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Summary:ASTERISK-07543: Call limits
Reporter:Badalian Vyacheslav (slavon)Labels:
Date Opened:2006-08-17 01:58:41Date Closed:2007-02-26 10:06:47.000-0600
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Core/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) a74995008190.txt
Description:office*CLI> sip show inuse
* User name               In use          Limit
......
a74995036894-4            0               1
a74995036894-3            1               1
a74995036894-2            1               1
a74995036894-1            1               1
a74995008002              1               1
.....
office*CLI> sip show channels
Peer             User/ANR    Call ID      Seq (Tx/Rx)  Form  Hold     Last Message
195.239.254.158  0060133074  4f1cd834135  00102/00000  ulaw  No       Tx: ACK
10.10.181.48     a749950080  04ca4e3d61b  00101/00102  ulaw  No       Rx: ACK
....
Aug 17 09:57:12 ERROR[4866]: chan_sip.c:2253 update_call_counter: Call from user 'a74995036894-3' rejected due to usage limit of 1
Aug 17 09:57:12 NOTICE[4866]: chan_sip.c:10501 handle_request_invite: Failed to place call for user a74995036894-3, too many calls
...

Why In use 1 on a74995036894-*?

sip reload and all ok 1-2 days...  

office*CLI> show version
Asterisk 1.2.9.1 built by root @ oops on a i686 running Linux on 2006-06-30 05:18:25 UTC
Comments:By: Serge Vecher (serge-v) 2006-08-17 08:37:44

if the last message is ACK, most likely those calls are still up. You should see them with 'show channels'

Please upgrade to 1.2.10. Are you doing any modifications to the code?
By: Olle Johansson (oej) 2006-08-17 09:45:25

In order to understand this, we need to see a bit more of the transactions.

Also, be aware that with a call limit of 1, attended transfers will not be possible.
By: Badalian Vyacheslav (slavon) 2006-08-17 13:34:22

> if the last message is ACK, most likely those calls are still up. You should see them with 'show channels'

U can see that 4 sip peers IN USE. but active 1 sip channel. 195.239.254.158 - is sip trunk to voip provider, 10.10.181.48 = a74995008002. 195.239.254.158 <=> a74995008002 (10.10.181.48). ONLY 1 ACTIVE session! Why IN USE a74995036894-3, a74995036894-2, a74995036894-1?

> Please upgrade to 1.2.10. Are you doing any modifications to the code?
I wait for gentoo packages ver bump to update =(
Are you doing any modifications to the code - nope.

>In order to understand this, we need to see a bit more of the transactions.
>Also, be aware that with a call limit of 1, attended transfers will not be possible.

I understand that if call-limit = 1 work only *2 and #1 transfer, hardware transfer feature in LinkSys PAP2 is off. I specal set it to 1.
I can't do more debug info because it have random time =( if u say what i can see info on freezed devices - i see it.

show sip peer a74995036894-1 - not see any unnormal info.
show channels - u can see up. not have any active channel on this peers to debug or see it =(

sorry for my english ;)
By: Serge Vecher (serge-v) 2006-09-06 11:57:59

is this still an issue with latest releases?
By: Badalian Vyacheslav (slavon) 2006-09-06 12:34:37

Today was update to 1.2.11. Will see...
By: David Liu (deltapath) 2006-09-08 01:53:36

Hi there,

I am using 1.2.11 and I had the same problem as you.

In my sip.conf  I have setup call-limit=1

It was all fine for a long time until today.

sip show inuse shows the channel in use while show channels doesn't show it is active.

And I got the exact same error message as you did.
Call to user 'userXXX' rejected due to usage limit of 1

I did sip reload as you did and everything is working again.

So I guess 1.2.11 doesn't fix the problem.
By: Badalian Vyacheslav (slavon) 2006-09-11 07:08:23

office*CLI> sip show peers
a74995036894-4/a749950368  10.10.105.65     D          5061     UNREACHABLE
a74995036894-3/a749950368  10.10.105.65     D          5060     UNREACHABLE
a74995036894-2/a749950368  (Unspecified)    D          0        UNKNOWN
a74995036894-1/a749950368  (Unspecified)    D          0        UNKNOWN
...
office*CLI> sip show inuse
* User name               In use          Limit
a74995036894-4            0               1
a74995036894-3            0               1
a74995036894-2            1               1
a74995036894-1            1               1
* Peer name               In use          Limit
a74995036894-4            0               1
a74995036894-3            0               1
a74995036894-2            0               1
a74995036894-1            0               1

Whit 5 min... it's turn on...

office*CLI> sip show peers
a74995036894-4/a749950368  10.10.105.65     D          5061     OK (19 ms)
a74995036894-3/a749950368  10.10.105.65     D          5060     OK (14 ms)
a74995036894-2/a749950368  10.10.105.64     D          5061     OK (16 ms)
a74995036894-1/a749950368  10.10.105.64     D          5060     OK (13 ms)
...
office*CLI> sip show inuse
* User name               In use          Limit
a74995036894-4            0               1
a74995036894-3            0               1
a74995036894-2            1               1
a74995036894-1            1               1
* Peer name               In use          Limit
a74995036894-4            0               1
a74995036894-3            0               1
a74995036894-2            0               1
a74995036894-1            0               1
By: Serge Vecher (serge-v) 2006-09-13 09:43:15

alright, 1.2.12.1 is next...
By: Serge Vecher (serge-v) 2006-09-13 09:46:12

deltapath, slavon: If you can reproduce this with 1.2.12.1, please perform the following to capture the SIP Debug needed to diagnose this further.

1) Prepare test environment (reduce the ammount of unrelated traffic on the server);
2) Make sure your logger.conf has the following line:
  console => notice,warning,error,debug
3) restart Asterik.
4) Enable SIP transaction logging with the following CLI commands:
set debug 4
set verbose 4
sip debug
5) Save complete console log to file and _attach_ said file to the bug.

By: Serge Vecher (serge-v) 2006-10-03 16:17:59

feel free to reopen if you can still reproduce and attach debugging information as requested ...
By: Badalian Vyacheslav (slavon) 2006-11-22 08:28:58.000-0600

Hello.

I fix for me this bug. I add to crontab this line
*/20 * * * *    root    asterisk -rx "sip reload" 1>/dev/null

I try in new version (1.2.13) turn off my crontab hack and after 2 weeks i have 1 "zombie" peer. In channels no have connections, but used call limit = 1.

I can't reproduce this bug. I will try turnoff crontab hack again and attach for you full log between last active call and calllimit reject message.

Thanks.
By: Badalian Vyacheslav (slavon) 2006-11-27 05:01:13.000-0600

Please see.
a74995008190 register and first call can't be complited =(

Full log crop from "REGISTER" to Call Try

i can add full log for this time if u need.. gzipped size - 9,8 mb
By: Olle Johansson (oej) 2006-12-01 04:00:12.000-0600

There's something you do that doesn't release the call from the call counters. I would very much like to know what that is. Any transfers, redirects, forwards, strange hangups?
By: Badalian Vyacheslav (slavon) 2006-12-01 04:28:40.000-0600

I think (but i don't know) what do if in call progress have connection problems with ATA(LinkSys PAP2T). But after ReRegister counters of User/Peer don't flash to zero =( Or another logic: after connection problems and drop call - counters dot't flash to zero...

all transfers, redirects and other reject. I use call-limit for user can't transfer call using "Flash" button on phone (function in ATA) and for check busy (ata may use WaitCall function)
By: Olle Johansson (oej) 2007-01-08 05:53:02.000-0600

We need to pinpoint this - there's one action and propably not a regular call, that causes this. Can you try and check inuse before and after transfers?
By: Serge Vecher (serge-v) 2007-01-30 13:52:53.000-0600

I wonder if the patch from 8909 will fix this situation for you. Can you backport patch added to 1.4 in rev.52208 and see if that works for you?
By: Olle Johansson (oej) 2007-02-01 15:13:18.000-0600

Fix committed to 1.2 rev 53090, please test. (also to 1.4 and trunk)
By: Olle Johansson (oej) 2007-02-01 15:17:37.000-0600

Please check with 1.2 rev 53090. Thanks.
By: Badalian Vyacheslav (slavon) 2007-02-02 01:14:40.000-0600

Sorry... i not use SVN.. i wait for next relies and check it
By: Olle Johansson (oej) 2007-02-02 09:31:03.000-0600

I believe this is fixed, and as we don't get any test results, I'll go ahead and close. If the problem re-appears, then re-open this report or open a new one. THanks.
By: Badalian Vyacheslav (slavon) 2007-02-05 03:35:56.000-0600

SVN-branch-1.4-r52494

Have this problem.
sip reload don't clear inuse counters! reload chan_sip.so also don't clear counters! Have many zombie InUse... also have in sip show channels many channels what close many time ago.

after some time i put dital log with debug
By: Badalian Vyacheslav (slavon) 2007-02-05 09:27:02.000-0600

i turn off (comment)
notifyringing = yes
notifyhold = yes
limitonpeers = yes

and restart astersik

and channels on sip show channels not more zombie...
By: Serge Vecher (serge-v) 2007-02-05 13:24:49.000-0600

slavon: can you please try with 1.4 branch revision > 53000?
By: Badalian Vyacheslav (slavon) 2007-02-06 03:00:54.000-0600

hmm... svn up say (today sync 2007-02-06 12:00:46)
Updated to revision 105.
Updated to revision 53281.

asterisk say
asterisk*CLI> show version
Asterisk SVN-branch-1.4-r52494 built by root @ asterisk on a i686 running Linux on 2007-02-06 08:58:58 UTC
By: Serge Vecher (serge-v) 2007-02-07 10:53:54.000-0600

It is best to delete the repository, do a clean check, and make clean.
By: Badalian Vyacheslav (slavon) 2007-02-08 06:10:00.000-0600

asterisk*CLI> show channels
Channel              Location             State   Application(Data)
SIP/a74957871909-1-0 7871909@out-custom-a Ringing AppDial((Outgoing Line))
SIP/87.255.5.254-082 7871909@cisco3600-in Ring    Dial(SIP/a74957871909-1|18|tT)
SIP/cisco3600-trunk- 89169542750@cisco360 Down    AppDial((Outgoing Line))
SIP/87.255.0.210-082 89169542750@out-def- Ring    Dial(SIP/003100000889169542750
SIP/cisco3600-trunk- (None)               Up      Bridged Call(SIP/87.255.5.115-
SIP/87.255.5.115-082 7305790@out-def-city Up      Dial(SIP/00310000087305790@cis
6 active channels
3 active calls


asterisk*CLI> sip show channels
Peer             User/ANR    Call ID      Seq (Tx/Rx)  Form  Hold     Last Message
87.255.5.254     0031000008  37d289af371  00102/00000  ulaw  No       Init: INVITE
87.255.0.210     117         251d4180-1d  00101/00101  ulaw  No       Rx: INVITE
10.10.14.66      (None)      e67bbdd3-9c  00101/37961  unkn  No       Rx: REGISTER
87.255.5.254     0031000008  6b39af6921c  00102/00000  ulaw  No       Tx: ACK
10.10.105.65     a749950368  2382a9d4-c8  00101/00102  ulaw  No       Rx: ACK
87.255.5.254     0031000008  00bc54ac336  00102/00000  ulaw  No       Tx: ACK
87.255.5.115     130         D95E8279-E2  00101/00101  ulaw  No       Rx: ACK
10.10.181.48     a749950080  1fa177073c3  00102/00000  unkn  No       Init: INVITE
10.10.181.48     a749950080  71c6fd9f21d  00102/00000  unkn  No       Init: INVITE
10.10.181.48     a749950080  1468acb36d8  00102/00000  unkn  No       Init: INVITE
10.10.181.48     a749950080  32ffbe78644  00102/00000  unkn  No       Init: INVITE
10.10.135.14     381         3088b00a098  00102/00000  unkn  No       Init: INVITE
10.10.208.28     a749577575  37c49ccc3bf  00102/00000  unkn  No       Init: INVITE
10.10.155.177    382         07f5fa377e5  00102/00000  unkn  No       Init: INVITE
14 active SIP channels


381 - Freeze... I Can't call to him... his can call
By: Olle Johansson (oej) 2007-02-14 10:36:24.000-0600

I've added a change to 1.2 svn version which may help this issue, it did for me.

Also, if you still have issues after upgrading to that version, test with and without "canreinvite=no" to see if there's any change. Thanks.
By: Badalian Vyacheslav (slavon) 2007-02-14 11:42:43.000-0600

i allready use 1.4 SVN version...
canreinvite=no  - i need always use on all peers...
By: Serge Vecher (serge-v) 2007-02-21 11:39:34.000-0600

ok, please test 1.4 revision 55914 or higher
By: Badalian Vyacheslav (slavon) 2007-02-23 03:56:08.000-0600

* Peer name               In use          Limit
a74957871909-8            -1/0/0          1

Connected to Asterisk SVN-trunk-r56209M currently running on asterisk (pid = 3935)

asterisk asterisk # svn info
Path: .
URL: http://svn.digium.com/svn/asterisk/trunk
Repository Root: http://svn.digium.com/svn/asterisk
Repository UUID: 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Revision: 56228
Node Kind: directory
Schedule: normal
Last Changed Author: kpfleming
Last Changed Rev: 56209
Last Changed Date: 2007-02-22 20:36:46 +0300 (Thu, 22 Feb 2007)
By: Serge Vecher (serge-v) 2007-02-23 09:40:45.000-0600

ok, please produce a new log as per my note 0051641
By: Badalian Vyacheslav (slavon) 2007-02-24 06:24:25.000-0600

Sorry. I turn off all call-limits... i can't more time debug this bug... problems with clients on this subject =(... If u realy say what bug fixed - i try turn on its to see that it fixed or not...

Thanks.
By: Serge Vecher (serge-v) 2007-02-26 10:06:46.000-0600

We can't do much without proper debugging information. There are a number of reports of call-limit related bugs fixed in latest versions of Asterisk. If you have time and opportunity to test and provide debugging information, please reopen the bug with that information provided.