.

When update_call_counter has to decrease the inuse counter, it do that only if the SIP_INC_COUNT flag is set.

Well, I hope i used flags in the right way...

Whoiswes: could you please test this patch?

The patch is made from revision 8111 of TRUNK.

By: Kenneth Holm (saitech) 2006-01-16 17:42:02.000-0600

i can confirm, that the patch works fine with 1.2.1 final. I havent seen any problems in the past 10 minutes test. SO somethings is right now :)

By: wes (whoiswes) 2006-01-16 20:37:55.000-0600

I will install and test tomorrow morning and get you guys some feedback.

Thanks for the patch cyrax, hopefully you got what I missed!

W

By: wes (whoiswes) 2006-01-18 14:18:10.000-0600

The patch is working fine for me as well.  I only have three phones set up, but I can send and receive calls from my call-limit=1 device without any issues, and the counter is being incremented and decremented correctly.

One small thing - call-limit=1 for a type=friend should allow one outgoing and one incoming call at the same time, correct?  This is not happening on my system, so I'm not sure if this is due to the patch or something else.

Everything else works fine.

Thanks Cyrax!

Wes

By: Kenneth Holm (saitech) 2006-01-18 15:24:47.000-0600

If im correct, incominglimit and outgoinglimit has been merged to call-limit, and call-limit is the limit of overall connections. If you define call-limit=1 will it be only 1 sip call at a time.

I can be wrong, if i am, its an error, though my asterisk box is functioning fine, with call-limit = 1, and restrict to only 1 call at a time.

By: BJ Weschke (bweschke) 2006-01-18 15:29:54.000-0600

saitech is right. If no one else has any objections, I'll try and commit this latest patch later on tonight.

By: wes (whoiswes) 2006-01-18 15:35:45.000-0600

Saitech,

To keep this ticket clean, I'll move the call-limit discussion here:
http://forums.digium.com/viewtopic.php?p=12157ASTERISK-11587

By: BJ Weschke (bweschke) 2006-01-18 20:56:50.000-0600

cyrax: before we can commit your patch, is there a disclaimer on file for you? please advise.

By: Daniele Martini (cyrax) 2006-01-19 05:46:39.000-0600

I just sent the disclaimer..
(from +39-045-89XXXXX)
my name is Daniele Martini

By: wes (whoiswes) 2006-01-19 07:27:03.000-0600

I had a thought this morning, and came in to test it.

The bug as originally posted was about having the queue deliver only one call at a time to an agent.  This can be done through call-limit, obviously, but what if you need to have multiple lines available, but only want the queue to deliver one call at a time (which makes sense to me, and is actually how the bulk of our company would work - one line to make outgoings, but if anyone is in queue, we'd want the second line to ring in, but only one at a time).

Anyways, I tested this morning, and even though the queue recognizes the agent is busy, it will attempt to route another call to that agent.  In other words, the only reason the second call doesn't ring through is because of call-limit being 1.  To me, this seems flawed.

I just tested with call-limit=2, and sure enough the queue will immediately send the second caller in queue to the agent, even though they have an active call.  Again, to me, this isn't right.

I'm not sure how to proceed with this, but please let me know what your thoughts are, if any.

Thanks,

Wes

By: BJ Weschke (bweschke) 2006-01-19 07:40:30.000-0600

whoiswes: this is known behavior with app_queue. The way you'd fix this for now would be to add PauseQueueMember to the dialplan when dialing back your agents and UnPauseQueueMember when the call frees up (the 'h' exten).

By: wes (whoiswes) 2006-01-19 09:54:28.000-0600

We don't use AgentCallBackLogin - we use AddQueueMember and RemoveQueueMember through a PHP interface I built - our agents don't sign themselves in and out, our managers do it for them (long story).

I'll have to come up with something quirky it looks like - other than putting callers in queue, there is nothing in our dialplans that controls queues or agents - we leave everything up to app_queue.

Thanks, if I come up with anything I'll post it here.

By: Boris Erdmann (be4ct) 2006-01-21 07:27:59.000-0600

bweschke, whoiswes:

I made a patch which provides a config switch in queues.conf
"ring-inuse" by which you can disable ringing channels
that are in use (chan_sip seems to provide correct AST_DEVICE_INUSE
status).

It simply copies the behaviour of not ringing paused members in
ring_entry() of app_queue.c.

Any interest in that? (thou this is becoming off topic)

UPDATE:

OK, I will shortly motivate why the Pause/UnpuaseQueueMember solution
does not work out so well (*having* written the "necessary" macros):

Since SIP phones can handle more than one call it is necessary to do
global counters on each phone about how many calls are currently taking place.
Else you don't know when to unpause. This alone is tricky to get right
through the whole dialplan (even more so since macros don't seem to support
the 'h' extension in 1.2.x)
Well, counters? What atomic operations does the dialplan provide if you
think of hanging up and receiving another call at the same time. Your counters
will get mixed up if you don't implement some locking.



By: wes (whoiswes) 2006-01-21 07:46:29.000-0600

yes, i'm definitely interested, and it sounds like that would be a great addition to the source code!

hit me off list - whoiswes AT gmail DOT com - i'd love to try your patch out, if you wouldn't mind.

By: wes (whoiswes) 2006-01-21 09:45:21.000-0600

I just finished testing be4ct's patch, and it fixes app_queue so that if an agent is busy, the queue won't even try to call them.

http://bugs.digium.com/view.php?id=6315

I'm posting this here as it is relevant to the original bug description.

In any case, many thanks to everyone involved - in less than a week, both of the major issues we were having with asterisk have been fixed - I am blown away by how much all of you helped and how kind you were to a newb like me.

I hope to be able to contribute something meaningful someday to return the favor.

Thanks again!

Wes

By: Daniele Martini (cyrax) 2006-01-22 06:46:45.000-0600

so, is the call_limit patch going to go in the repository ?

By: BJ Weschke (bweschke) 2006-01-22 08:09:57.000-0600

cyrax: your patch was committed to /trunk this morning. Thanks!

By: BJ Weschke (bweschke) 2006-01-22 08:10:29.000-0600

committed cyrax's patch to /trunk

By: Digium Subversion (svnbot) 2008-01-15 16:21:11.000-0600

Repository: asterisk
Revision: 8432

U   trunk/channels/chan_sip.c

------------------------------------------------------------------------
r8432 | bweschke | 2008-01-15 16:21:11 -0600 (Tue, 15 Jan 2008) | 2 lines

Correct some scenarios where CALL_LIMIT could not be getting adjusted properly allowing chan_sip to send calls when it really shouldn't. Bug ASTERISK-5953

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=8432

By: Digium Subversion (svnbot) 2008-01-15 16:21:12.000-0600

Repository: asterisk
Revision: 8433

U   branches/1.2/channels/chan_sip.c

------------------------------------------------------------------------
r8433 | bweschke | 2008-01-15 16:21:12 -0600 (Tue, 15 Jan 2008) | 3 lines

Bug fix: Correct some scenarios where CALL_LIMIT could not be getting adjusted properly allowing chan_sip to send calls when it really shouldn't. Bug ASTERISK-5953


------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=8433

By: Digium Subversion (svnbot) 2008-01-15 16:21:20.000-0600

Repository: asterisk
Revision: 8441

_U  team/oej/astum/
U   team/oej/astum/apps/Makefile
U   team/oej/astum/apps/app_authenticate.c
U   team/oej/astum/apps/app_chanspy.c
U   team/oej/astum/apps/app_curl.c
U   team/oej/astum/apps/app_db.c
U   team/oej/astum/apps/app_dial.c
U   team/oej/astum/apps/app_dictate.c
U   team/oej/astum/apps/app_directory.c
U   team/oej/astum/apps/app_disa.c
U   team/oej/astum/apps/app_exec.c
U   team/oej/astum/apps/app_externalivr.c
U   team/oej/astum/apps/app_festival.c
U   team/oej/astum/apps/app_hasnewvoicemail.c
U   team/oej/astum/apps/app_image.c
U   team/oej/astum/apps/app_macro.c
U   team/oej/astum/apps/app_meetme.c
U   team/oej/astum/apps/app_mixmonitor.c
U   team/oej/astum/apps/app_osplookup.c
U   team/oej/astum/apps/app_page.c
U   team/oej/astum/apps/app_parkandannounce.c
U   team/oej/astum/apps/app_playback.c
U   team/oej/astum/apps/app_privacy.c
U   team/oej/astum/apps/app_queue.c
U   team/oej/astum/apps/app_random.c
U   team/oej/astum/apps/app_read.c
U   team/oej/astum/apps/app_readfile.c
U   team/oej/astum/apps/app_record.c
U   team/oej/astum/apps/app_sayunixtime.c
U   team/oej/astum/apps/app_senddtmf.c
U   team/oej/astum/apps/app_sendtext.c
U   team/oej/astum/apps/app_setcallerid.c
U   team/oej/astum/apps/app_skel.c
U   team/oej/astum/apps/app_sql_postgres.c
U   team/oej/astum/apps/app_stack.c
U   team/oej/astum/apps/app_talkdetect.c
U   team/oej/astum/apps/app_transfer.c
U   team/oej/astum/apps/app_url.c
U   team/oej/astum/apps/app_userevent.c
U   team/oej/astum/apps/app_verbose.c
U   team/oej/astum/apps/app_voicemail.c
U   team/oej/astum/apps/app_while.c
U   team/oej/astum/apps/app_zapras.c
U   team/oej/astum/callerid.c
U   team/oej/astum/channel.c
U   team/oej/astum/channels/chan_agent.c
U   team/oej/astum/channels/chan_iax2.c
U   team/oej/astum/channels/chan_mgcp.c
U   team/oej/astum/channels/chan_sip.c
U   team/oej/astum/codecs/codec_a_mu.c
U   team/oej/astum/codecs/codec_adpcm.c
U   team/oej/astum/codecs/codec_alaw.c
U   team/oej/astum/codecs/codec_g723_1.c
U   team/oej/astum/codecs/codec_g726.c
U   team/oej/astum/codecs/codec_gsm.c
U   team/oej/astum/codecs/codec_ilbc.c
U   team/oej/astum/codecs/codec_lpc10.c
U   team/oej/astum/codecs/codec_speex.c
U   team/oej/astum/codecs/codec_ulaw.c
U   team/oej/astum/codecs/gsm/Makefile
U   team/oej/astum/configs/sip.conf.sample
U   team/oej/astum/doc/README.variables
U   team/oej/astum/funcs/func_cdr.c
U   team/oej/astum/funcs/func_cut.c
U   team/oej/astum/funcs/func_logic.c
U   team/oej/astum/funcs/func_math.c
U   team/oej/astum/funcs/func_md5.c
U   team/oej/astum/funcs/func_odbc.c
U   team/oej/astum/funcs/func_rand.c
U   team/oej/astum/funcs/func_strings.c
U   team/oej/astum/include/asterisk/utils.h
U   team/oej/astum/pbx.c
U   team/oej/astum/res/res_agi.c
U   team/oej/astum/res/res_crypto.c
U   team/oej/astum/res/res_features.c
U   team/oej/astum/res/res_indications.c
U   team/oej/astum/res/res_monitor.c
U   team/oej/astum/res/res_musiconhold.c
U   team/oej/astum/res/res_osp.c

------------------------------------------------------------------------
r8441 | oej | 2008-01-15 16:21:19 -0600 (Tue, 15 Jan 2008) | 167 lines

Merged revisions 8359-8362,8368,8370-8372,8378-8381,8387,8393,8401-8404,8407-8411,8413,8420-8422,8426,8428,8432,8436 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
r8359 | mattf | 2006-01-21 00:19:49 +0100 (Sat, 21 Jan 2006) | 2 lines

Fix comments in sip.conf (ASTERISK-5977)

........
r8360 | russell | 2006-01-21 00:23:00 +0100 (Sat, 21 Jan 2006) | 2 lines

formatting and doxygen fixes (issue ASTERISK-6140)

........
r8361 | mattf | 2006-01-21 00:28:37 +0100 (Sat, 21 Jan 2006) | 2 lines

fix for codec_gsm on ia64 (ASTERISK-6134)

........
r8362 | russell | 2006-01-21 01:42:25 +0100 (Sat, 21 Jan 2006) | 2 lines

remove some useless checks after calls to ast_strdupa

........
r8368 | russell | 2006-01-21 04:09:01 +0100 (Sat, 21 Jan 2006) | 2 lines

remove optimization where its benefits are negligible

........
r8370 | tilghman | 2006-01-21 06:05:45 +0100 (Sat, 21 Jan 2006) | 2 lines

Bug 6096 - callerid_parse cleanup

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r8371 | tilghman | 2006-01-21 06:15:56 +0100 (Sat, 21 Jan 2006) | 2 lines

Bug 5515 - Devicestate and API documentation update

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r8372 | tilghman | 2006-01-21 06:22:18 +0100 (Sat, 21 Jan 2006) | 2 lines

Bug 4872 - Make Asterisk paths available to AGIs via environmental variables

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r8378 | russell | 2006-01-21 09:01:24 +0100 (Sat, 21 Jan 2006) | 2 lines

remove some useless checks of the result of ast_strdupa

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r8379 | russell | 2006-01-21 09:13:12 +0100 (Sat, 21 Jan 2006) | 2 lines

remove lots of useless checks of the result of ast_strdupa

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r8380 | russell | 2006-01-21 09:23:57 +0100 (Sat, 21 Jan 2006) | 2 lines

remove useless checks of the result of ast_strdupa

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r8381 | russell | 2006-01-21 09:45:39 +0100 (Sat, 21 Jan 2006) | 2 lines

conversions to use allocation wrappers (issue ASTERISK-6117)

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r8387 | russell | 2006-01-21 18:50:04 +0100 (Sat, 21 Jan 2006) | 3 lines

revert my pass through the tree to remove checks of the result of ast_strdupa
(revisions 8378 through 8381)

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r8393 | russell | 2006-01-21 19:11:40 +0100 (Sat, 21 Jan 2006) | 3 lines

don't do the memcpy inside of ast_strdupa if we know that __builtin_alloca
was not successful

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r8401 | russell | 2006-01-21 21:20:06 +0100 (Sat, 21 Jan 2006) | 3 lines

finish reverting my pass through the tree to remove checks of the result of
ast_strdupa, this one is revision 8362

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r8402 | tilghman | 2006-01-21 21:32:17 +0100 (Sat, 21 Jan 2006) | 2 lines

Bug 5936 - Cannot AddQueueMember on realtime queue, if queue not yet loaded (different fix than 1.2)

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r8403 | russell | 2006-01-21 21:57:06 +0100 (Sat, 21 Jan 2006) | 2 lines

on this pass, only remove duplicate log messages

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r8404 | russell | 2006-01-21 21:58:57 +0100 (Sat, 21 Jan 2006) | 1 line


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r8407 | russell | 2006-01-21 22:29:06 +0100 (Sat, 21 Jan 2006) | 2 lines

clean up formatting to conform to coding guidelines and fix some typos (issue ASTERISK-6101)

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r8408 | russell | 2006-01-21 22:50:09 +0100 (Sat, 21 Jan 2006) | 5 lines

- conversion to allocation wrappers
- get rid of some compiler warnings due to usused variables and const-ification
- remove some unnecessary uses of malloc/strncpy/free and replace with ast_strdupa
(based on the patch from issue ASTERISK-6115)

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r8409 | russell | 2006-01-21 22:57:01 +0100 (Sat, 21 Jan 2006) | 2 lines

oops, don't build app_sql_postgres by default ...

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r8410 | russell | 2006-01-21 23:09:06 +0100 (Sat, 21 Jan 2006) | 5 lines

- conversions to allocation wrappers
- replace malloc/memset with ast_calloc
- replace malloc/ast_copy_string with ast_strdup
(based on patch from issue ASTERISK-6139)

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r8411 | russell | 2006-01-22 00:05:19 +0100 (Sun, 22 Jan 2006) | 2 lines

const-ify some fields in the ast_exten and ast_include structures (issue ASTERISK-6110)

........
r8413 | russell | 2006-01-22 00:17:52 +0100 (Sun, 22 Jan 2006) | 1 line


........
r8420 | russell | 2006-01-22 03:06:33 +0100 (Sun, 22 Jan 2006) | 1 line


........
r8421 | russell | 2006-01-22 03:10:19 +0100 (Sun, 22 Jan 2006) | 3 lines

prevent writing outside of the provided workspace when calculating a
substring (issue ASTERISK-6111)

........
r8422 | russell | 2006-01-22 03:23:38 +0100 (Sun, 22 Jan 2006) | 2 lines

add 'show channeltype' CLI command (issue ASTERISK-5992, with small modifications)

........
r8426 | tilghman | 2006-01-22 08:18:02 +0100 (Sun, 22 Jan 2006) | 2 lines

Bug 6148 - Add PARKEDAT variable; also cleaned up application help to fit 80-column screen.

........
r8428 | tilghman | 2006-01-22 09:09:02 +0100 (Sun, 22 Jan 2006) | 2 lines

Bug 6262 - New function STRPTIME

........
r8432 | bweschke | 2006-01-22 16:08:51 +0100 (Sun, 22 Jan 2006) | 2 lines

Correct some scenarios where CALL_LIMIT could not be getting adjusted properly allowing chan_sip to send calls when it really shouldn't. Bug ASTERISK-5953

........
r8436 | tilghman | 2006-01-22 18:28:42 +0100 (Sun, 22 Jan 2006) | 2 lines

Bug 6312 - Macro janitor

........

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=8441

By: Digium Subversion (svnbot) 2008-01-15 16:21:34.000-0600

Repository: asterisk
Revision: 8453

_U  team/russell/autoconf_and_menuselect/
U   team/russell/autoconf_and_menuselect/app.c
U   team/russell/autoconf_and_menuselect/apps/Makefile
U   team/russell/autoconf_and_menuselect/apps/app_authenticate.c
U   team/russell/autoconf_and_menuselect/apps/app_chanspy.c
U   team/russell/autoconf_and_menuselect/apps/app_curl.c
U   team/russell/autoconf_and_menuselect/apps/app_db.c
U   team/russell/autoconf_and_menuselect/apps/app_dial.c
U   team/russell/autoconf_and_menuselect/apps/app_dictate.c
U   team/russell/autoconf_and_menuselect/apps/app_directory.c
U   team/russell/autoconf_and_menuselect/apps/app_disa.c
U   team/russell/autoconf_and_menuselect/apps/app_exec.c
U   team/russell/autoconf_and_menuselect/apps/app_externalivr.c
U   team/russell/autoconf_and_menuselect/apps/app_festival.c
U   team/russell/autoconf_and_menuselect/apps/app_hasnewvoicemail.c
U   team/russell/autoconf_and_menuselect/apps/app_image.c
U   team/russell/autoconf_and_menuselect/apps/app_macro.c
U   team/russell/autoconf_and_menuselect/apps/app_meetme.c
U   team/russell/autoconf_and_menuselect/apps/app_mixmonitor.c
U   team/russell/autoconf_and_menuselect/apps/app_osplookup.c
U   team/russell/autoconf_and_menuselect/apps/app_page.c
U   team/russell/autoconf_and_menuselect/apps/app_parkandannounce.c
U   team/russell/autoconf_and_menuselect/apps/app_playback.c
U   team/russell/autoconf_and_menuselect/apps/app_privacy.c
U   team/russell/autoconf_and_menuselect/apps/app_queue.c
U   team/russell/autoconf_and_menuselect/apps/app_random.c
U   team/russell/autoconf_and_menuselect/apps/app_read.c
U   team/russell/autoconf_and_menuselect/apps/app_readfile.c
U   team/russell/autoconf_and_menuselect/apps/app_record.c
U   team/russell/autoconf_and_menuselect/apps/app_sayunixtime.c
U   team/russell/autoconf_and_menuselect/apps/app_senddtmf.c
U   team/russell/autoconf_and_menuselect/apps/app_sendtext.c
U   team/russell/autoconf_and_menuselect/apps/app_setcallerid.c
U   team/russell/autoconf_and_menuselect/apps/app_skel.c
U   team/russell/autoconf_and_menuselect/apps/app_sql_postgres.c
U   team/russell/autoconf_and_menuselect/apps/app_stack.c
U   team/russell/autoconf_and_menuselect/apps/app_talkdetect.c
U   team/russell/autoconf_and_menuselect/apps/app_transfer.c
U   team/russell/autoconf_and_menuselect/apps/app_url.c
U   team/russell/autoconf_and_menuselect/apps/app_userevent.c
U   team/russell/autoconf_and_menuselect/apps/app_verbose.c
U   team/russell/autoconf_and_menuselect/apps/app_voicemail.c
U   team/russell/autoconf_and_menuselect/apps/app_while.c
U   team/russell/autoconf_and_menuselect/apps/app_zapras.c
U   team/russell/autoconf_and_menuselect/callerid.c
U   team/russell/autoconf_and_menuselect/channel.c
U   team/russell/autoconf_and_menuselect/channels/chan_agent.c
U   team/russell/autoconf_and_menuselect/channels/chan_iax2.c
U   team/russell/autoconf_and_menuselect/channels/chan_sip.c
U   team/russell/autoconf_and_menuselect/doc/README.variables
U   team/russell/autoconf_and_menuselect/funcs/func_cdr.c
U   team/russell/autoconf_and_menuselect/funcs/func_cut.c
U   team/russell/autoconf_and_menuselect/funcs/func_logic.c
U   team/russell/autoconf_and_menuselect/funcs/func_math.c
U   team/russell/autoconf_and_menuselect/funcs/func_md5.c
U   team/russell/autoconf_and_menuselect/funcs/func_strings.c
U   team/russell/autoconf_and_menuselect/include/asterisk/astosp.h
U   team/russell/autoconf_and_menuselect/pbx.c
U   team/russell/autoconf_and_menuselect/res/res_crypto.c
U   team/russell/autoconf_and_menuselect/res/res_features.c
U   team/russell/autoconf_and_menuselect/res/res_indications.c
U   team/russell/autoconf_and_menuselect/res/res_monitor.c
U   team/russell/autoconf_and_menuselect/res/res_musiconhold.c
U   team/russell/autoconf_and_menuselect/res/res_osp.c

------------------------------------------------------------------------
r8453 | russell | 2008-01-15 16:21:33 -0600 (Tue, 15 Jan 2008) | 122 lines

Merged revisions 8401-8404,8407-8411,8413,8420-8422,8426,8428,8432,8436,8438,8443,8446-8447 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
r8401 | russell | 2006-01-21 15:20:06 -0500 (Sat, 21 Jan 2006) | 3 lines

finish reverting my pass through the tree to remove checks of the result of
ast_strdupa, this one is revision 8362

................
r8402 | tilghman | 2006-01-21 15:32:17 -0500 (Sat, 21 Jan 2006) | 2 lines

Bug 5936 - Cannot AddQueueMember on realtime queue, if queue not yet loaded (different fix than 1.2)

................
r8403 | russell | 2006-01-21 15:57:06 -0500 (Sat, 21 Jan 2006) | 2 lines

on this pass, only remove duplicate log messages

................
r8404 | russell | 2006-01-21 15:58:57 -0500 (Sat, 21 Jan 2006) | 1 line


................
r8407 | russell | 2006-01-21 16:29:06 -0500 (Sat, 21 Jan 2006) | 2 lines

clean up formatting to conform to coding guidelines and fix some typos (issue ASTERISK-6101)

................
r8408 | russell | 2006-01-21 16:50:09 -0500 (Sat, 21 Jan 2006) | 5 lines

- conversion to allocation wrappers
- get rid of some compiler warnings due to usused variables and const-ification
- remove some unnecessary uses of malloc/strncpy/free and replace with ast_strdupa
(based on the patch from issue ASTERISK-6115)

................
r8409 | russell | 2006-01-21 16:57:01 -0500 (Sat, 21 Jan 2006) | 2 lines

oops, don't build app_sql_postgres by default ...

................
r8410 | russell | 2006-01-21 17:09:06 -0500 (Sat, 21 Jan 2006) | 5 lines

- conversions to allocation wrappers
- replace malloc/memset with ast_calloc
- replace malloc/ast_copy_string with ast_strdup
(based on patch from issue ASTERISK-6139)

................
r8411 | russell | 2006-01-21 18:05:19 -0500 (Sat, 21 Jan 2006) | 2 lines

const-ify some fields in the ast_exten and ast_include structures (issue ASTERISK-6110)

................
r8413 | russell | 2006-01-21 18:17:52 -0500 (Sat, 21 Jan 2006) | 1 line


................
r8420 | russell | 2006-01-21 21:06:33 -0500 (Sat, 21 Jan 2006) | 1 line


................
r8421 | russell | 2006-01-21 21:10:19 -0500 (Sat, 21 Jan 2006) | 3 lines

prevent writing outside of the provided workspace when calculating a
substring (issue ASTERISK-6111)

................
r8422 | russell | 2006-01-21 21:23:38 -0500 (Sat, 21 Jan 2006) | 2 lines

add 'show channeltype' CLI command (issue ASTERISK-5992, with small modifications)

................
r8426 | tilghman | 2006-01-22 02:18:02 -0500 (Sun, 22 Jan 2006) | 2 lines

Bug 6148 - Add PARKEDAT variable; also cleaned up application help to fit 80-column screen.

................
r8428 | tilghman | 2006-01-22 03:09:02 -0500 (Sun, 22 Jan 2006) | 2 lines

Bug 6262 - New function STRPTIME

................
r8432 | bweschke | 2006-01-22 10:08:51 -0500 (Sun, 22 Jan 2006) | 2 lines

Correct some scenarios where CALL_LIMIT could not be getting adjusted properly allowing chan_sip to send calls when it really shouldn't. Bug ASTERISK-5953

................
r8436 | tilghman | 2006-01-22 12:28:42 -0500 (Sun, 22 Jan 2006) | 2 lines

Bug 6312 - Macro janitor

................
r8438 | russell | 2006-01-22 12:53:14 -0500 (Sun, 22 Jan 2006) | 10 lines

Merged revisions 8437 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r8437 | russell | 2006-01-22 12:47:13 -0500 (Sun, 22 Jan 2006) | 2 lines

fix MixMonitor crash (issue ASTERISK-6161, probably others)

........

................
r8443 | russell | 2006-01-22 13:42:06 -0500 (Sun, 22 Jan 2006) | 2 lines

eliminate some compiler warnings

................
r8446 | russell | 2006-01-22 14:04:37 -0500 (Sun, 22 Jan 2006) | 1 line


................
r8447 | russell | 2006-01-22 14:09:50 -0500 (Sun, 22 Jan 2006) | 2 lines

fix memory leak from not freeing the list of queue members when freeing a queue

................

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=8453

By: Digium Subversion (svnbot) 2008-01-15 16:21:36.000-0600

Repository: asterisk
Revision: 8455

_U  team/russell/make_output/
U   team/russell/make_output/apps/Makefile
U   team/russell/make_output/apps/app_authenticate.c
U   team/russell/make_output/apps/app_chanspy.c
U   team/russell/make_output/apps/app_curl.c
U   team/russell/make_output/apps/app_db.c
U   team/russell/make_output/apps/app_dial.c
U   team/russell/make_output/apps/app_dictate.c
U   team/russell/make_output/apps/app_directory.c
U   team/russell/make_output/apps/app_disa.c
U   team/russell/make_output/apps/app_exec.c
U   team/russell/make_output/apps/app_externalivr.c
U   team/russell/make_output/apps/app_festival.c
U   team/russell/make_output/apps/app_hasnewvoicemail.c
U   team/russell/make_output/apps/app_image.c
U   team/russell/make_output/apps/app_macro.c
U   team/russell/make_output/apps/app_meetme.c
U   team/russell/make_output/apps/app_mixmonitor.c
U   team/russell/make_output/apps/app_osplookup.c
U   team/russell/make_output/apps/app_page.c
U   team/russell/make_output/apps/app_parkandannounce.c
U   team/russell/make_output/apps/app_playback.c
U   team/russell/make_output/apps/app_privacy.c
U   team/russell/make_output/apps/app_queue.c
U   team/russell/make_output/apps/app_random.c
U   team/russell/make_output/apps/app_read.c
U   team/russell/make_output/apps/app_readfile.c
U   team/russell/make_output/apps/app_realtime.c
U   team/russell/make_output/apps/app_record.c
U   team/russell/make_output/apps/app_sayunixtime.c
U   team/russell/make_output/apps/app_senddtmf.c
U   team/russell/make_output/apps/app_sendtext.c
U   team/russell/make_output/apps/app_setcallerid.c
U   team/russell/make_output/apps/app_skel.c
U   team/russell/make_output/apps/app_sql_postgres.c
U   team/russell/make_output/apps/app_stack.c
U   team/russell/make_output/apps/app_talkdetect.c
U   team/russell/make_output/apps/app_transfer.c
U   team/russell/make_output/apps/app_url.c
U   team/russell/make_output/apps/app_userevent.c
U   team/russell/make_output/apps/app_verbose.c
U   team/russell/make_output/apps/app_voicemail.c
U   team/russell/make_output/apps/app_while.c
U   team/russell/make_output/apps/app_zapras.c
U   team/russell/make_output/channels/chan_agent.c
U   team/russell/make_output/channels/chan_iax2.c
U   team/russell/make_output/channels/chan_sip.c
U   team/russell/make_output/doc/README.variables
U   team/russell/make_output/funcs/func_cdr.c
U   team/russell/make_output/funcs/func_cut.c
U   team/russell/make_output/funcs/func_logic.c
U   team/russell/make_output/funcs/func_math.c
U   team/russell/make_output/funcs/func_md5.c
U   team/russell/make_output/funcs/func_odbc.c
U   team/russell/make_output/funcs/func_rand.c
U   team/russell/make_output/funcs/func_strings.c
U   team/russell/make_output/include/asterisk/astosp.h
U   team/russell/make_output/res/res_crypto.c
U   team/russell/make_output/res/res_features.c
U   team/russell/make_output/res/res_indications.c
U   team/russell/make_output/res/res_monitor.c
U   team/russell/make_output/res/res_musiconhold.c
U   team/russell/make_output/res/res_osp.c

------------------------------------------------------------------------
r8455 | russell | 2008-01-15 16:21:36 -0600 (Tue, 15 Jan 2008) | 122 lines

Merged revisions 8401-8404,8407-8411,8413,8420-8422,8426,8428,8432,8436,8438,8443,8446-8447 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
r8401 | russell | 2006-01-21 15:20:06 -0500 (Sat, 21 Jan 2006) | 3 lines

finish reverting my pass through the tree to remove checks of the result of
ast_strdupa, this one is revision 8362

................
r8402 | tilghman | 2006-01-21 15:32:17 -0500 (Sat, 21 Jan 2006) | 2 lines

Bug 5936 - Cannot AddQueueMember on realtime queue, if queue not yet loaded (different fix than 1.2)

................
r8403 | russell | 2006-01-21 15:57:06 -0500 (Sat, 21 Jan 2006) | 2 lines

on this pass, only remove duplicate log messages

................
r8404 | russell | 2006-01-21 15:58:57 -0500 (Sat, 21 Jan 2006) | 1 line


................
r8407 | russell | 2006-01-21 16:29:06 -0500 (Sat, 21 Jan 2006) | 2 lines

clean up formatting to conform to coding guidelines and fix some typos (issue ASTERISK-6101)

................
r8408 | russell | 2006-01-21 16:50:09 -0500 (Sat, 21 Jan 2006) | 5 lines

- conversion to allocation wrappers
- get rid of some compiler warnings due to usused variables and const-ification
- remove some unnecessary uses of malloc/strncpy/free and replace with ast_strdupa
(based on the patch from issue ASTERISK-6115)

................
r8409 | russell | 2006-01-21 16:57:01 -0500 (Sat, 21 Jan 2006) | 2 lines

oops, don't build app_sql_postgres by default ...

................
r8410 | russell | 2006-01-21 17:09:06 -0500 (Sat, 21 Jan 2006) | 5 lines

- conversions to allocation wrappers
- replace malloc/memset with ast_calloc
- replace malloc/ast_copy_string with ast_strdup
(based on patch from issue ASTERISK-6139)

................
r8411 | russell | 2006-01-21 18:05:19 -0500 (Sat, 21 Jan 2006) | 2 lines

const-ify some fields in the ast_exten and ast_include structures (issue ASTERISK-6110)

................
r8413 | russell | 2006-01-21 18:17:52 -0500 (Sat, 21 Jan 2006) | 1 line


................
r8420 | russell | 2006-01-21 21:06:33 -0500 (Sat, 21 Jan 2006) | 1 line


................
r8421 | russell | 2006-01-21 21:10:19 -0500 (Sat, 21 Jan 2006) | 3 lines

prevent writing outside of the provided workspace when calculating a
substring (issue ASTERISK-6111)

................
r8422 | russell | 2006-01-21 21:23:38 -0500 (Sat, 21 Jan 2006) | 2 lines

add 'show channeltype' CLI command (issue ASTERISK-5992, with small modifications)

................
r8426 | tilghman | 2006-01-22 02:18:02 -0500 (Sun, 22 Jan 2006) | 2 lines

Bug 6148 - Add PARKEDAT variable; also cleaned up application help to fit 80-column screen.

................
r8428 | tilghman | 2006-01-22 03:09:02 -0500 (Sun, 22 Jan 2006) | 2 lines

Bug 6262 - New function STRPTIME

................
r8432 | bweschke | 2006-01-22 10:08:51 -0500 (Sun, 22 Jan 2006) | 2 lines

Correct some scenarios where CALL_LIMIT could not be getting adjusted properly allowing chan_sip to send calls when it really shouldn't. Bug ASTERISK-5953

................
r8436 | tilghman | 2006-01-22 12:28:42 -0500 (Sun, 22 Jan 2006) | 2 lines

Bug 6312 - Macro janitor

................
r8438 | russell | 2006-01-22 12:53:14 -0500 (Sun, 22 Jan 2006) | 10 lines

Merged revisions 8437 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r8437 | russell | 2006-01-22 12:47:13 -0500 (Sun, 22 Jan 2006) | 2 lines

fix MixMonitor crash (issue ASTERISK-6161, probably others)

........

................
r8443 | russell | 2006-01-22 13:42:06 -0500 (Sun, 22 Jan 2006) | 2 lines

eliminate some compiler warnings

................
r8446 | russell | 2006-01-22 14:04:37 -0500 (Sun, 22 Jan 2006) | 1 line


................
r8447 | russell | 2006-01-22 14:09:50 -0500 (Sun, 22 Jan 2006) | 2 lines

fix memory leak from not freeing the list of queue members when freeing a queue

................

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=8455

By: Digium Subversion (svnbot) 2008-01-15 16:22:02.000-0600

Repository: asterisk
Revision: 8478

_U  team/crichter/0.3.0/
U   team/crichter/0.3.0/apps/app_queue.c
U   team/crichter/0.3.0/channel.c
U   team/crichter/0.3.0/channels/chan_sip.c
U   team/crichter/0.3.0/pbx.c

------------------------------------------------------------------------
r8478 | crichter | 2008-01-15 16:22:01 -0600 (Tue, 15 Jan 2008) | 47 lines

Merged revisions 8394,8412,8414,8418,8429,8433,8437,8445 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r8394 | tilghman | 2006-01-21 19:29:39 +0100 (Sa, 21 Jan 2006) | 2 lines

Bug 5936 - AddQueueMember fails on realtime queue, if queue not yet loaded

........
r8412 | russell | 2006-01-22 00:17:06 +0100 (So, 22 Jan 2006) | 2 lines

prevent the possibility of writing outside of the available workspace (issue ASTERISK-6111)

........
r8414 | russell | 2006-01-22 00:43:14 +0100 (So, 22 Jan 2006) | 2 lines

temporarily revert substring fix pending the result of the discussion in issue ASTERISK-6111

........
r8418 | russell | 2006-01-22 03:05:41 +0100 (So, 22 Jan 2006) | 3 lines

add a modified fix to prevent writing outside of the provided workspace when
calculating a substring (issue ASTERISK-6111)

........
r8429 | tilghman | 2006-01-22 09:52:49 +0100 (So, 22 Jan 2006) | 2 lines

Bug 6281 - Cannot set more than a single header with SIPAddHeader

........
r8433 | bweschke | 2006-01-22 16:13:41 +0100 (So, 22 Jan 2006) | 3 lines

Bug fix: Correct some scenarios where CALL_LIMIT could not be getting adjusted properly allowing chan_sip to send calls when it really shouldn't. Bug ASTERISK-5953


........
r8437 | russell | 2006-01-22 18:47:13 +0100 (So, 22 Jan 2006) | 2 lines

fix MixMonitor crash (issue ASTERISK-6161, probably others)

........
r8445 | russell | 2006-01-22 20:03:53 +0100 (So, 22 Jan 2006) | 2 lines

fix memory leak from not freeing the queue member list when freeing an old queue

........

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=8478

By: Digium Subversion (svnbot) 2008-01-15 16:28:25.000-0600

Repository: asterisk
Revision: 8891

_U  team/oej/managerstuff/
D   team/oej/managerstuff/ChangeLog
U   team/oej/managerstuff/apps/app_dial.c
U   team/oej/managerstuff/apps/app_festival.c
U   team/oej/managerstuff/apps/app_meetme.c
U   team/oej/managerstuff/apps/app_milliwatt.c
U   team/oej/managerstuff/apps/app_queue.c
U   team/oej/managerstuff/ast_expr2.c
U   team/oej/managerstuff/ast_expr2.fl
U   team/oej/managerstuff/ast_expr2.h
U   team/oej/managerstuff/ast_expr2.y
U   team/oej/managerstuff/ast_expr2f.c
U   team/oej/managerstuff/asterisk.c
U   team/oej/managerstuff/channel.c
U   team/oej/managerstuff/channels/chan_features.c
U   team/oej/managerstuff/channels/chan_iax2.c
U   team/oej/managerstuff/channels/chan_sip.c
U   team/oej/managerstuff/channels/chan_zap.c
U   team/oej/managerstuff/loader.c
U   team/oej/managerstuff/logger.c
U   team/oej/managerstuff/pbx.c
U   team/oej/managerstuff/res/res_features.c
U   team/oej/managerstuff/utils/astman.c

------------------------------------------------------------------------
r8891 | oej | 2008-01-15 16:28:24 -0600 (Tue, 15 Jan 2008) | 202 lines

Merged revisions 8112,8122,8124,8134,8140,8162,8173,8194,8232,8242,8276,8281,8320,8347,8394,8412,8414,8418,8429,8433,8437,8445,8537,8562,8573,8588,8600,8608,8619,8632,8666,8677,8710,8729,8758,8785,8808 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r8112 | kpfleming | 2006-01-17 00:51:37 +0100 (Tue, 17 Jan 2006) | 2 lines

do rlimit check _after_ reading config file, in case 'dumpcore' is specified there

........
r8122 | kpfleming | 2006-01-17 14:11:55 +0100 (Tue, 17 Jan 2006) | 2 lines

update CLI copyright notice

........
r8124 | mogorman | 2006-01-17 17:55:30 +0100 (Tue, 17 Jan 2006) | 3 lines

Fixed code ordering of logger_init and queue_log_init
bug 6263

........
r8134 | mattf | 2006-01-17 19:29:57 +0100 (Tue, 17 Jan 2006) | 2 lines

Backport of fix for ASTERISK-5936

........
r8140 | mogorman | 2006-01-17 21:10:29 +0100 (Tue, 17 Jan 2006) | 3 lines

Stop any generators running on a channel when
festival is called as described in 5996

........
r8162 | mogorman | 2006-01-18 01:47:04 +0100 (Wed, 18 Jan 2006) | 4 lines

Changed order of autoload so that pbx_ comes before
channels, and in doing so cause bug 6002 to not
be an issue

........
r8173 | russell | 2006-01-18 03:49:21 +0100 (Wed, 18 Jan 2006) | 2 lines

remove ChangeLog from the 1.2 branch.  It will only be present in the tags.

........
r8194 | mogorman | 2006-01-18 22:02:06 +0100 (Wed, 18 Jan 2006) | 3 lines

Solves issue with the login proccess in meetme
patch from 6136

........
r8232 | russell | 2006-01-19 05:17:45 +0100 (Thu, 19 Jan 2006) | 3 lines

fix a seg fault due to assuming that space gets allocatted on the stack in the
same order that we declare the variables (issue ASTERISK-6130)

........
r8242 | russell | 2006-01-19 05:56:48 +0100 (Thu, 19 Jan 2006) | 3 lines

fix Message-Account header to use the ip address if the fromdomain
isn't set (issue ASTERISK-6118)

........
r8276 | tilghman | 2006-01-19 20:14:37 +0100 (Thu, 19 Jan 2006) | 2 lines

Bug 6072 - Memory leaks in the expression parser

........
r8281 | oej | 2006-01-19 20:40:28 +0100 (Thu, 19 Jan 2006) | 2 lines

Enable "musicclass" setting for sip peers as per the config sample.

........
r8320 | mogorman | 2006-01-20 02:00:46 +0100 (Fri, 20 Jan 2006) | 3 lines

solved problem with delayreject and iax trunking
bug 4291

........
r8347 | russell | 2006-01-20 19:34:42 +0100 (Fri, 20 Jan 2006) | 2 lines

fix invalid value of prev_q (issue ASTERISK-6142)

........
r8394 | tilghman | 2006-01-21 19:29:39 +0100 (Sat, 21 Jan 2006) | 2 lines

Bug 5936 - AddQueueMember fails on realtime queue, if queue not yet loaded

........
r8412 | russell | 2006-01-22 00:17:06 +0100 (Sun, 22 Jan 2006) | 2 lines

prevent the possibility of writing outside of the available workspace (issue ASTERISK-6111)

........
r8414 | russell | 2006-01-22 00:43:14 +0100 (Sun, 22 Jan 2006) | 2 lines

temporarily revert substring fix pending the result of the discussion in issue ASTERISK-6111

........
r8418 | russell | 2006-01-22 03:05:41 +0100 (Sun, 22 Jan 2006) | 3 lines

add a modified fix to prevent writing outside of the provided workspace when
calculating a substring (issue ASTERISK-6111)

........
r8429 | tilghman | 2006-01-22 09:52:49 +0100 (Sun, 22 Jan 2006) | 2 lines

Bug 6281 - Cannot set more than a single header with SIPAddHeader

........
r8433 | bweschke | 2006-01-22 16:13:41 +0100 (Sun, 22 Jan 2006) | 3 lines

Bug fix: Correct some scenarios where CALL_LIMIT could not be getting adjusted properly allowing chan_sip to send calls when it really shouldn't. Bug ASTERISK-5953


........
r8437 | russell | 2006-01-22 18:47:13 +0100 (Sun, 22 Jan 2006) | 2 lines

fix MixMonitor crash (issue ASTERISK-6161, probably others)

........
r8445 | russell | 2006-01-22 20:03:53 +0100 (Sun, 22 Jan 2006) | 2 lines

fix memory leak from not freeing the queue member list when freeing an old queue

........
r8537 | oej | 2006-01-24 14:15:13 +0100 (Tue, 24 Jan 2006) | 2 lines

Issue ASTERISK-6148 - never send response to ACK. (Reported by whiskerp)

........
r8562 | oej | 2006-01-24 20:21:15 +0100 (Tue, 24 Jan 2006) | 2 lines

Issue 6114: Don't hangup on BYE/ALSO with no channel.

........
r8573 | mattf | 2006-01-24 21:37:30 +0100 (Tue, 24 Jan 2006) | 2 lines

Backport fix for ASTERISK-6071, hangup on polarity reversal

........
r8588 | kpfleming | 2006-01-24 23:32:09 +0100 (Tue, 24 Jan 2006) | 2 lines

ensure that channel cannot become zombie after we check but before we try to start indications

........
r8600 | russell | 2006-01-24 23:55:32 +0100 (Tue, 24 Jan 2006) | 2 lines

completely arbitrary whitespace change for testing something with svnmerge ...

........
r8608 | kpfleming | 2006-01-25 02:50:52 +0100 (Wed, 25 Jan 2006) | 2 lines

ensure hangup cause code is handled properly when channel does not return a frame (issue ASTERISK-6186)

........
r8619 | russell | 2006-01-25 06:38:36 +0100 (Wed, 25 Jan 2006) | 2 lines

don't leak almost 200 bytes for each new channel (issue ASTERISK-6170)

........
r8632 | oej | 2006-01-25 10:46:43 +0100 (Wed, 25 Jan 2006) | 2 lines

Issue ASTERISK-6276 - the "timebomb" bug. Patch by Markster over GPRS

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r8666 | russell | 2006-01-25 19:39:44 +0100 (Wed, 25 Jan 2006) | 2 lines

fix memory leak (inspired by issue ASTERISK-6190)

........
r8677 | russell | 2006-01-25 20:14:43 +0100 (Wed, 25 Jan 2006) | 3 lines

don't call ast_update_realtime with uninitialized variables if we get a
registration with an expirey of 0 seconds (issue ASTERISK-6016)

........
r8710 | oej | 2006-01-26 15:39:36 +0100 (Thu, 26 Jan 2006) | 2 lines

Issue 5898: Registrations does not get deleted if there's an active SIP dialog

........
r8729 | russell | 2006-01-26 20:42:35 +0100 (Thu, 26 Jan 2006) | 2 lines

fix problem with dtmf on e&m (issue ASTERISK-6203)

........
r8758 | tilghman | 2006-01-27 01:52:12 +0100 (Fri, 27 Jan 2006) | 2 lines

Bug 6072 - Revisions to the source bison and flex files don't auto-regenerate these files

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r8785 | oej | 2006-01-27 09:02:16 +0100 (Fri, 27 Jan 2006) | 2 lines

Issue 6362 - Register without Contact: and Expires: fails (reporter: op)

........
r8808 | oej | 2006-01-28 14:52:15 +0100 (Sat, 28 Jan 2006) | 3 lines

Issue 6182 - Don't remove scheduled event until it's really done.
(reported by malverian)

........

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=8891

By: Digium Subversion (svnbot) 2008-01-15 16:28:31.000-0600

Repository: asterisk
Revision: 8894

_U  team/oej/moduletest/
U   team/oej/moduletest/apps/app_dial.c
U   team/oej/moduletest/apps/app_queue.c
U   team/oej/moduletest/ast_expr2.c
U   team/oej/moduletest/ast_expr2.h
U   team/oej/moduletest/ast_expr2f.c
U   team/oej/moduletest/asterisk.c
U   team/oej/moduletest/channel.c
U   team/oej/moduletest/channels/chan_features.c
U   team/oej/moduletest/channels/chan_iax2.c
U   team/oej/moduletest/channels/chan_sip.c
U   team/oej/moduletest/channels/chan_zap.c
U   team/oej/moduletest/pbx.c
U   team/oej/moduletest/utils/astman.c

------------------------------------------------------------------------
r8894 | oej | 2008-01-15 16:28:30 -0600 (Tue, 15 Jan 2008) | 135 lines

Merged revisions 8320,8347,8394,8412,8414,8418,8429,8433,8437,8445,8537,8562,8573,8588,8600,8608,8619,8632,8666,8677,8710,8729,8758,8785,8808 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r8320 | mogorman | 2006-01-20 02:00:46 +0100 (Fri, 20 Jan 2006) | 3 lines

solved problem with delayreject and iax trunking
bug 4291

........
r8347 | russell | 2006-01-20 19:34:42 +0100 (Fri, 20 Jan 2006) | 2 lines

fix invalid value of prev_q (issue ASTERISK-6142)

........
r8394 | tilghman | 2006-01-21 19:29:39 +0100 (Sat, 21 Jan 2006) | 2 lines

Bug 5936 - AddQueueMember fails on realtime queue, if queue not yet loaded

........
r8412 | russell | 2006-01-22 00:17:06 +0100 (Sun, 22 Jan 2006) | 2 lines

prevent the possibility of writing outside of the available workspace (issue ASTERISK-6111)

........
r8414 | russell | 2006-01-22 00:43:14 +0100 (Sun, 22 Jan 2006) | 2 lines

temporarily revert substring fix pending the result of the discussion in issue ASTERISK-6111

........
r8418 | russell | 2006-01-22 03:05:41 +0100 (Sun, 22 Jan 2006) | 3 lines

add a modified fix to prevent writing outside of the provided workspace when
calculating a substring (issue ASTERISK-6111)

........
r8429 | tilghman | 2006-01-22 09:52:49 +0100 (Sun, 22 Jan 2006) | 2 lines

Bug 6281 - Cannot set more than a single header with SIPAddHeader

........
r8433 | bweschke | 2006-01-22 16:13:41 +0100 (Sun, 22 Jan 2006) | 3 lines

Bug fix: Correct some scenarios where CALL_LIMIT could not be getting adjusted properly allowing chan_sip to send calls when it really shouldn't. Bug ASTERISK-5953


........
r8437 | russell | 2006-01-22 18:47:13 +0100 (Sun, 22 Jan 2006) | 2 lines

fix MixMonitor crash (issue ASTERISK-6161, probably others)

........
r8445 | russell | 2006-01-22 20:03:53 +0100 (Sun, 22 Jan 2006) | 2 lines

fix memory leak from not freeing the queue member list when freeing an old queue

........
r8537 | oej | 2006-01-24 14:15:13 +0100 (Tue, 24 Jan 2006) | 2 lines

Issue ASTERISK-6148 - never send response to ACK. (Reported by whiskerp)

........
r8562 | oej | 2006-01-24 20:21:15 +0100 (Tue, 24 Jan 2006) | 2 lines

Issue 6114: Don't hangup on BYE/ALSO with no channel.

........
r8573 | mattf | 2006-01-24 21:37:30 +0100 (Tue, 24 Jan 2006) | 2 lines

Backport fix for ASTERISK-6071, hangup on polarity reversal

........
r8588 | kpfleming | 2006-01-24 23:32:09 +0100 (Tue, 24 Jan 2006) | 2 lines

ensure that channel cannot become zombie after we check but before we try to start indications

........
r8600 | russell | 2006-01-24 23:55:32 +0100 (Tue, 24 Jan 2006) | 2 lines

completely arbitrary whitespace change for testing something with svnmerge ...

........
r8608 | kpfleming | 2006-01-25 02:50:52 +0100 (Wed, 25 Jan 2006) | 2 lines

ensure hangup cause code is handled properly when channel does not return a frame (issue ASTERISK-6186)

........
r8619 | russell | 2006-01-25 06:38:36 +0100 (Wed, 25 Jan 2006) | 2 lines

don't leak almost 200 bytes for each new channel (issue ASTERISK-6170)

........
r8632 | oej | 2006-01-25 10:46:43 +0100 (Wed, 25 Jan 2006) | 2 lines

Issue ASTERISK-6276 - the "timebomb" bug. Patch by Markster over GPRS

........
r8666 | russell | 2006-01-25 19:39:44 +0100 (Wed, 25 Jan 2006) | 2 lines

fix memory leak (inspired by issue ASTERISK-6190)

........
r8677 | russell | 2006-01-25 20:14:43 +0100 (Wed, 25 Jan 2006) | 3 lines

don't call ast_update_realtime with uninitialized variables if we get a
registration with an expirey of 0 seconds (issue ASTERISK-6016)

........
r8710 | oej | 2006-01-26 15:39:36 +0100 (Thu, 26 Jan 2006) | 2 lines

Issue 5898: Registrations does not get deleted if there's an active SIP dialog

........
r8729 | russell | 2006-01-26 20:42:35 +0100 (Thu, 26 Jan 2006) | 2 lines

fix problem with dtmf on e&m (issue ASTERISK-6203)

........
r8758 | tilghman | 2006-01-27 01:52:12 +0100 (Fri, 27 Jan 2006) | 2 lines

Bug 6072 - Revisions to the source bison and flex files don't auto-regenerate these files

........
r8785 | oej | 2006-01-27 09:02:16 +0100 (Fri, 27 Jan 2006) | 2 lines

Issue 6362 - Register without Contact: and Expires: fails (reporter: op)

........
r8808 | oej | 2006-01-28 14:52:15 +0100 (Sat, 28 Jan 2006) | 3 lines

Issue 6182 - Don't remove scheduled event until it's really done.
(reported by malverian)

........

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=8894